From e64001e8efc107992fd835770f6383d0dc731594 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 20 Nov 2013 13:17:07 +0000 Subject: ASoC: wm5110: Add extra AIF2 channels Signed-off-by: D.J. Barrow Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 8 ++++++++ sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/wm5110.c | 48 ++++++++++++++++++++++++++++++++++++++++++++-- 3 files changed, 55 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6f05b17d196..6977bf9f19a 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -292,6 +292,10 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "AIF1RX8", "AIF2RX1", "AIF2RX2", + "AIF2RX3", + "AIF2RX4", + "AIF2RX5", + "AIF2RX6", "AIF3RX1", "AIF3RX2", "SLIMRX1", @@ -395,6 +399,10 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { 0x27, 0x28, /* AIF2RX1 */ 0x29, + 0x2a, + 0x2b, + 0x2c, + 0x2d, 0x30, /* AIF3RX1 */ 0x31, 0x38, /* SLIMRX1 */ diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 9e81b639269..1f96672c7c1 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -81,7 +81,7 @@ struct arizona_priv { unsigned int spk_ena_pending:1; }; -#define ARIZONA_NUM_MIXER_INPUTS 99 +#define ARIZONA_NUM_MIXER_INPUTS 103 extern const unsigned int arizona_mixer_tlv[]; extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index bbd64384ca1..181de7df2c8 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -302,6 +302,10 @@ ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX3", ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX4", ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX5", ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), @@ -361,6 +365,10 @@ ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX3, ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX4, ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX5, ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX6, ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); @@ -561,11 +569,27 @@ SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX3", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX5", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX6", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX6_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX3", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX5", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX6", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX6_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, @@ -703,6 +727,10 @@ ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), +ARIZONA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"), +ARIZONA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"), +ARIZONA_MIXER_WIDGETS(AIF2TX5, "AIF2TX5"), +ARIZONA_MIXER_WIDGETS(AIF2TX6, "AIF2TX6"), ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), @@ -764,6 +792,10 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"), { name, "AIF1RX8", "AIF1RX8" }, \ { name, "AIF2RX1", "AIF2RX1" }, \ { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF2RX3", "AIF2RX3" }, \ + { name, "AIF2RX4", "AIF2RX4" }, \ + { name, "AIF2RX5", "AIF2RX5" }, \ + { name, "AIF2RX6", "AIF2RX6" }, \ { name, "AIF3RX1", "AIF3RX1" }, \ { name, "AIF3RX2", "AIF3RX2" }, \ { name, "SLIMRX1", "SLIMRX1" }, \ @@ -861,9 +893,17 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF2 Capture", NULL, "AIF2TX1" }, { "AIF2 Capture", NULL, "AIF2TX2" }, + { "AIF2 Capture", NULL, "AIF2TX3" }, + { "AIF2 Capture", NULL, "AIF2TX4" }, + { "AIF2 Capture", NULL, "AIF2TX5" }, + { "AIF2 Capture", NULL, "AIF2TX6" }, { "AIF2RX1", NULL, "AIF2 Playback" }, { "AIF2RX2", NULL, "AIF2 Playback" }, + { "AIF2RX3", NULL, "AIF2 Playback" }, + { "AIF2RX4", NULL, "AIF2 Playback" }, + { "AIF2RX5", NULL, "AIF2 Playback" }, + { "AIF2RX6", NULL, "AIF2 Playback" }, { "AIF3 Capture", NULL, "AIF3TX1" }, { "AIF3 Capture", NULL, "AIF3TX2" }, @@ -947,6 +987,10 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + ARIZONA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"), + ARIZONA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"), + ARIZONA_MIXER_ROUTES("AIF2TX5", "AIF2TX5"), + ARIZONA_MIXER_ROUTES("AIF2TX6", "AIF2TX6"), ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), @@ -1067,14 +1111,14 @@ static struct snd_soc_dai_driver wm5110_dai[] = { .playback = { .stream_name = "AIF2 Playback", .channels_min = 1, - .channels_max = 2, + .channels_max = 6, .rates = WM5110_RATES, .formats = WM5110_FORMATS, }, .capture = { .stream_name = "AIF2 Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 6, .rates = WM5110_RATES, .formats = WM5110_FORMATS, }, -- cgit v1.2.3-70-g09d2 From 254dc326dbfd23c2678fafad1b84fc0e42ac4374 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Nov 2013 16:04:03 +0000 Subject: ASoC: wm5110: Expose input high pass filter controls Acked-by: Mark Brown Signed-off-by: Charles Keepax Signed-off-by: Lee Jones --- drivers/mfd/wm5110-tables.c | 4 ++++ include/linux/mfd/arizona/registers.h | 37 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.c | 10 ++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 19 ++++++++++++++++++ 5 files changed, 71 insertions(+) (limited to 'sound/soc') diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 4430404471c..3b079f6b021 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -518,6 +518,7 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000300, 0x0000 }, /* R768 - Input Enables */ { 0x00000308, 0x0000 }, /* R776 - Input Rate */ { 0x00000309, 0x0022 }, /* R777 - Input Volume Ramp */ + { 0x0000030C, 0x0002 }, /* R780 - HPF Control */ { 0x00000310, 0x2080 }, /* R784 - IN1L Control */ { 0x00000311, 0x0180 }, /* R785 - ADC Digital Volume 1L */ { 0x00000312, 0x0000 }, /* R786 - DMIC1L Control */ @@ -539,6 +540,7 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000328, 0x2000 }, /* R808 - IN4L Control */ { 0x00000329, 0x0180 }, /* R809 - ADC Digital Volume 4L */ { 0x0000032A, 0x0000 }, /* R810 - DMIC4L Control */ + { 0x0000032C, 0x0000 }, /* R812 - IN4R Control */ { 0x0000032D, 0x0180 }, /* R813 - ADC Digital Volume 4R */ { 0x0000032E, 0x0000 }, /* R814 - DMIC4R Control */ { 0x00000400, 0x0000 }, /* R1024 - Output Enables 1 */ @@ -1512,6 +1514,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_INPUT_ENABLES_STATUS: case ARIZONA_INPUT_RATE: case ARIZONA_INPUT_VOLUME_RAMP: + case ARIZONA_HPF_CONTROL: case ARIZONA_IN1L_CONTROL: case ARIZONA_ADC_DIGITAL_VOLUME_1L: case ARIZONA_DMIC1L_CONTROL: @@ -1533,6 +1536,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_IN4L_CONTROL: case ARIZONA_ADC_DIGITAL_VOLUME_4L: case ARIZONA_DMIC4L_CONTROL: + case ARIZONA_IN4R_CONTROL: case ARIZONA_ADC_DIGITAL_VOLUME_4R: case ARIZONA_DMIC4R_CONTROL: case ARIZONA_OUTPUT_ENABLES_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 4706d3d46e5..cdf1f5acbe5 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -139,6 +139,7 @@ #define ARIZONA_INPUT_ENABLES_STATUS 0x301 #define ARIZONA_INPUT_RATE 0x308 #define ARIZONA_INPUT_VOLUME_RAMP 0x309 +#define ARIZONA_HPF_CONTROL 0x30C #define ARIZONA_IN1L_CONTROL 0x310 #define ARIZONA_ADC_DIGITAL_VOLUME_1L 0x311 #define ARIZONA_DMIC1L_CONTROL 0x312 @@ -160,6 +161,7 @@ #define ARIZONA_IN4L_CONTROL 0x328 #define ARIZONA_ADC_DIGITAL_VOLUME_4L 0x329 #define ARIZONA_DMIC4L_CONTROL 0x32A +#define ARIZONA_IN4R_CONTROL 0x32C #define ARIZONA_ADC_DIGITAL_VOLUME_4R 0x32D #define ARIZONA_DMIC4R_CONTROL 0x32E #define ARIZONA_OUTPUT_ENABLES_1 0x400 @@ -2292,9 +2294,19 @@ #define ARIZONA_IN_VI_RAMP_SHIFT 0 /* IN_VI_RAMP - [2:0] */ #define ARIZONA_IN_VI_RAMP_WIDTH 3 /* IN_VI_RAMP - [2:0] */ +/* + * R780 (0x30C) - HPF Control + */ +#define ARIZONA_IN_HPF_CUT_MASK 0x0007 /* IN_HPF_CUT [2:0] */ +#define ARIZONA_IN_HPF_CUT_SHIFT 0 /* IN_HPF_CUT [2:0] */ +#define ARIZONA_IN_HPF_CUT_WIDTH 3 /* IN_HPF_CUT [2:0] */ + /* * R784 (0x310) - IN1L Control */ +#define ARIZONA_IN1L_HPF_MASK 0x8000 /* IN1L_HPF - [15] */ +#define ARIZONA_IN1L_HPF_SHIFT 15 /* IN1L_HPF - [15] */ +#define ARIZONA_IN1L_HPF_WIDTH 1 /* IN1L_HPF - [15] */ #define ARIZONA_IN1_OSR_MASK 0x6000 /* IN1_OSR - [14:13] */ #define ARIZONA_IN1_OSR_SHIFT 13 /* IN1_OSR - [14:13] */ #define ARIZONA_IN1_OSR_WIDTH 2 /* IN1_OSR - [14:13] */ @@ -2333,6 +2345,9 @@ /* * R788 (0x314) - IN1R Control */ +#define ARIZONA_IN1R_HPF_MASK 0x8000 /* IN1R_HPF - [15] */ +#define ARIZONA_IN1R_HPF_SHIFT 15 /* IN1R_HPF - [15] */ +#define ARIZONA_IN1R_HPF_WIDTH 1 /* IN1R_HPF - [15] */ #define ARIZONA_IN1R_PGA_VOL_MASK 0x00FE /* IN1R_PGA_VOL - [7:1] */ #define ARIZONA_IN1R_PGA_VOL_SHIFT 1 /* IN1R_PGA_VOL - [7:1] */ #define ARIZONA_IN1R_PGA_VOL_WIDTH 7 /* IN1R_PGA_VOL - [7:1] */ @@ -2362,6 +2377,9 @@ /* * R792 (0x318) - IN2L Control */ +#define ARIZONA_IN2L_HPF_MASK 0x8000 /* IN2L_HPF - [15] */ +#define ARIZONA_IN2L_HPF_SHIFT 15 /* IN2L_HPF - [15] */ +#define ARIZONA_IN2L_HPF_WIDTH 1 /* IN2L_HPF - [15] */ #define ARIZONA_IN2_OSR_MASK 0x6000 /* IN2_OSR - [14:13] */ #define ARIZONA_IN2_OSR_SHIFT 13 /* IN2_OSR - [14:13] */ #define ARIZONA_IN2_OSR_WIDTH 2 /* IN2_OSR - [14:13] */ @@ -2400,6 +2418,9 @@ /* * R796 (0x31C) - IN2R Control */ +#define ARIZONA_IN2R_HPF_MASK 0x8000 /* IN2R_HPF - [15] */ +#define ARIZONA_IN2R_HPF_SHIFT 15 /* IN2R_HPF - [15] */ +#define ARIZONA_IN2R_HPF_WIDTH 1 /* IN2R_HPF - [15] */ #define ARIZONA_IN2R_PGA_VOL_MASK 0x00FE /* IN2R_PGA_VOL - [7:1] */ #define ARIZONA_IN2R_PGA_VOL_SHIFT 1 /* IN2R_PGA_VOL - [7:1] */ #define ARIZONA_IN2R_PGA_VOL_WIDTH 7 /* IN2R_PGA_VOL - [7:1] */ @@ -2429,6 +2450,9 @@ /* * R800 (0x320) - IN3L Control */ +#define ARIZONA_IN3L_HPF_MASK 0x8000 /* IN3L_HPF - [15] */ +#define ARIZONA_IN3L_HPF_SHIFT 15 /* IN3L_HPF - [15] */ +#define ARIZONA_IN3L_HPF_WIDTH 1 /* IN3L_HPF - [15] */ #define ARIZONA_IN3_OSR_MASK 0x6000 /* IN3_OSR - [14:13] */ #define ARIZONA_IN3_OSR_SHIFT 13 /* IN3_OSR - [14:13] */ #define ARIZONA_IN3_OSR_WIDTH 2 /* IN3_OSR - [14:13] */ @@ -2467,6 +2491,9 @@ /* * R804 (0x324) - IN3R Control */ +#define ARIZONA_IN3R_HPF_MASK 0x8000 /* IN3R_HPF - [15] */ +#define ARIZONA_IN3R_HPF_SHIFT 15 /* IN3R_HPF - [15] */ +#define ARIZONA_IN3R_HPF_WIDTH 1 /* IN3R_HPF - [15] */ #define ARIZONA_IN3R_PGA_VOL_MASK 0x00FE /* IN3R_PGA_VOL - [7:1] */ #define ARIZONA_IN3R_PGA_VOL_SHIFT 1 /* IN3R_PGA_VOL - [7:1] */ #define ARIZONA_IN3R_PGA_VOL_WIDTH 7 /* IN3R_PGA_VOL - [7:1] */ @@ -2496,6 +2523,9 @@ /* * R808 (0x328) - IN4 Control */ +#define ARIZONA_IN4L_HPF_MASK 0x8000 /* IN4L_HPF - [15] */ +#define ARIZONA_IN4L_HPF_SHIFT 15 /* IN4L_HPF - [15] */ +#define ARIZONA_IN4L_HPF_WIDTH 1 /* IN4L_HPF - [15] */ #define ARIZONA_IN4_OSR_MASK 0x6000 /* IN4_OSR - [14:13] */ #define ARIZONA_IN4_OSR_SHIFT 13 /* IN4_OSR - [14:13] */ #define ARIZONA_IN4_OSR_WIDTH 2 /* IN4_OSR - [14:13] */ @@ -2525,6 +2555,13 @@ #define ARIZONA_IN4L_DMIC_DLY_SHIFT 0 /* IN4L_DMIC_DLY - [5:0] */ #define ARIZONA_IN4L_DMIC_DLY_WIDTH 6 /* IN4L_DMIC_DLY - [5:0] */ +/* + * R812 (0x32C) - IN4R Control + */ +#define ARIZONA_IN4R_HPF_MASK 0x8000 /* IN4R_HPF - [15] */ +#define ARIZONA_IN4R_HPF_SHIFT 15 /* IN4R_HPF - [15] */ +#define ARIZONA_IN4R_HPF_WIDTH 1 /* IN4R_HPF - [15] */ + /* * R813 (0x32D) - ADC Digital Volume 4R */ diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 657808ba141..708326265a3 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -560,6 +560,16 @@ const struct soc_enum arizona_ng_hold = 4, arizona_ng_hold_text); EXPORT_SYMBOL_GPL(arizona_ng_hold); +static const char * const arizona_in_hpf_cut_text[] = { + "2.5Hz", "5Hz", "10Hz", "20Hz", "40Hz" +}; + +const struct soc_enum arizona_in_hpf_cut_enum = + SOC_ENUM_SINGLE(ARIZONA_HPF_CONTROL, ARIZONA_IN_HPF_CUT_SHIFT, + ARRAY_SIZE(arizona_in_hpf_cut_text), + arizona_in_hpf_cut_text); +EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum); + static const char * const arizona_in_dmic_osr_text[] = { "1.536MHz", "3.072MHz", "6.144MHz", }; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 9e81b639269..f8e63865a1c 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -199,6 +199,7 @@ extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; extern const struct soc_enum arizona_ng_hold; +extern const struct soc_enum arizona_in_hpf_cut_enum; extern const struct soc_enum arizona_in_dmic_osr[]; extern int arizona_in_ev(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index bbd64384ca1..ea18e88e0a0 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -76,6 +76,25 @@ SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum), + +SOC_SINGLE("IN1L HPF Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN1R HPF Switch", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN2L HPF Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN2R HPF Switch", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN3L HPF Switch", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN3R HPF Switch", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN4L HPF Switch", ARIZONA_IN4L_CONTROL, + ARIZONA_IN4L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN4R HPF Switch", ARIZONA_IN4R_CONTROL, + ARIZONA_IN4R_HPF_SHIFT, 1, 0), + SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, -- cgit v1.2.3-70-g09d2 From 3635bf09a89cf92b80ac44198c5c8f0989624ea6 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 13 Nov 2013 18:56:24 +0800 Subject: ASoC: soc-pcm: add symmetry for channels and sample bits Some SoCs can only work in mono or stereo mode at one time. So if we let them capture a mono stream while playing a stereo stream, there might be a problem occur to one of these two streams: double paced or slowed down. In soc-pcm.c, we have soc_pcm_apply_symmetry() to apply the rate symmetry. But we don't have one for channels. Likewise, we can treat symmetric_rate as a solution for those SoCs or CODECs which can not handle asymmetrical LRCLK. But it's also impossible for them to handle asymmetrical BCLK. And accodring to BCLK = LRCLK * channel number * slot size(fixed or sample bits), sample bits might also be a problem if they are not using a fixed slot size. Thus, this patch applys symmetry for channels and sample bits. Meanwhile, there might be a race between two substreams if starting simultaneously. Previously, we only added warning to compalin but still using conservative way to let it carry on. However, this patch rejects the second stream with any unmatched parameter to make sure the first existing stream won't be broken. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 6 +++ include/sound/soc.h | 2 + sound/soc/soc-pcm.c | 130 +++++++++++++++++++++++++++++++++++++++--------- 3 files changed, 115 insertions(+), 23 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 800c101bb09..243d3b68969 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -220,6 +220,8 @@ struct snd_soc_dai_driver { struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; unsigned int symmetric_rates:1; + unsigned int symmetric_channels:1; + unsigned int symmetric_samplebits:1; /* probe ordering - for components with runtime dependencies */ int probe_order; @@ -244,6 +246,8 @@ struct snd_soc_dai { unsigned int capture_active:1; /* stream is in use */ unsigned int playback_active:1; /* stream is in use */ unsigned int symmetric_rates:1; + unsigned int symmetric_channels:1; + unsigned int symmetric_samplebits:1; struct snd_pcm_runtime *runtime; unsigned int active; unsigned char probed:1; @@ -258,6 +262,8 @@ struct snd_soc_dai { /* Symmetry data - only valid if symmetry is being enforced */ unsigned int rate; + unsigned int channels; + unsigned int sample_bits; /* parent platform/codec */ struct snd_soc_platform *platform; diff --git a/include/sound/soc.h b/include/sound/soc.h index 1f741cb24f3..1cda7d343d1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -879,6 +879,8 @@ struct snd_soc_dai_link { /* Symmetry requirements */ unsigned int symmetric_rates:1; + unsigned int symmetric_channels:1; + unsigned int symmetric_samplebits:1; /* Do not create a PCM for this DAI link (Backend link) */ unsigned int no_pcm:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 42782c01e41..ed1e077114a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -84,30 +84,97 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret; - if (!soc_dai->driver->symmetric_rates && - !rtd->dai_link->symmetric_rates) - return 0; + if (soc_dai->rate && (soc_dai->driver->symmetric_rates || + rtd->dai_link->symmetric_rates)) { + dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %dHz rate\n", + soc_dai->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + soc_dai->rate, soc_dai->rate); + if (ret < 0) { + dev_err(soc_dai->dev, + "ASoC: Unable to apply rate constraint: %d\n", + ret); + return ret; + } + } - /* This can happen if multiple streams are starting simultaneously - - * the second can need to get its constraints before the first has - * picked a rate. Complain and allow the application to carry on. - */ - if (!soc_dai->rate) { - dev_warn(soc_dai->dev, - "ASoC: Not enforcing symmetric_rates due to race\n"); - return 0; + if (soc_dai->channels && (soc_dai->driver->symmetric_channels || + rtd->dai_link->symmetric_channels)) { + dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d channel(s)\n", + soc_dai->channels); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + soc_dai->channels, + soc_dai->channels); + if (ret < 0) { + dev_err(soc_dai->dev, + "ASoC: Unable to apply channel symmetry constraint: %d\n", + ret); + return ret; + } } - dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %dHz rate\n", soc_dai->rate); + if (soc_dai->sample_bits && (soc_dai->driver->symmetric_samplebits || + rtd->dai_link->symmetric_samplebits)) { + dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d sample bits\n", + soc_dai->sample_bits); - ret = snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - soc_dai->rate, soc_dai->rate); - if (ret < 0) { - dev_err(soc_dai->dev, - "ASoC: Unable to apply rate symmetry constraint: %d\n", - ret); - return ret; + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + soc_dai->sample_bits, + soc_dai->sample_bits); + if (ret < 0) { + dev_err(soc_dai->dev, + "ASoC: Unable to apply sample bits symmetry constraint: %d\n", + ret); + return ret; + } + } + + return 0; +} + +static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int rate, channels, sample_bits, symmetry; + + rate = params_rate(params); + channels = params_channels(params); + sample_bits = snd_pcm_format_physical_width(params_format(params)); + + /* reject unmatched parameters when applying symmetry */ + symmetry = cpu_dai->driver->symmetric_rates || + codec_dai->driver->symmetric_rates || + rtd->dai_link->symmetric_rates; + if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) { + dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", + cpu_dai->rate, rate); + return -EINVAL; + } + + symmetry = cpu_dai->driver->symmetric_channels || + codec_dai->driver->symmetric_channels || + rtd->dai_link->symmetric_channels; + if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) { + dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n", + cpu_dai->channels, channels); + return -EINVAL; + } + + symmetry = cpu_dai->driver->symmetric_samplebits || + codec_dai->driver->symmetric_samplebits || + rtd->dai_link->symmetric_samplebits; + if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) { + dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n", + cpu_dai->sample_bits, sample_bits); + return -EINVAL; } return 0; @@ -384,11 +451,17 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec->active--; /* clear the corresponding DAIs rate when inactive */ - if (!cpu_dai->active) + if (!cpu_dai->active) { cpu_dai->rate = 0; + cpu_dai->channels = 0; + cpu_dai->sample_bits = 0; + } - if (!codec_dai->active) + if (!codec_dai->active) { codec_dai->rate = 0; + codec_dai->channels = 0; + codec_dai->sample_bits = 0; + } /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. @@ -525,6 +598,10 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + ret = soc_pcm_params_symmetry(substream, params); + if (ret) + goto out; + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { @@ -561,9 +638,16 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - /* store the rate for each DAIs */ + /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); + cpu_dai->channels = params_channels(params); + cpu_dai->sample_bits = + snd_pcm_format_physical_width(params_format(params)); + codec_dai->rate = params_rate(params); + codec_dai->channels = params_channels(params); + codec_dai->sample_bits = + snd_pcm_format_physical_width(params_format(params)); out: mutex_unlock(&rtd->pcm_mutex); -- cgit v1.2.3-70-g09d2 From d3383420c969c25deffd33270ebe321e8401191a Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 20 Nov 2013 18:37:09 +0800 Subject: ASoC: soc-pcm: move DAIs parameters cleaning into hw_free() We're now applying soc_hw_params_symmetry() to reject unmatched parameters while we clear parameters in soc_pcm_close(). So here's a use case might be broken by this mechanism: aplay -Dhw:0 44100.wav 48000.wav 32000.wav In this case, we call soc_pcm_open()->soc_pcm_hw_params()->soc_pcm_hw_free() ->soc_pcm_hw_params()->soc_pcm_hw_free()->soc_pcm_close() in order. As we only clear parameters in soc_pcm_close(). The parameters would be remained in the system even if the playback of 44100.wav is finished. Thus, this patch is trying to move parameters cleaning into hw_free() so that the system can continue to serve this kind of use case. Also, since we set them in hw_params(), it should be better to clear them in hw_free() for symmetry. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ed1e077114a..170ff969575 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -450,19 +450,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; - /* clear the corresponding DAIs rate when inactive */ - if (!cpu_dai->active) { - cpu_dai->rate = 0; - cpu_dai->channels = 0; - cpu_dai->sample_bits = 0; - } - - if (!codec_dai->active) { - codec_dai->rate = 0; - codec_dai->channels = 0; - codec_dai->sample_bits = 0; - } - /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ @@ -682,6 +669,19 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + /* clear the corresponding DAIs parameters when going to be inactive */ + if (cpu_dai->active == 1) { + cpu_dai->rate = 0; + cpu_dai->channels = 0; + cpu_dai->sample_bits = 0; + } + + if (codec_dai->active == 1) { + codec_dai->rate = 0; + codec_dai->channels = 0; + codec_dai->sample_bits = 0; + } + /* apply codec digital mute */ if (!codec->active) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); -- cgit v1.2.3-70-g09d2 From 74c375cb85d7374734e6e53af41c724d9a937f8e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:43 -0200 Subject: ASoC: ad193x: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index aea7e52cf71..12c27eb363d 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -413,7 +413,7 @@ static struct spi_driver ad193x_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static const struct regmap_config ad193x_i2c_regmap_config = { .val_bits = 8, @@ -470,7 +470,7 @@ static int __init ad193x_modinit(void) { int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&ad193x_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register AD193X I2C driver: %d\n", @@ -495,7 +495,7 @@ static void __exit ad193x_modexit(void) spi_unregister_driver(&ad193x_spi_driver); #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&ad193x_i2c_driver); #endif } -- cgit v1.2.3-70-g09d2 From 04c3a852f51ff40f32a29e14078432038b5bcdbc Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:44 -0200 Subject: ASoC: adav80x: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 14a7c169d00..f7bf4555274 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -939,7 +939,7 @@ static struct spi_driver adav80x_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static const struct regmap_config adav80x_i2c_regmap_config = { .val_bits = 8, .pad_bits = 1, @@ -985,7 +985,7 @@ static int __init adav80x_init(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&adav80x_i2c_driver); if (ret) return ret; @@ -1001,7 +1001,7 @@ module_init(adav80x_init); static void __exit adav80x_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&adav80x_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From b34d7cf355116f5107fad8d42fb91d067a3b15bc Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:46 -0200 Subject: ASoC: ak4642: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 090d499bb7e..2f861c9b1d6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -511,7 +511,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -576,7 +576,7 @@ static struct i2c_driver ak4642_i2c_driver = { static int __init ak4642_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&ak4642_i2c_driver); #endif return ret; @@ -586,7 +586,7 @@ module_init(ak4642_modinit); static void __exit ak4642_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&ak4642_i2c_driver); #endif -- cgit v1.2.3-70-g09d2 From d8764646e1cc0ad209af29396e59f836b7d8f164 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Wed, 20 Nov 2013 10:04:15 +0100 Subject: ASoC: fsl-ssi: Move ac97 specific setup to seperate function This is a pure cleanup patch to increase code readability. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 57 +++++++++++++++++++++++++++---------------------- 1 file changed, 32 insertions(+), 25 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 35e277379b8..fb8f52a0e7b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -321,6 +321,36 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) return ret; } +static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) +{ + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + + /* + * Setup the clock control register + */ + write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), + &ssi->stccr); + write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), + &ssi->srccr); + + /* + * Enable AC97 mode and startup the SSI + */ + write_ssi(CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV, + &ssi->sacnt); + write_ssi(0xff, &ssi->saccdis); + write_ssi(0x300, &ssi->saccen); + + /* + * Enable SSI, Transmit and Receive. AC97 has to communicate with the + * codec before a stream is started. + */ + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | + CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE); + + write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor); +} + static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; @@ -387,31 +417,8 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) * because it is also running without an active substream. Normally SSI * is only enabled when there is a substream. */ - if (ssi_private->imx_ac97) { - /* - * Setup the clock control register - */ - write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), - &ssi->stccr); - write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13), - &ssi->srccr); - - /* - * Enable AC97 mode and startup the SSI - */ - write_ssi(CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV, - &ssi->sacnt); - write_ssi(0xff, &ssi->saccdis); - write_ssi(0x300, &ssi->saccen); - - /* - * Enable SSI, Transmit and Receive - */ - write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | - CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE); - - write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor); - } + if (ssi_private->imx_ac97) + fsl_ssi_setup_ac97(ssi_private); return 0; } -- cgit v1.2.3-70-g09d2 From c600e95360dac3a3b88f0a2106214dff8e5f56be Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 19 Nov 2013 14:12:25 +0200 Subject: ASoC: hdmi-codec: Add SNDRV_PCM_FMTBIT_32_LE playback format The new playback format is needed for tda998x HDMI audio support. At the moment the only other user of this codec is omap-hdmi-audio. This change should not break anything because omap-hdmi-audio-dai, the cpu-dai of omap-hdmi-audio, enforces sufficient constraints to available sample formats. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index 68342b121c9..32797a8e4ee 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -44,7 +44,7 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, }, .capture = { .stream_name = "Capture", -- cgit v1.2.3-70-g09d2 From 1b488a481c39d9cd36535b6c15fe474546e6460b Mon Sep 17 00:00:00 2001 From: Victor Kamensky Date: Sat, 16 Nov 2013 02:01:19 +0200 Subject: ASoC: omap: mcbsp, mcpdm, dmic: raw read and write endian fix All OMAP IP blocks expect LE data, but CPU may operate in BE mode. Need to use endian neutral functions to read/write h/w registers. I.e instead of __raw_read[lw] and __raw_write[lw] functions code need to use read[lw]_relaxed and write[lw]_relaxed functions. If the first simply reads/writes register, the second will byteswap it if host operates in BE mode. Changes are trivial sed like replacement of __raw_xxx functions with xxx_relaxed variant. Signed-off-by: Victor Kamensky Signed-off-by: Taras Kondratiuk Acked-by: Jarkko Nikula Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/mcbsp.c | 12 ++++++------ sound/soc/omap/omap-dmic.c | 4 ++-- sound/soc/omap/omap-mcpdm.c | 4 ++-- 3 files changed, 10 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 83433fdea32..86c75384c3c 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -36,10 +36,10 @@ static void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) if (mcbsp->pdata->reg_size == 2) { ((u16 *)mcbsp->reg_cache)[reg] = (u16)val; - __raw_writew((u16)val, addr); + writew_relaxed((u16)val, addr); } else { ((u32 *)mcbsp->reg_cache)[reg] = val; - __raw_writel(val, addr); + writel_relaxed(val, addr); } } @@ -48,22 +48,22 @@ static int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, bool from_cache) void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; if (mcbsp->pdata->reg_size == 2) { - return !from_cache ? __raw_readw(addr) : + return !from_cache ? readw_relaxed(addr) : ((u16 *)mcbsp->reg_cache)[reg]; } else { - return !from_cache ? __raw_readl(addr) : + return !from_cache ? readl_relaxed(addr) : ((u32 *)mcbsp->reg_cache)[reg]; } } static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) { - __raw_writel(val, mcbsp->st_data->io_base_st + reg); + writel_relaxed(val, mcbsp->st_data->io_base_st + reg); } static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg) { - return __raw_readl(mcbsp->st_data->io_base_st + reg); + return readl_relaxed(mcbsp->st_data->io_base_st + reg); } #define MCBSP_READ(mcbsp, reg) \ diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 12e566be379..1bd531d718f 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -61,12 +61,12 @@ struct omap_dmic { static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val) { - __raw_writel(val, dmic->io_base + reg); + writel_relaxed(val, dmic->io_base + reg); } static inline int omap_dmic_read(struct omap_dmic *dmic, u16 reg) { - return __raw_readl(dmic->io_base + reg); + return readl_relaxed(dmic->io_base + reg); } static inline void omap_dmic_start(struct omap_dmic *dmic) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index cd9ee167959..2f5b1536477 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -74,12 +74,12 @@ struct omap_mcpdm { static inline void omap_mcpdm_write(struct omap_mcpdm *mcpdm, u16 reg, u32 val) { - __raw_writel(val, mcpdm->io_base + reg); + writel_relaxed(val, mcpdm->io_base + reg); } static inline int omap_mcpdm_read(struct omap_mcpdm *mcpdm, u16 reg) { - return __raw_readl(mcpdm->io_base + reg); + return readl_relaxed(mcpdm->io_base + reg); } #ifdef DEBUG -- cgit v1.2.3-70-g09d2 From a37377314ff068c83f425979142263a17a6f18af Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 8 Nov 2013 12:46:53 +0300 Subject: ASoC: rcar: some dubious one-bit signed bitfields Because these are signed they can either be 0 or -1 instead of 0 and 1 as intended. It doesn't cause a problem from what I can see, but it's dangerous and Sparse complains: sound/soc/sh/rcar/rsnd.h:177:25: error: dubious one-bit signed bitfield Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/sh/rcar/rsnd.h | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 9e463e50e7e..b5ac3a2afc5 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -174,11 +174,11 @@ struct rsnd_dai { struct rsnd_dai_stream playback; struct rsnd_dai_stream capture; - int clk_master:1; - int bit_clk_inv:1; - int frm_clk_inv:1; - int sys_delay:1; - int data_alignment:1; + unsigned int clk_master:1; + unsigned int bit_clk_inv:1; + unsigned int frm_clk_inv:1; + unsigned int sys_delay:1; + unsigned int data_alignment:1; }; #define rsnd_dai_nr(priv) ((priv)->dai_nr) -- cgit v1.2.3-70-g09d2 From 06b2bd23057f9dad149f0dda436c7426c87b986f Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:52 -0200 Subject: ASoC: uda1380: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index fd0a314bc20..726df6d43c2 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -794,7 +794,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda1380 = { .num_dapm_routes = ARRAY_SIZE(uda1380_dapm_routes), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int uda1380_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -840,7 +840,7 @@ static struct i2c_driver uda1380_i2c_driver = { static int __init uda1380_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&uda1380_i2c_driver); if (ret != 0) pr_err("Failed to register UDA1380 I2C driver: %d\n", ret); @@ -851,7 +851,7 @@ module_init(uda1380_modinit); static void __exit uda1380_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&uda1380_i2c_driver); #endif } -- cgit v1.2.3-70-g09d2 From 784cbf8ab4641c874806a938b9c863c91d70fbe5 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 21 Nov 2013 13:32:24 +0200 Subject: ASoC: Rename mid-x86 directory to intel We have other Intel platforms coming having the Smart Sound Technology (SST) so rename the mid-x86 directory to intel as originally directory name reflected only Intel MID platform. Signed-off-by: Jarkko Nikula Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/Kconfig | 2 +- sound/soc/Makefile | 2 +- sound/soc/intel/Kconfig | 13 + sound/soc/intel/Makefile | 5 + sound/soc/intel/mfld_machine.c | 427 +++++++++++++++++++++++ sound/soc/intel/sst_dsp.h | 134 +++++++ sound/soc/intel/sst_platform.c | 735 +++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst_platform.h | 157 +++++++++ sound/soc/mid-x86/Kconfig | 13 - sound/soc/mid-x86/Makefile | 5 - sound/soc/mid-x86/mfld_machine.c | 427 ----------------------- sound/soc/mid-x86/sst_dsp.h | 134 ------- sound/soc/mid-x86/sst_platform.c | 735 --------------------------------------- sound/soc/mid-x86/sst_platform.h | 157 --------- 14 files changed, 1473 insertions(+), 1473 deletions(-) create mode 100644 sound/soc/intel/Kconfig create mode 100644 sound/soc/intel/Makefile create mode 100644 sound/soc/intel/mfld_machine.c create mode 100644 sound/soc/intel/sst_dsp.h create mode 100644 sound/soc/intel/sst_platform.c create mode 100644 sound/soc/intel/sst_platform.h delete mode 100644 sound/soc/mid-x86/Kconfig delete mode 100644 sound/soc/mid-x86/Makefile delete mode 100644 sound/soc/mid-x86/mfld_machine.c delete mode 100644 sound/soc/mid-x86/sst_dsp.h delete mode 100644 sound/soc/mid-x86/sst_platform.c delete mode 100644 sound/soc/mid-x86/sst_platform.h (limited to 'sound/soc') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5138b849305..463a9e25e04 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -42,7 +42,7 @@ source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" -source "sound/soc/mid-x86/Kconfig" +source "sound/soc/intel/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8b9e70105dd..ff291d3e60a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -16,7 +16,7 @@ obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += dwc/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += jz4740/ -obj-$(CONFIG_SND_SOC) += mid-x86/ +obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig new file mode 100644 index 00000000000..61c10bf503d --- /dev/null +++ b/sound/soc/intel/Kconfig @@ -0,0 +1,13 @@ +config SND_MFLD_MACHINE + tristate "SOC Machine Audio driver for Intel Medfield MID platform" + depends on INTEL_SCU_IPC + select SND_SOC_SN95031 + select SND_SST_PLATFORM + help + This adds support for ASoC machine driver for Intel(R) MID Medfield platform + used as alsa device in audio substem in Intel(R) MID devices + Say Y if you have such a device + If unsure select "N". + +config SND_SST_PLATFORM + tristate diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile new file mode 100644 index 00000000000..63988333946 --- /dev/null +++ b/sound/soc/intel/Makefile @@ -0,0 +1,5 @@ +snd-soc-sst-platform-objs := sst_platform.o +snd-soc-mfld-machine-objs := mfld_machine.o + +obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o +obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c new file mode 100644 index 00000000000..d3d4c32434f --- /dev/null +++ b/sound/soc/intel/mfld_machine.c @@ -0,0 +1,427 @@ +/* + * mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../codecs/sn95031.h" + +#define MID_MONO 1 +#define MID_STEREO 2 +#define MID_MAX_CAP 5 +#define MFLD_JACK_INSERT 0x04 + +enum soc_mic_bias_zones { + MFLD_MV_START = 0, + /* mic bias volutage range for Headphones*/ + MFLD_MV_HP = 400, + /* mic bias volutage range for American Headset*/ + MFLD_MV_AM_HS = 650, + /* mic bias volutage range for Headset*/ + MFLD_MV_HS = 2000, + MFLD_MV_UNDEFINED, +}; + +static unsigned int hs_switch; +static unsigned int lo_dac; + +struct mfld_mc_private { + void __iomem *int_base; + u8 interrupt_status; +}; + +struct snd_soc_jack mfld_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin mfld_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "AMIC1", + .mask = SND_JACK_MICROPHONE, + }, +}; + +/* jack detection voltage zones */ +static struct snd_soc_jack_zone mfld_zones[] = { + {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE}, + {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET}, +}; + +/* sound card controls */ +static const char *headset_switch_text[] = {"Earpiece", "Headset"}; + +static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"}; + +static const struct soc_enum headset_enum = + SOC_ENUM_SINGLE_EXT(2, headset_switch_text); + +static const struct soc_enum lo_enum = + SOC_ENUM_SINGLE_EXT(4, lo_text); + +static int headset_get_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = hs_switch; + return 0; +} + +static int headset_set_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == hs_switch) + return 0; + + if (ucontrol->value.integer.value[0]) { + pr_debug("hs_set HS path\n"); + snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + } else { + pr_debug("hs_set EP path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); + snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); + } + snd_soc_dapm_sync(&codec->dapm); + hs_switch = ucontrol->value.integer.value[0]; + + return 0; +} + +static void lo_enable_out_pins(struct snd_soc_codec *codec) +{ + snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); + snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); + if (hs_switch) { + snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + } else { + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); + snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); + } +} + +static int lo_get_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = lo_dac; + return 0; +} + +static int lo_set_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == lo_dac) + return 0; + + /* we dont want to work with last state of lineout so just enable all + * pins and then disable pins not required + */ + lo_enable_out_pins(codec); + switch (ucontrol->value.integer.value[0]) { + case 0: + pr_debug("set vibra path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT"); + snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0); + break; + + case 1: + pr_debug("set hs path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); + break; + + case 2: + pr_debug("set spkr path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44); + break; + + case 3: + pr_debug("set null path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66); + break; + } + snd_soc_dapm_sync(&codec->dapm); + lo_dac = ucontrol->value.integer.value[0]; + return 0; +} + +static const struct snd_kcontrol_new mfld_snd_controls[] = { + SOC_ENUM_EXT("Playback Switch", headset_enum, + headset_get_switch, headset_set_switch), + SOC_ENUM_EXT("Lineout Mux", lo_enum, + lo_get_switch, lo_set_switch), +}; + +static const struct snd_soc_dapm_widget mfld_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route mfld_map[] = { + {"Headphones", NULL, "HPOUTR"}, + {"Headphones", NULL, "HPOUTL"}, + {"Mic", NULL, "AMIC1"}, +}; + +static void mfld_jack_check(unsigned int intr_status) +{ + struct mfld_jack_data jack_data; + + jack_data.mfld_jack = &mfld_jack; + jack_data.intr_id = intr_status; + + sn95031_jack_detection(&jack_data); + /* TODO: add american headset detection post gpiolib support */ +} + +static int mfld_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret_val; + + /* Add jack sense widgets */ + snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets)); + + /* Set up the map */ + snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map)); + + /* always connected */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Mic"); + + ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls, + ARRAY_SIZE(mfld_snd_controls)); + if (ret_val) { + pr_err("soc_add_controls failed %d", ret_val); + return ret_val; + } + /* default is earpiece pin, userspace sets it explcitly */ + snd_soc_dapm_disable_pin(dapm, "Headphones"); + /* default is lineout NC, userspace sets it explcitly */ + snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); + snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); + lo_dac = 3; + hs_switch = 0; + /* we dont use linein in this so set to NC */ + snd_soc_dapm_disable_pin(dapm, "LINEINL"); + snd_soc_dapm_disable_pin(dapm, "LINEINR"); + + /* Headset and button jack detection */ + ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1, &mfld_jack); + if (ret_val) { + pr_err("jack creation failed\n"); + return ret_val; + } + + ret_val = snd_soc_jack_add_pins(&mfld_jack, + ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); + if (ret_val) { + pr_err("adding jack pins failed\n"); + return ret_val; + } + ret_val = snd_soc_jack_add_zones(&mfld_jack, + ARRAY_SIZE(mfld_zones), mfld_zones); + if (ret_val) { + pr_err("adding jack zones failed\n"); + return ret_val; + } + + /* we want to check if anything is inserted at boot, + * so send a fake event to codec and it will read adc + * to find if anything is there or not */ + mfld_jack_check(MFLD_JACK_INSERT); + return ret_val; +} + +static struct snd_soc_dai_link mfld_msic_dailink[] = { + { + .name = "Medfield Headset", + .stream_name = "Headset", + .cpu_dai_name = "Headset-cpu-dai", + .codec_dai_name = "SN95031 Headset", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = mfld_init, + }, + { + .name = "Medfield Speaker", + .stream_name = "Speaker", + .cpu_dai_name = "Speaker-cpu-dai", + .codec_dai_name = "SN95031 Speaker", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, + { + .name = "Medfield Vibra", + .stream_name = "Vibra1", + .cpu_dai_name = "Vibra1-cpu-dai", + .codec_dai_name = "SN95031 Vibra1", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, + { + .name = "Medfield Haptics", + .stream_name = "Vibra2", + .cpu_dai_name = "Vibra2-cpu-dai", + .codec_dai_name = "SN95031 Vibra2", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, + { + .name = "Medfield Compress", + .stream_name = "Speaker", + .cpu_dai_name = "Compress-cpu-dai", + .codec_dai_name = "SN95031 Speaker", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_mfld = { + .name = "medfield_audio", + .owner = THIS_MODULE, + .dai_link = mfld_msic_dailink, + .num_links = ARRAY_SIZE(mfld_msic_dailink), +}; + +static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) +{ + struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; + + memcpy_fromio(&mc_private->interrupt_status, + ((void *)(mc_private->int_base)), + sizeof(u8)); + return IRQ_WAKE_THREAD; +} + +static irqreturn_t snd_mfld_jack_detection(int irq, void *data) +{ + struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; + + if (mfld_jack.codec == NULL) + return IRQ_HANDLED; + mfld_jack_check(mc_drv_ctx->interrupt_status); + + return IRQ_HANDLED; +} + +static int snd_mfld_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0, irq; + struct mfld_mc_private *mc_drv_ctx; + struct resource *irq_mem; + + pr_debug("snd_mfld_mc_probe called\n"); + + /* retrive the irq number */ + irq = platform_get_irq(pdev, 0); + + /* audio interrupt base of SRAM location where + * interrupts are stored by System FW */ + mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC); + if (!mc_drv_ctx) { + pr_err("allocation failed\n"); + return -ENOMEM; + } + + irq_mem = platform_get_resource_byname( + pdev, IORESOURCE_MEM, "IRQ_BASE"); + if (!irq_mem) { + pr_err("no mem resource given\n"); + return -ENODEV; + } + mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start, + resource_size(irq_mem)); + if (!mc_drv_ctx->int_base) { + pr_err("Mapping of cache failed\n"); + return -ENOMEM; + } + /* register for interrupt */ + ret_val = devm_request_threaded_irq(&pdev->dev, irq, + snd_mfld_jack_intr_handler, + snd_mfld_jack_detection, + IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); + if (ret_val) { + pr_err("cannot register IRQ\n"); + return ret_val; + } + /* register the soc card */ + snd_soc_card_mfld.dev = &pdev->dev; + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld); + if (ret_val) { + pr_debug("snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, mc_drv_ctx); + pr_debug("successfully exited probe\n"); + return 0; +} + +static struct platform_driver snd_mfld_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "msic_audio", + }, + .probe = snd_mfld_mc_probe, +}; + +module_platform_driver(snd_mfld_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver"); +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:msic-audio"); diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst_dsp.h new file mode 100644 index 00000000000..0fce1de284f --- /dev/null +++ b/sound/soc/intel/sst_dsp.h @@ -0,0 +1,134 @@ +#ifndef __SST_DSP_H__ +#define __SST_DSP_H__ +/* + * sst_dsp.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-12 Intel Corporation + * Authors: Vinod Koul + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +enum sst_codec_types { + /* AUDIO/MUSIC CODEC Type Definitions */ + SST_CODEC_TYPE_UNKNOWN = 0, + SST_CODEC_TYPE_PCM, /* Pass through Audio codec */ + SST_CODEC_TYPE_MP3, + SST_CODEC_TYPE_MP24, + SST_CODEC_TYPE_AAC, + SST_CODEC_TYPE_AACP, + SST_CODEC_TYPE_eAACP, +}; + +enum stream_type { + SST_STREAM_TYPE_NONE = 0, + SST_STREAM_TYPE_MUSIC = 1, +}; + +struct snd_pcm_params { + u16 codec; /* codec type */ + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 reserved; /* Bitrate in bits per second */ + u32 sfreq; /* Sampling rate in Hz */ + u8 use_offload_path; + u8 reserved2; + u16 reserved3; + u8 channel_map[8]; +} __packed; + +/* MP3 Music Parameters Message */ +struct snd_mp3_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u8 crc_check; /* crc_check - disable (0) or enable (1) */ + u8 reserved1; /* unused*/ + u16 reserved2; /* Unused */ +} __packed; + +#define AAC_BIT_STREAM_ADTS 0 +#define AAC_BIT_STREAM_ADIF 1 +#define AAC_BIT_STREAM_RAW 2 + +/* AAC Music Parameters Message */ +struct snd_aac_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo*/ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */ + u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */ + u16 reser2; + u32 externalsr; /*sampling rate of basic AAC raw bit stream*/ + u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/ + u8 reser1; + u16 reser3; +} __packed; + +/* WMA Music Parameters Message */ +struct snd_wma_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 brate; /* Use the hard coded value. */ + u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ + u32 channel_mask; /* Channel Mask */ + u16 format_tag; /* Format Tag */ + u16 block_align; /* packet size */ + u16 wma_encode_opt;/* Encoder option */ + u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB */ + u8 reserved; /* reserved */ +} __packed; + +/* Codec params struture */ +union snd_sst_codec_params { + struct snd_pcm_params pcm_params; + struct snd_mp3_params mp3_params; + struct snd_aac_params aac_params; + struct snd_wma_params wma_params; +} __packed; + +/* Address and size info of a frame buffer */ +struct sst_address_info { + u32 addr; /* Address at IA */ + u32 size; /* Size of the buffer */ +}; + +struct snd_sst_alloc_params_ext { + struct sst_address_info ring_buf_info[8]; + u8 sg_count; + u8 reserved; + u16 reserved2; + u32 frag_size; /*Number of samples after which period elapsed + message is sent valid only if path = 0*/ +} __packed; + +struct snd_sst_stream_params { + union snd_sst_codec_params uc; +} __packed; + +struct snd_sst_params { + u32 stream_id; + u8 codec; + u8 ops; + u8 stream_type; + u8 device_type; + struct snd_sst_stream_params sparams; + struct snd_sst_alloc_params_ext aparams; +}; + +#endif /* __SST_DSP_H__ */ diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst_platform.c new file mode 100644 index 00000000000..b6b5eb698d3 --- /dev/null +++ b/sound/soc/intel/sst_platform.c @@ -0,0 +1,735 @@ +/* + * sst_platform.c - Intel MID Platform driver + * + * Copyright (C) 2010-2013 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include +#include +#include +#include +#include +#include "sst_platform.h" + +static struct sst_device *sst; +static DEFINE_MUTEX(sst_lock); + +int sst_register_dsp(struct sst_device *dev) +{ + if (WARN_ON(!dev)) + return -EINVAL; + if (!try_module_get(dev->dev->driver->owner)) + return -ENODEV; + mutex_lock(&sst_lock); + if (sst) { + pr_err("we already have a device %s\n", sst->name); + module_put(dev->dev->driver->owner); + mutex_unlock(&sst_lock); + return -EEXIST; + } + pr_debug("registering device %s\n", dev->name); + sst = dev; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_register_dsp); + +int sst_unregister_dsp(struct sst_device *dev) +{ + if (WARN_ON(!dev)) + return -EINVAL; + if (dev != sst) + return -EINVAL; + + mutex_lock(&sst_lock); + + if (!sst) { + mutex_unlock(&sst_lock); + return -EIO; + } + + module_put(sst->dev->driver->owner); + pr_debug("unreg %s\n", sst->name); + sst = NULL; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_unregister_dsp); + +static struct snd_pcm_hardware sst_platform_pcm_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_DOUBLE | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP| + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 | + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 | + SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32), + .rates = (SNDRV_PCM_RATE_8000| + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000), + .rate_min = SST_MIN_RATE, + .rate_max = SST_MAX_RATE, + .channels_min = SST_MIN_CHANNEL, + .channels_max = SST_MAX_CHANNEL, + .buffer_bytes_max = SST_MAX_BUFFER, + .period_bytes_min = SST_MIN_PERIOD_BYTES, + .period_bytes_max = SST_MAX_PERIOD_BYTES, + .periods_min = SST_MIN_PERIODS, + .periods_max = SST_MAX_PERIODS, + .fifo_size = SST_FIFO_SIZE, +}; + +/* MFLD - MSIC */ +static struct snd_soc_dai_driver sst_platform_dai[] = { +{ + .name = "Headset-cpu-dai", + .id = 0, + .playback = { + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Speaker-cpu-dai", + .id = 1, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Vibra1-cpu-dai", + .id = 2, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_MONO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Vibra2-cpu-dai", + .id = 3, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Compress-cpu-dai", + .compress_dai = 1, + .playback = { + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +}; + +static const struct snd_soc_component_driver sst_component = { + .name = "sst", +}; + +/* helper functions */ +static inline void sst_set_stream_status(struct sst_runtime_stream *stream, + int state) +{ + unsigned long flags; + spin_lock_irqsave(&stream->status_lock, flags); + stream->stream_status = state; + spin_unlock_irqrestore(&stream->status_lock, flags); +} + +static inline int sst_get_stream_status(struct sst_runtime_stream *stream) +{ + int state; + unsigned long flags; + + spin_lock_irqsave(&stream->status_lock, flags); + state = stream->stream_status; + spin_unlock_irqrestore(&stream->status_lock, flags); + return state; +} + +static void sst_fill_pcm_params(struct snd_pcm_substream *substream, + struct sst_pcm_params *param) +{ + + param->codec = SST_CODEC_TYPE_PCM; + param->num_chan = (u8) substream->runtime->channels; + param->pcm_wd_sz = substream->runtime->sample_bits; + param->reserved = 0; + param->sfreq = substream->runtime->rate; + param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); + param->period_count = substream->runtime->period_size; + param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); + pr_debug("period_cnt = %d\n", param->period_count); + pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); +} + +static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream = + substream->runtime->private_data; + struct sst_pcm_params param = {0}; + struct sst_stream_params str_params = {0}; + int ret_val; + + /* set codec params and inform SST driver the same */ + sst_fill_pcm_params(substream, ¶m); + substream->runtime->dma_area = substream->dma_buffer.area; + str_params.sparams = param; + str_params.codec = param.codec; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + str_params.ops = STREAM_OPS_PLAYBACK; + str_params.device_type = substream->pcm->device + 1; + pr_debug("Playbck stream,Device %d\n", + substream->pcm->device); + } else { + str_params.ops = STREAM_OPS_CAPTURE; + str_params.device_type = SND_SST_DEVICE_CAPTURE; + pr_debug("Capture stream,Device %d\n", + substream->pcm->device); + } + ret_val = stream->ops->open(&str_params); + pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); + if (ret_val < 0) + return ret_val; + + stream->stream_info.str_id = ret_val; + pr_debug("str id : %d\n", stream->stream_info.str_id); + return ret_val; +} + +static void sst_period_elapsed(void *mad_substream) +{ + struct snd_pcm_substream *substream = mad_substream; + struct sst_runtime_stream *stream; + int status; + + if (!substream || !substream->runtime) + return; + stream = substream->runtime->private_data; + if (!stream) + return; + status = sst_get_stream_status(stream); + if (status != SST_PLATFORM_RUNNING) + return; + snd_pcm_period_elapsed(substream); +} + +static int sst_platform_init_stream(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream = + substream->runtime->private_data; + int ret_val; + + pr_debug("setting buffer ptr param\n"); + sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.period_elapsed = sst_period_elapsed; + stream->stream_info.mad_substream = substream; + stream->stream_info.buffer_ptr = 0; + stream->stream_info.sfreq = substream->runtime->rate; + ret_val = stream->ops->device_control( + SST_SND_STREAM_INIT, &stream->stream_info); + if (ret_val) + pr_err("control_set ret error %d\n", ret_val); + return ret_val; + +} +/* end -- helper functions */ + +static int sst_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sst_runtime_stream *stream; + int ret_val; + + pr_debug("sst_platform_open called\n"); + + snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); + ret_val = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret_val < 0) + return ret_val; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + spin_lock_init(&stream->status_lock); + + /* get the sst ops */ + mutex_lock(&sst_lock); + if (!sst) { + pr_err("no device available to run\n"); + mutex_unlock(&sst_lock); + kfree(stream); + return -ENODEV; + } + if (!try_module_get(sst->dev->driver->owner)) { + mutex_unlock(&sst_lock); + kfree(stream); + return -ENODEV; + } + stream->ops = sst->ops; + mutex_unlock(&sst_lock); + + stream->stream_info.str_id = 0; + sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.mad_substream = substream; + /* allocate memory for SST API set */ + runtime->private_data = stream; + + return 0; +} + +static int sst_platform_close(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + pr_debug("sst_platform_close called\n"); + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + if (str_id) + ret_val = stream->ops->close(str_id); + module_put(sst->dev->driver->owner); + kfree(stream); + return ret_val; +} + +static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + pr_debug("sst_platform_pcm_prepare called\n"); + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + if (stream->stream_info.str_id) { + ret_val = stream->ops->device_control( + SST_SND_DROP, &str_id); + return ret_val; + } + + ret_val = sst_platform_alloc_stream(substream); + if (ret_val < 0) + return ret_val; + snprintf(substream->pcm->id, sizeof(substream->pcm->id), + "%d", stream->stream_info.str_id); + + ret_val = sst_platform_init_stream(substream); + if (ret_val) + return ret_val; + substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER; + return ret_val; +} + +static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + int ret_val = 0, str_id; + struct sst_runtime_stream *stream; + int str_cmd, status; + + pr_debug("sst_platform_pcm_trigger called\n"); + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pr_debug("sst: Trigger Start\n"); + str_cmd = SST_SND_START; + status = SST_PLATFORM_RUNNING; + stream->stream_info.mad_substream = substream; + break; + case SNDRV_PCM_TRIGGER_STOP: + pr_debug("sst: in stop\n"); + str_cmd = SST_SND_DROP; + status = SST_PLATFORM_DROPPED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + pr_debug("sst: in pause\n"); + str_cmd = SST_SND_PAUSE; + status = SST_PLATFORM_PAUSED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + pr_debug("sst: in pause release\n"); + str_cmd = SST_SND_RESUME; + status = SST_PLATFORM_RUNNING; + break; + default: + return -EINVAL; + } + ret_val = stream->ops->device_control(str_cmd, &str_id); + if (!ret_val) + sst_set_stream_status(stream, status); + + return ret_val; +} + + +static snd_pcm_uframes_t sst_platform_pcm_pointer + (struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val, status; + struct pcm_stream_info *str_info; + + stream = substream->runtime->private_data; + status = sst_get_stream_status(stream); + if (status == SST_PLATFORM_INIT) + return 0; + str_info = &stream->stream_info; + ret_val = stream->ops->device_control( + SST_SND_BUFFER_POINTER, str_info); + if (ret_val) { + pr_err("sst: error code = %d\n", ret_val); + return ret_val; + } + return stream->stream_info.buffer_ptr; +} + +static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + + return 0; +} + +static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static struct snd_pcm_ops sst_platform_ops = { + .open = sst_platform_open, + .close = sst_platform_close, + .ioctl = snd_pcm_lib_ioctl, + .prepare = sst_platform_pcm_prepare, + .trigger = sst_platform_pcm_trigger, + .pointer = sst_platform_pcm_pointer, + .hw_params = sst_platform_pcm_hw_params, + .hw_free = sst_platform_pcm_hw_free, +}; + +static void sst_pcm_free(struct snd_pcm *pcm) +{ + pr_debug("sst_pcm_free called\n"); + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + int retval = 0; + + pr_debug("sst_pcm_new called\n"); + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + SST_MIN_BUFFER, SST_MAX_BUFFER); + if (retval) { + pr_err("dma buffer allocationf fail\n"); + return retval; + } + } + return retval; +} + +/* compress stream operations */ +static void sst_compr_fragment_elapsed(void *arg) +{ + struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg; + + pr_debug("fragment elapsed by driver\n"); + if (cstream) + snd_compr_fragment_elapsed(cstream); +} + +static int sst_platform_compr_open(struct snd_compr_stream *cstream) +{ + + int ret_val = 0; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sst_runtime_stream *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + spin_lock_init(&stream->status_lock); + + /* get the sst ops */ + if (!sst || !try_module_get(sst->dev->driver->owner)) { + pr_err("no device available to run\n"); + ret_val = -ENODEV; + goto out_ops; + } + stream->compr_ops = sst->compr_ops; + + stream->id = 0; + sst_set_stream_status(stream, SST_PLATFORM_INIT); + runtime->private_data = stream; + return 0; +out_ops: + kfree(stream); + return ret_val; +} + +static int sst_platform_compr_free(struct snd_compr_stream *cstream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + stream = cstream->runtime->private_data; + /*need to check*/ + str_id = stream->id; + if (str_id) + ret_val = stream->compr_ops->close(str_id); + module_put(sst->dev->driver->owner); + kfree(stream); + pr_debug("%s: %d\n", __func__, ret_val); + return 0; +} + +static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params) +{ + struct sst_runtime_stream *stream; + int retval; + struct snd_sst_params str_params; + struct sst_compress_cb cb; + + stream = cstream->runtime->private_data; + /* construct fw structure for this*/ + memset(&str_params, 0, sizeof(str_params)); + + str_params.ops = STREAM_OPS_PLAYBACK; + str_params.stream_type = SST_STREAM_TYPE_MUSIC; + str_params.device_type = SND_SST_DEVICE_COMPRESS; + + switch (params->codec.id) { + case SND_AUDIOCODEC_MP3: { + str_params.codec = SST_CODEC_TYPE_MP3; + str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3; + str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in; + str_params.sparams.uc.mp3_params.pcm_wd_sz = 16; + break; + } + + case SND_AUDIOCODEC_AAC: { + str_params.codec = SST_CODEC_TYPE_AAC; + str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC; + str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in; + str_params.sparams.uc.aac_params.pcm_wd_sz = 16; + if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS) + str_params.sparams.uc.aac_params.bs_format = + AAC_BIT_STREAM_ADTS; + else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW) + str_params.sparams.uc.aac_params.bs_format = + AAC_BIT_STREAM_RAW; + else { + pr_err("Undefined format%d\n", params->codec.format); + return -EINVAL; + } + str_params.sparams.uc.aac_params.externalsr = + params->codec.sample_rate; + break; + } + + default: + pr_err("codec not supported, id =%d\n", params->codec.id); + return -EINVAL; + } + + str_params.aparams.ring_buf_info[0].addr = + virt_to_phys(cstream->runtime->buffer); + str_params.aparams.ring_buf_info[0].size = + cstream->runtime->buffer_size; + str_params.aparams.sg_count = 1; + str_params.aparams.frag_size = cstream->runtime->fragment_size; + + cb.param = cstream; + cb.compr_cb = sst_compr_fragment_elapsed; + + retval = stream->compr_ops->open(&str_params, &cb); + if (retval < 0) { + pr_err("stream allocation failed %d\n", retval); + return retval; + } + + stream->id = retval; + return 0; +} + +static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->control(cmd, stream->id); +} + +static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct sst_runtime_stream *stream; + + stream = cstream->runtime->private_data; + stream->compr_ops->tstamp(stream->id, tstamp); + tstamp->byte_offset = tstamp->copied_total % + (u32)cstream->runtime->buffer_size; + pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset); + return 0; +} + +static int sst_platform_compr_ack(struct snd_compr_stream *cstream, + size_t bytes) +{ + struct sst_runtime_stream *stream; + + stream = cstream->runtime->private_data; + stream->compr_ops->ack(stream->id, (unsigned long)bytes); + stream->bytes_written += bytes; + + return 0; +} + +static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream, + struct snd_compr_caps *caps) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->get_caps(caps); +} + +static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, + struct snd_compr_codec_caps *codec) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->get_codec_caps(codec); +} + +static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->set_metadata(stream->id, metadata); +} + +static struct snd_compr_ops sst_platform_compr_ops = { + + .open = sst_platform_compr_open, + .free = sst_platform_compr_free, + .set_params = sst_platform_compr_set_params, + .set_metadata = sst_platform_compr_set_metadata, + .trigger = sst_platform_compr_trigger, + .pointer = sst_platform_compr_pointer, + .ack = sst_platform_compr_ack, + .get_caps = sst_platform_compr_get_caps, + .get_codec_caps = sst_platform_compr_get_codec_caps, +}; + +static struct snd_soc_platform_driver sst_soc_platform_drv = { + .ops = &sst_platform_ops, + .compr_ops = &sst_platform_compr_ops, + .pcm_new = sst_pcm_new, + .pcm_free = sst_pcm_free, +}; + +static int sst_platform_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("sst_platform_probe called\n"); + sst = NULL; + ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); + if (ret) { + pr_err("registering soc platform failed\n"); + return ret; + } + + ret = snd_soc_register_component(&pdev->dev, &sst_component, + sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); + if (ret) { + pr_err("registering cpu dais failed\n"); + snd_soc_unregister_platform(&pdev->dev); + } + return ret; +} + +static int sst_platform_remove(struct platform_device *pdev) +{ + + snd_soc_unregister_component(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); + pr_debug("sst_platform_remove success\n"); + return 0; +} + +static struct platform_driver sst_platform_driver = { + .driver = { + .name = "sst-platform", + .owner = THIS_MODULE, + }, + .probe = sst_platform_probe, + .remove = sst_platform_remove, +}; + +module_platform_driver(sst_platform_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sst-platform"); diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst_platform.h new file mode 100644 index 00000000000..cacc9066ec5 --- /dev/null +++ b/sound/soc/intel/sst_platform.h @@ -0,0 +1,157 @@ +/* + * sst_platform.h - Intel MID Platform driver header file + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ + +#ifndef __SST_PLATFORMDRV_H__ +#define __SST_PLATFORMDRV_H__ + +#include "sst_dsp.h" + +#define SST_MONO 1 +#define SST_STEREO 2 +#define SST_MAX_CAP 5 + +#define SST_MIN_RATE 8000 +#define SST_MAX_RATE 48000 +#define SST_MIN_CHANNEL 1 +#define SST_MAX_CHANNEL 5 +#define SST_MAX_BUFFER (800*1024) +#define SST_MIN_BUFFER (800*1024) +#define SST_MIN_PERIOD_BYTES 32 +#define SST_MAX_PERIOD_BYTES SST_MAX_BUFFER +#define SST_MIN_PERIODS 2 +#define SST_MAX_PERIODS (1024*2) +#define SST_FIFO_SIZE 0 + +struct pcm_stream_info { + int str_id; + void *mad_substream; + void (*period_elapsed) (void *mad_substream); + unsigned long long buffer_ptr; + int sfreq; +}; + +enum sst_drv_status { + SST_PLATFORM_INIT = 1, + SST_PLATFORM_STARTED, + SST_PLATFORM_RUNNING, + SST_PLATFORM_PAUSED, + SST_PLATFORM_DROPPED, +}; + +enum sst_controls { + SST_SND_ALLOC = 0x00, + SST_SND_PAUSE = 0x01, + SST_SND_RESUME = 0x02, + SST_SND_DROP = 0x03, + SST_SND_FREE = 0x04, + SST_SND_BUFFER_POINTER = 0x05, + SST_SND_STREAM_INIT = 0x06, + SST_SND_START = 0x07, + SST_MAX_CONTROLS = 0x07, +}; + +enum sst_stream_ops { + STREAM_OPS_PLAYBACK = 0, + STREAM_OPS_CAPTURE, +}; + +enum sst_audio_device_type { + SND_SST_DEVICE_HEADSET = 1, + SND_SST_DEVICE_IHF, + SND_SST_DEVICE_VIBRA, + SND_SST_DEVICE_HAPTIC, + SND_SST_DEVICE_CAPTURE, + SND_SST_DEVICE_COMPRESS, +}; + +/* PCM Parameters */ +struct sst_pcm_params { + u16 codec; /* codec type */ + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 reserved; /* Bitrate in bits per second */ + u32 sfreq; /* Sampling rate in Hz */ + u32 ring_buffer_size; + u32 period_count; /* period elapsed in samples*/ + u32 ring_buffer_addr; +}; + +struct sst_stream_params { + u32 result; + u32 stream_id; + u8 codec; + u8 ops; + u8 stream_type; + u8 device_type; + struct sst_pcm_params sparams; +}; + +struct sst_compress_cb { + void *param; + void (*compr_cb)(void *param); +}; + +struct compress_sst_ops { + const char *name; + int (*open) (struct snd_sst_params *str_params, + struct sst_compress_cb *cb); + int (*control) (unsigned int cmd, unsigned int str_id); + int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp); + int (*ack) (unsigned int str_id, unsigned long bytes); + int (*close) (unsigned int str_id); + int (*get_caps) (struct snd_compr_caps *caps); + int (*get_codec_caps) (struct snd_compr_codec_caps *codec); + int (*set_metadata) (unsigned int str_id, + struct snd_compr_metadata *mdata); + +}; + +struct sst_ops { + int (*open) (struct sst_stream_params *str_param); + int (*device_control) (int cmd, void *arg); + int (*close) (unsigned int str_id); +}; + +struct sst_runtime_stream { + int stream_status; + unsigned int id; + size_t bytes_written; + struct pcm_stream_info stream_info; + struct sst_ops *ops; + struct compress_sst_ops *compr_ops; + spinlock_t status_lock; +}; + +struct sst_device { + char *name; + struct device *dev; + struct sst_ops *ops; + struct compress_sst_ops *compr_ops; +}; + +int sst_register_dsp(struct sst_device *sst); +int sst_unregister_dsp(struct sst_device *sst); +#endif diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig deleted file mode 100644 index 61c10bf503d..00000000000 --- a/sound/soc/mid-x86/Kconfig +++ /dev/null @@ -1,13 +0,0 @@ -config SND_MFLD_MACHINE - tristate "SOC Machine Audio driver for Intel Medfield MID platform" - depends on INTEL_SCU_IPC - select SND_SOC_SN95031 - select SND_SST_PLATFORM - help - This adds support for ASoC machine driver for Intel(R) MID Medfield platform - used as alsa device in audio substem in Intel(R) MID devices - Say Y if you have such a device - If unsure select "N". - -config SND_SST_PLATFORM - tristate diff --git a/sound/soc/mid-x86/Makefile b/sound/soc/mid-x86/Makefile deleted file mode 100644 index 63988333946..00000000000 --- a/sound/soc/mid-x86/Makefile +++ /dev/null @@ -1,5 +0,0 @@ -snd-soc-sst-platform-objs := sst_platform.o -snd-soc-mfld-machine-objs := mfld_machine.o - -obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o -obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c deleted file mode 100644 index d3d4c32434f..00000000000 --- a/sound/soc/mid-x86/mfld_machine.c +++ /dev/null @@ -1,427 +0,0 @@ -/* - * mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform - * - * Copyright (C) 2010 Intel Corp - * Author: Vinod Koul - * Author: Harsha Priya - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../codecs/sn95031.h" - -#define MID_MONO 1 -#define MID_STEREO 2 -#define MID_MAX_CAP 5 -#define MFLD_JACK_INSERT 0x04 - -enum soc_mic_bias_zones { - MFLD_MV_START = 0, - /* mic bias volutage range for Headphones*/ - MFLD_MV_HP = 400, - /* mic bias volutage range for American Headset*/ - MFLD_MV_AM_HS = 650, - /* mic bias volutage range for Headset*/ - MFLD_MV_HS = 2000, - MFLD_MV_UNDEFINED, -}; - -static unsigned int hs_switch; -static unsigned int lo_dac; - -struct mfld_mc_private { - void __iomem *int_base; - u8 interrupt_status; -}; - -struct snd_soc_jack mfld_jack; - -/*Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin mfld_jack_pins[] = { - { - .pin = "Headphones", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "AMIC1", - .mask = SND_JACK_MICROPHONE, - }, -}; - -/* jack detection voltage zones */ -static struct snd_soc_jack_zone mfld_zones[] = { - {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE}, - {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET}, -}; - -/* sound card controls */ -static const char *headset_switch_text[] = {"Earpiece", "Headset"}; - -static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"}; - -static const struct soc_enum headset_enum = - SOC_ENUM_SINGLE_EXT(2, headset_switch_text); - -static const struct soc_enum lo_enum = - SOC_ENUM_SINGLE_EXT(4, lo_text); - -static int headset_get_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = hs_switch; - return 0; -} - -static int headset_set_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - if (ucontrol->value.integer.value[0] == hs_switch) - return 0; - - if (ucontrol->value.integer.value[0]) { - pr_debug("hs_set HS path\n"); - snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); - } else { - pr_debug("hs_set EP path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); - } - snd_soc_dapm_sync(&codec->dapm); - hs_switch = ucontrol->value.integer.value[0]; - - return 0; -} - -static void lo_enable_out_pins(struct snd_soc_codec *codec) -{ - snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR"); - snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR"); - snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); - snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); - if (hs_switch) { - snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); - } else { - snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); - } -} - -static int lo_get_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = lo_dac; - return 0; -} - -static int lo_set_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - if (ucontrol->value.integer.value[0] == lo_dac) - return 0; - - /* we dont want to work with last state of lineout so just enable all - * pins and then disable pins not required - */ - lo_enable_out_pins(codec); - switch (ucontrol->value.integer.value[0]) { - case 0: - pr_debug("set vibra path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT"); - snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0); - break; - - case 1: - pr_debug("set hs path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); - break; - - case 2: - pr_debug("set spkr path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44); - break; - - case 3: - pr_debug("set null path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66); - break; - } - snd_soc_dapm_sync(&codec->dapm); - lo_dac = ucontrol->value.integer.value[0]; - return 0; -} - -static const struct snd_kcontrol_new mfld_snd_controls[] = { - SOC_ENUM_EXT("Playback Switch", headset_enum, - headset_get_switch, headset_set_switch), - SOC_ENUM_EXT("Lineout Mux", lo_enum, - lo_get_switch, lo_set_switch), -}; - -static const struct snd_soc_dapm_widget mfld_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), - SND_SOC_DAPM_MIC("Mic", NULL), -}; - -static const struct snd_soc_dapm_route mfld_map[] = { - {"Headphones", NULL, "HPOUTR"}, - {"Headphones", NULL, "HPOUTL"}, - {"Mic", NULL, "AMIC1"}, -}; - -static void mfld_jack_check(unsigned int intr_status) -{ - struct mfld_jack_data jack_data; - - jack_data.mfld_jack = &mfld_jack; - jack_data.intr_id = intr_status; - - sn95031_jack_detection(&jack_data); - /* TODO: add american headset detection post gpiolib support */ -} - -static int mfld_init(struct snd_soc_pcm_runtime *runtime) -{ - struct snd_soc_codec *codec = runtime->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret_val; - - /* Add jack sense widgets */ - snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets)); - - /* Set up the map */ - snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map)); - - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Headphones"); - snd_soc_dapm_enable_pin(dapm, "Mic"); - - ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls, - ARRAY_SIZE(mfld_snd_controls)); - if (ret_val) { - pr_err("soc_add_controls failed %d", ret_val); - return ret_val; - } - /* default is earpiece pin, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "Headphones"); - /* default is lineout NC, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); - snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); - lo_dac = 3; - hs_switch = 0; - /* we dont use linein in this so set to NC */ - snd_soc_dapm_disable_pin(dapm, "LINEINL"); - snd_soc_dapm_disable_pin(dapm, "LINEINR"); - - /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1, &mfld_jack); - if (ret_val) { - pr_err("jack creation failed\n"); - return ret_val; - } - - ret_val = snd_soc_jack_add_pins(&mfld_jack, - ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); - if (ret_val) { - pr_err("adding jack pins failed\n"); - return ret_val; - } - ret_val = snd_soc_jack_add_zones(&mfld_jack, - ARRAY_SIZE(mfld_zones), mfld_zones); - if (ret_val) { - pr_err("adding jack zones failed\n"); - return ret_val; - } - - /* we want to check if anything is inserted at boot, - * so send a fake event to codec and it will read adc - * to find if anything is there or not */ - mfld_jack_check(MFLD_JACK_INSERT); - return ret_val; -} - -static struct snd_soc_dai_link mfld_msic_dailink[] = { - { - .name = "Medfield Headset", - .stream_name = "Headset", - .cpu_dai_name = "Headset-cpu-dai", - .codec_dai_name = "SN95031 Headset", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = mfld_init, - }, - { - .name = "Medfield Speaker", - .stream_name = "Speaker", - .cpu_dai_name = "Speaker-cpu-dai", - .codec_dai_name = "SN95031 Speaker", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Vibra", - .stream_name = "Vibra1", - .cpu_dai_name = "Vibra1-cpu-dai", - .codec_dai_name = "SN95031 Vibra1", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Haptics", - .stream_name = "Vibra2", - .cpu_dai_name = "Vibra2-cpu-dai", - .codec_dai_name = "SN95031 Vibra2", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Compress", - .stream_name = "Speaker", - .cpu_dai_name = "Compress-cpu-dai", - .codec_dai_name = "SN95031 Speaker", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, -}; - -/* SoC card */ -static struct snd_soc_card snd_soc_card_mfld = { - .name = "medfield_audio", - .owner = THIS_MODULE, - .dai_link = mfld_msic_dailink, - .num_links = ARRAY_SIZE(mfld_msic_dailink), -}; - -static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) -{ - struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; - - memcpy_fromio(&mc_private->interrupt_status, - ((void *)(mc_private->int_base)), - sizeof(u8)); - return IRQ_WAKE_THREAD; -} - -static irqreturn_t snd_mfld_jack_detection(int irq, void *data) -{ - struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; - - if (mfld_jack.codec == NULL) - return IRQ_HANDLED; - mfld_jack_check(mc_drv_ctx->interrupt_status); - - return IRQ_HANDLED; -} - -static int snd_mfld_mc_probe(struct platform_device *pdev) -{ - int ret_val = 0, irq; - struct mfld_mc_private *mc_drv_ctx; - struct resource *irq_mem; - - pr_debug("snd_mfld_mc_probe called\n"); - - /* retrive the irq number */ - irq = platform_get_irq(pdev, 0); - - /* audio interrupt base of SRAM location where - * interrupts are stored by System FW */ - mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC); - if (!mc_drv_ctx) { - pr_err("allocation failed\n"); - return -ENOMEM; - } - - irq_mem = platform_get_resource_byname( - pdev, IORESOURCE_MEM, "IRQ_BASE"); - if (!irq_mem) { - pr_err("no mem resource given\n"); - return -ENODEV; - } - mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start, - resource_size(irq_mem)); - if (!mc_drv_ctx->int_base) { - pr_err("Mapping of cache failed\n"); - return -ENOMEM; - } - /* register for interrupt */ - ret_val = devm_request_threaded_irq(&pdev->dev, irq, - snd_mfld_jack_intr_handler, - snd_mfld_jack_detection, - IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); - if (ret_val) { - pr_err("cannot register IRQ\n"); - return ret_val; - } - /* register the soc card */ - snd_soc_card_mfld.dev = &pdev->dev; - ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld); - if (ret_val) { - pr_debug("snd_soc_register_card failed %d\n", ret_val); - return ret_val; - } - platform_set_drvdata(pdev, mc_drv_ctx); - pr_debug("successfully exited probe\n"); - return 0; -} - -static struct platform_driver snd_mfld_mc_driver = { - .driver = { - .owner = THIS_MODULE, - .name = "msic_audio", - }, - .probe = snd_mfld_mc_probe, -}; - -module_platform_driver(snd_mfld_mc_driver); - -MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver"); -MODULE_AUTHOR("Vinod Koul "); -MODULE_AUTHOR("Harsha Priya "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:msic-audio"); diff --git a/sound/soc/mid-x86/sst_dsp.h b/sound/soc/mid-x86/sst_dsp.h deleted file mode 100644 index 0fce1de284f..00000000000 --- a/sound/soc/mid-x86/sst_dsp.h +++ /dev/null @@ -1,134 +0,0 @@ -#ifndef __SST_DSP_H__ -#define __SST_DSP_H__ -/* - * sst_dsp.h - Intel SST Driver for audio engine - * - * Copyright (C) 2008-12 Intel Corporation - * Authors: Vinod Koul - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -enum sst_codec_types { - /* AUDIO/MUSIC CODEC Type Definitions */ - SST_CODEC_TYPE_UNKNOWN = 0, - SST_CODEC_TYPE_PCM, /* Pass through Audio codec */ - SST_CODEC_TYPE_MP3, - SST_CODEC_TYPE_MP24, - SST_CODEC_TYPE_AAC, - SST_CODEC_TYPE_AACP, - SST_CODEC_TYPE_eAACP, -}; - -enum stream_type { - SST_STREAM_TYPE_NONE = 0, - SST_STREAM_TYPE_MUSIC = 1, -}; - -struct snd_pcm_params { - u16 codec; /* codec type */ - u8 num_chan; /* 1=Mono, 2=Stereo */ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u32 reserved; /* Bitrate in bits per second */ - u32 sfreq; /* Sampling rate in Hz */ - u8 use_offload_path; - u8 reserved2; - u16 reserved3; - u8 channel_map[8]; -} __packed; - -/* MP3 Music Parameters Message */ -struct snd_mp3_params { - u16 codec; - u8 num_chan; /* 1=Mono, 2=Stereo */ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u8 crc_check; /* crc_check - disable (0) or enable (1) */ - u8 reserved1; /* unused*/ - u16 reserved2; /* Unused */ -} __packed; - -#define AAC_BIT_STREAM_ADTS 0 -#define AAC_BIT_STREAM_ADIF 1 -#define AAC_BIT_STREAM_RAW 2 - -/* AAC Music Parameters Message */ -struct snd_aac_params { - u16 codec; - u8 num_chan; /* 1=Mono, 2=Stereo*/ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */ - u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */ - u16 reser2; - u32 externalsr; /*sampling rate of basic AAC raw bit stream*/ - u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/ - u8 reser1; - u16 reser3; -} __packed; - -/* WMA Music Parameters Message */ -struct snd_wma_params { - u16 codec; - u8 num_chan; /* 1=Mono, 2=Stereo */ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u32 brate; /* Use the hard coded value. */ - u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ - u32 channel_mask; /* Channel Mask */ - u16 format_tag; /* Format Tag */ - u16 block_align; /* packet size */ - u16 wma_encode_opt;/* Encoder option */ - u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB */ - u8 reserved; /* reserved */ -} __packed; - -/* Codec params struture */ -union snd_sst_codec_params { - struct snd_pcm_params pcm_params; - struct snd_mp3_params mp3_params; - struct snd_aac_params aac_params; - struct snd_wma_params wma_params; -} __packed; - -/* Address and size info of a frame buffer */ -struct sst_address_info { - u32 addr; /* Address at IA */ - u32 size; /* Size of the buffer */ -}; - -struct snd_sst_alloc_params_ext { - struct sst_address_info ring_buf_info[8]; - u8 sg_count; - u8 reserved; - u16 reserved2; - u32 frag_size; /*Number of samples after which period elapsed - message is sent valid only if path = 0*/ -} __packed; - -struct snd_sst_stream_params { - union snd_sst_codec_params uc; -} __packed; - -struct snd_sst_params { - u32 stream_id; - u8 codec; - u8 ops; - u8 stream_type; - u8 device_type; - struct snd_sst_stream_params sparams; - struct snd_sst_alloc_params_ext aparams; -}; - -#endif /* __SST_DSP_H__ */ diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c deleted file mode 100644 index b6b5eb698d3..00000000000 --- a/sound/soc/mid-x86/sst_platform.c +++ /dev/null @@ -1,735 +0,0 @@ -/* - * sst_platform.c - Intel MID Platform driver - * - * Copyright (C) 2010-2013 Intel Corp - * Author: Vinod Koul - * Author: Harsha Priya - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * - */ -#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt - -#include -#include -#include -#include -#include -#include -#include -#include -#include "sst_platform.h" - -static struct sst_device *sst; -static DEFINE_MUTEX(sst_lock); - -int sst_register_dsp(struct sst_device *dev) -{ - if (WARN_ON(!dev)) - return -EINVAL; - if (!try_module_get(dev->dev->driver->owner)) - return -ENODEV; - mutex_lock(&sst_lock); - if (sst) { - pr_err("we already have a device %s\n", sst->name); - module_put(dev->dev->driver->owner); - mutex_unlock(&sst_lock); - return -EEXIST; - } - pr_debug("registering device %s\n", dev->name); - sst = dev; - mutex_unlock(&sst_lock); - return 0; -} -EXPORT_SYMBOL_GPL(sst_register_dsp); - -int sst_unregister_dsp(struct sst_device *dev) -{ - if (WARN_ON(!dev)) - return -EINVAL; - if (dev != sst) - return -EINVAL; - - mutex_lock(&sst_lock); - - if (!sst) { - mutex_unlock(&sst_lock); - return -EIO; - } - - module_put(sst->dev->driver->owner); - pr_debug("unreg %s\n", sst->name); - sst = NULL; - mutex_unlock(&sst_lock); - return 0; -} -EXPORT_SYMBOL_GPL(sst_unregister_dsp); - -static struct snd_pcm_hardware sst_platform_pcm_hw = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_DOUBLE | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_MMAP| - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 | - SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 | - SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32), - .rates = (SNDRV_PCM_RATE_8000| - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000), - .rate_min = SST_MIN_RATE, - .rate_max = SST_MAX_RATE, - .channels_min = SST_MIN_CHANNEL, - .channels_max = SST_MAX_CHANNEL, - .buffer_bytes_max = SST_MAX_BUFFER, - .period_bytes_min = SST_MIN_PERIOD_BYTES, - .period_bytes_max = SST_MAX_PERIOD_BYTES, - .periods_min = SST_MIN_PERIODS, - .periods_max = SST_MAX_PERIODS, - .fifo_size = SST_FIFO_SIZE, -}; - -/* MFLD - MSIC */ -static struct snd_soc_dai_driver sst_platform_dai[] = { -{ - .name = "Headset-cpu-dai", - .id = 0, - .playback = { - .channels_min = SST_STEREO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, - .capture = { - .channels_min = 1, - .channels_max = 5, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Speaker-cpu-dai", - .id = 1, - .playback = { - .channels_min = SST_MONO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Vibra1-cpu-dai", - .id = 2, - .playback = { - .channels_min = SST_MONO, - .channels_max = SST_MONO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Vibra2-cpu-dai", - .id = 3, - .playback = { - .channels_min = SST_MONO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Compress-cpu-dai", - .compress_dai = 1, - .playback = { - .channels_min = SST_STEREO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -}; - -static const struct snd_soc_component_driver sst_component = { - .name = "sst", -}; - -/* helper functions */ -static inline void sst_set_stream_status(struct sst_runtime_stream *stream, - int state) -{ - unsigned long flags; - spin_lock_irqsave(&stream->status_lock, flags); - stream->stream_status = state; - spin_unlock_irqrestore(&stream->status_lock, flags); -} - -static inline int sst_get_stream_status(struct sst_runtime_stream *stream) -{ - int state; - unsigned long flags; - - spin_lock_irqsave(&stream->status_lock, flags); - state = stream->stream_status; - spin_unlock_irqrestore(&stream->status_lock, flags); - return state; -} - -static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct sst_pcm_params *param) -{ - - param->codec = SST_CODEC_TYPE_PCM; - param->num_chan = (u8) substream->runtime->channels; - param->pcm_wd_sz = substream->runtime->sample_bits; - param->reserved = 0; - param->sfreq = substream->runtime->rate; - param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); - param->period_count = substream->runtime->period_size; - param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); - pr_debug("period_cnt = %d\n", param->period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); -} - -static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream = - substream->runtime->private_data; - struct sst_pcm_params param = {0}; - struct sst_stream_params str_params = {0}; - int ret_val; - - /* set codec params and inform SST driver the same */ - sst_fill_pcm_params(substream, ¶m); - substream->runtime->dma_area = substream->dma_buffer.area; - str_params.sparams = param; - str_params.codec = param.codec; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.device_type = substream->pcm->device + 1; - pr_debug("Playbck stream,Device %d\n", - substream->pcm->device); - } else { - str_params.ops = STREAM_OPS_CAPTURE; - str_params.device_type = SND_SST_DEVICE_CAPTURE; - pr_debug("Capture stream,Device %d\n", - substream->pcm->device); - } - ret_val = stream->ops->open(&str_params); - pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); - if (ret_val < 0) - return ret_val; - - stream->stream_info.str_id = ret_val; - pr_debug("str id : %d\n", stream->stream_info.str_id); - return ret_val; -} - -static void sst_period_elapsed(void *mad_substream) -{ - struct snd_pcm_substream *substream = mad_substream; - struct sst_runtime_stream *stream; - int status; - - if (!substream || !substream->runtime) - return; - stream = substream->runtime->private_data; - if (!stream) - return; - status = sst_get_stream_status(stream); - if (status != SST_PLATFORM_RUNNING) - return; - snd_pcm_period_elapsed(substream); -} - -static int sst_platform_init_stream(struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream = - substream->runtime->private_data; - int ret_val; - - pr_debug("setting buffer ptr param\n"); - sst_set_stream_status(stream, SST_PLATFORM_INIT); - stream->stream_info.period_elapsed = sst_period_elapsed; - stream->stream_info.mad_substream = substream; - stream->stream_info.buffer_ptr = 0; - stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->ops->device_control( - SST_SND_STREAM_INIT, &stream->stream_info); - if (ret_val) - pr_err("control_set ret error %d\n", ret_val); - return ret_val; - -} -/* end -- helper functions */ - -static int sst_platform_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sst_runtime_stream *stream; - int ret_val; - - pr_debug("sst_platform_open called\n"); - - snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); - ret_val = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret_val < 0) - return ret_val; - - stream = kzalloc(sizeof(*stream), GFP_KERNEL); - if (!stream) - return -ENOMEM; - spin_lock_init(&stream->status_lock); - - /* get the sst ops */ - mutex_lock(&sst_lock); - if (!sst) { - pr_err("no device available to run\n"); - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; - } - if (!try_module_get(sst->dev->driver->owner)) { - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; - } - stream->ops = sst->ops; - mutex_unlock(&sst_lock); - - stream->stream_info.str_id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); - stream->stream_info.mad_substream = substream; - /* allocate memory for SST API set */ - runtime->private_data = stream; - - return 0; -} - -static int sst_platform_close(struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream; - int ret_val = 0, str_id; - - pr_debug("sst_platform_close called\n"); - stream = substream->runtime->private_data; - str_id = stream->stream_info.str_id; - if (str_id) - ret_val = stream->ops->close(str_id); - module_put(sst->dev->driver->owner); - kfree(stream); - return ret_val; -} - -static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream; - int ret_val = 0, str_id; - - pr_debug("sst_platform_pcm_prepare called\n"); - stream = substream->runtime->private_data; - str_id = stream->stream_info.str_id; - if (stream->stream_info.str_id) { - ret_val = stream->ops->device_control( - SST_SND_DROP, &str_id); - return ret_val; - } - - ret_val = sst_platform_alloc_stream(substream); - if (ret_val < 0) - return ret_val; - snprintf(substream->pcm->id, sizeof(substream->pcm->id), - "%d", stream->stream_info.str_id); - - ret_val = sst_platform_init_stream(substream); - if (ret_val) - return ret_val; - substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER; - return ret_val; -} - -static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - int ret_val = 0, str_id; - struct sst_runtime_stream *stream; - int str_cmd, status; - - pr_debug("sst_platform_pcm_trigger called\n"); - stream = substream->runtime->private_data; - str_id = stream->stream_info.str_id; - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - pr_debug("sst: Trigger Start\n"); - str_cmd = SST_SND_START; - status = SST_PLATFORM_RUNNING; - stream->stream_info.mad_substream = substream; - break; - case SNDRV_PCM_TRIGGER_STOP: - pr_debug("sst: in stop\n"); - str_cmd = SST_SND_DROP; - status = SST_PLATFORM_DROPPED; - break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - pr_debug("sst: in pause\n"); - str_cmd = SST_SND_PAUSE; - status = SST_PLATFORM_PAUSED; - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - pr_debug("sst: in pause release\n"); - str_cmd = SST_SND_RESUME; - status = SST_PLATFORM_RUNNING; - break; - default: - return -EINVAL; - } - ret_val = stream->ops->device_control(str_cmd, &str_id); - if (!ret_val) - sst_set_stream_status(stream, status); - - return ret_val; -} - - -static snd_pcm_uframes_t sst_platform_pcm_pointer - (struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream; - int ret_val, status; - struct pcm_stream_info *str_info; - - stream = substream->runtime->private_data; - status = sst_get_stream_status(stream); - if (status == SST_PLATFORM_INIT) - return 0; - str_info = &stream->stream_info; - ret_val = stream->ops->device_control( - SST_SND_BUFFER_POINTER, str_info); - if (ret_val) { - pr_err("sst: error code = %d\n", ret_val); - return ret_val; - } - return stream->stream_info.buffer_ptr; -} - -static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - - return 0; -} - -static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static struct snd_pcm_ops sst_platform_ops = { - .open = sst_platform_open, - .close = sst_platform_close, - .ioctl = snd_pcm_lib_ioctl, - .prepare = sst_platform_pcm_prepare, - .trigger = sst_platform_pcm_trigger, - .pointer = sst_platform_pcm_pointer, - .hw_params = sst_platform_pcm_hw_params, - .hw_free = sst_platform_pcm_hw_free, -}; - -static void sst_pcm_free(struct snd_pcm *pcm) -{ - pr_debug("sst_pcm_free called\n"); - snd_pcm_lib_preallocate_free_for_all(pcm); -} - -static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_pcm *pcm = rtd->pcm; - int retval = 0; - - pr_debug("sst_pcm_new called\n"); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || - pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - retval = snd_pcm_lib_preallocate_pages_for_all(pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - SST_MIN_BUFFER, SST_MAX_BUFFER); - if (retval) { - pr_err("dma buffer allocationf fail\n"); - return retval; - } - } - return retval; -} - -/* compress stream operations */ -static void sst_compr_fragment_elapsed(void *arg) -{ - struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg; - - pr_debug("fragment elapsed by driver\n"); - if (cstream) - snd_compr_fragment_elapsed(cstream); -} - -static int sst_platform_compr_open(struct snd_compr_stream *cstream) -{ - - int ret_val = 0; - struct snd_compr_runtime *runtime = cstream->runtime; - struct sst_runtime_stream *stream; - - stream = kzalloc(sizeof(*stream), GFP_KERNEL); - if (!stream) - return -ENOMEM; - - spin_lock_init(&stream->status_lock); - - /* get the sst ops */ - if (!sst || !try_module_get(sst->dev->driver->owner)) { - pr_err("no device available to run\n"); - ret_val = -ENODEV; - goto out_ops; - } - stream->compr_ops = sst->compr_ops; - - stream->id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); - runtime->private_data = stream; - return 0; -out_ops: - kfree(stream); - return ret_val; -} - -static int sst_platform_compr_free(struct snd_compr_stream *cstream) -{ - struct sst_runtime_stream *stream; - int ret_val = 0, str_id; - - stream = cstream->runtime->private_data; - /*need to check*/ - str_id = stream->id; - if (str_id) - ret_val = stream->compr_ops->close(str_id); - module_put(sst->dev->driver->owner); - kfree(stream); - pr_debug("%s: %d\n", __func__, ret_val); - return 0; -} - -static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, - struct snd_compr_params *params) -{ - struct sst_runtime_stream *stream; - int retval; - struct snd_sst_params str_params; - struct sst_compress_cb cb; - - stream = cstream->runtime->private_data; - /* construct fw structure for this*/ - memset(&str_params, 0, sizeof(str_params)); - - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.stream_type = SST_STREAM_TYPE_MUSIC; - str_params.device_type = SND_SST_DEVICE_COMPRESS; - - switch (params->codec.id) { - case SND_AUDIOCODEC_MP3: { - str_params.codec = SST_CODEC_TYPE_MP3; - str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3; - str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in; - str_params.sparams.uc.mp3_params.pcm_wd_sz = 16; - break; - } - - case SND_AUDIOCODEC_AAC: { - str_params.codec = SST_CODEC_TYPE_AAC; - str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC; - str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in; - str_params.sparams.uc.aac_params.pcm_wd_sz = 16; - if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS) - str_params.sparams.uc.aac_params.bs_format = - AAC_BIT_STREAM_ADTS; - else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW) - str_params.sparams.uc.aac_params.bs_format = - AAC_BIT_STREAM_RAW; - else { - pr_err("Undefined format%d\n", params->codec.format); - return -EINVAL; - } - str_params.sparams.uc.aac_params.externalsr = - params->codec.sample_rate; - break; - } - - default: - pr_err("codec not supported, id =%d\n", params->codec.id); - return -EINVAL; - } - - str_params.aparams.ring_buf_info[0].addr = - virt_to_phys(cstream->runtime->buffer); - str_params.aparams.ring_buf_info[0].size = - cstream->runtime->buffer_size; - str_params.aparams.sg_count = 1; - str_params.aparams.frag_size = cstream->runtime->fragment_size; - - cb.param = cstream; - cb.compr_cb = sst_compr_fragment_elapsed; - - retval = stream->compr_ops->open(&str_params, &cb); - if (retval < 0) { - pr_err("stream allocation failed %d\n", retval); - return retval; - } - - stream->id = retval; - return 0; -} - -static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->control(cmd, stream->id); -} - -static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, - struct snd_compr_tstamp *tstamp) -{ - struct sst_runtime_stream *stream; - - stream = cstream->runtime->private_data; - stream->compr_ops->tstamp(stream->id, tstamp); - tstamp->byte_offset = tstamp->copied_total % - (u32)cstream->runtime->buffer_size; - pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset); - return 0; -} - -static int sst_platform_compr_ack(struct snd_compr_stream *cstream, - size_t bytes) -{ - struct sst_runtime_stream *stream; - - stream = cstream->runtime->private_data; - stream->compr_ops->ack(stream->id, (unsigned long)bytes); - stream->bytes_written += bytes; - - return 0; -} - -static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream, - struct snd_compr_caps *caps) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->get_caps(caps); -} - -static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, - struct snd_compr_codec_caps *codec) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->get_codec_caps(codec); -} - -static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, - struct snd_compr_metadata *metadata) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->set_metadata(stream->id, metadata); -} - -static struct snd_compr_ops sst_platform_compr_ops = { - - .open = sst_platform_compr_open, - .free = sst_platform_compr_free, - .set_params = sst_platform_compr_set_params, - .set_metadata = sst_platform_compr_set_metadata, - .trigger = sst_platform_compr_trigger, - .pointer = sst_platform_compr_pointer, - .ack = sst_platform_compr_ack, - .get_caps = sst_platform_compr_get_caps, - .get_codec_caps = sst_platform_compr_get_codec_caps, -}; - -static struct snd_soc_platform_driver sst_soc_platform_drv = { - .ops = &sst_platform_ops, - .compr_ops = &sst_platform_compr_ops, - .pcm_new = sst_pcm_new, - .pcm_free = sst_pcm_free, -}; - -static int sst_platform_probe(struct platform_device *pdev) -{ - int ret; - - pr_debug("sst_platform_probe called\n"); - sst = NULL; - ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); - if (ret) { - pr_err("registering soc platform failed\n"); - return ret; - } - - ret = snd_soc_register_component(&pdev->dev, &sst_component, - sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); - if (ret) { - pr_err("registering cpu dais failed\n"); - snd_soc_unregister_platform(&pdev->dev); - } - return ret; -} - -static int sst_platform_remove(struct platform_device *pdev) -{ - - snd_soc_unregister_component(&pdev->dev); - snd_soc_unregister_platform(&pdev->dev); - pr_debug("sst_platform_remove success\n"); - return 0; -} - -static struct platform_driver sst_platform_driver = { - .driver = { - .name = "sst-platform", - .owner = THIS_MODULE, - }, - .probe = sst_platform_probe, - .remove = sst_platform_remove, -}; - -module_platform_driver(sst_platform_driver); - -MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); -MODULE_AUTHOR("Vinod Koul "); -MODULE_AUTHOR("Harsha Priya "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:sst-platform"); diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h deleted file mode 100644 index cacc9066ec5..00000000000 --- a/sound/soc/mid-x86/sst_platform.h +++ /dev/null @@ -1,157 +0,0 @@ -/* - * sst_platform.h - Intel MID Platform driver header file - * - * Copyright (C) 2010 Intel Corp - * Author: Vinod Koul - * Author: Harsha Priya - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * - */ - -#ifndef __SST_PLATFORMDRV_H__ -#define __SST_PLATFORMDRV_H__ - -#include "sst_dsp.h" - -#define SST_MONO 1 -#define SST_STEREO 2 -#define SST_MAX_CAP 5 - -#define SST_MIN_RATE 8000 -#define SST_MAX_RATE 48000 -#define SST_MIN_CHANNEL 1 -#define SST_MAX_CHANNEL 5 -#define SST_MAX_BUFFER (800*1024) -#define SST_MIN_BUFFER (800*1024) -#define SST_MIN_PERIOD_BYTES 32 -#define SST_MAX_PERIOD_BYTES SST_MAX_BUFFER -#define SST_MIN_PERIODS 2 -#define SST_MAX_PERIODS (1024*2) -#define SST_FIFO_SIZE 0 - -struct pcm_stream_info { - int str_id; - void *mad_substream; - void (*period_elapsed) (void *mad_substream); - unsigned long long buffer_ptr; - int sfreq; -}; - -enum sst_drv_status { - SST_PLATFORM_INIT = 1, - SST_PLATFORM_STARTED, - SST_PLATFORM_RUNNING, - SST_PLATFORM_PAUSED, - SST_PLATFORM_DROPPED, -}; - -enum sst_controls { - SST_SND_ALLOC = 0x00, - SST_SND_PAUSE = 0x01, - SST_SND_RESUME = 0x02, - SST_SND_DROP = 0x03, - SST_SND_FREE = 0x04, - SST_SND_BUFFER_POINTER = 0x05, - SST_SND_STREAM_INIT = 0x06, - SST_SND_START = 0x07, - SST_MAX_CONTROLS = 0x07, -}; - -enum sst_stream_ops { - STREAM_OPS_PLAYBACK = 0, - STREAM_OPS_CAPTURE, -}; - -enum sst_audio_device_type { - SND_SST_DEVICE_HEADSET = 1, - SND_SST_DEVICE_IHF, - SND_SST_DEVICE_VIBRA, - SND_SST_DEVICE_HAPTIC, - SND_SST_DEVICE_CAPTURE, - SND_SST_DEVICE_COMPRESS, -}; - -/* PCM Parameters */ -struct sst_pcm_params { - u16 codec; /* codec type */ - u8 num_chan; /* 1=Mono, 2=Stereo */ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u32 reserved; /* Bitrate in bits per second */ - u32 sfreq; /* Sampling rate in Hz */ - u32 ring_buffer_size; - u32 period_count; /* period elapsed in samples*/ - u32 ring_buffer_addr; -}; - -struct sst_stream_params { - u32 result; - u32 stream_id; - u8 codec; - u8 ops; - u8 stream_type; - u8 device_type; - struct sst_pcm_params sparams; -}; - -struct sst_compress_cb { - void *param; - void (*compr_cb)(void *param); -}; - -struct compress_sst_ops { - const char *name; - int (*open) (struct snd_sst_params *str_params, - struct sst_compress_cb *cb); - int (*control) (unsigned int cmd, unsigned int str_id); - int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp); - int (*ack) (unsigned int str_id, unsigned long bytes); - int (*close) (unsigned int str_id); - int (*get_caps) (struct snd_compr_caps *caps); - int (*get_codec_caps) (struct snd_compr_codec_caps *codec); - int (*set_metadata) (unsigned int str_id, - struct snd_compr_metadata *mdata); - -}; - -struct sst_ops { - int (*open) (struct sst_stream_params *str_param); - int (*device_control) (int cmd, void *arg); - int (*close) (unsigned int str_id); -}; - -struct sst_runtime_stream { - int stream_status; - unsigned int id; - size_t bytes_written; - struct pcm_stream_info stream_info; - struct sst_ops *ops; - struct compress_sst_ops *compr_ops; - spinlock_t status_lock; -}; - -struct sst_device { - char *name; - struct device *dev; - struct sst_ops *ops; - struct compress_sst_ops *compr_ops; -}; - -int sst_register_dsp(struct sst_device *sst); -int sst_unregister_dsp(struct sst_device *sst); -#endif -- cgit v1.2.3-70-g09d2 From 8e6ad35a31e7ebc59543df875fc970200df2cf68 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:51 -0200 Subject: ASoC: wm8510: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8510.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 6ed5433943e..7df7d457275 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -684,7 +684,7 @@ static struct spi_driver wm8510_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8510_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -735,7 +735,7 @@ static struct i2c_driver wm8510_i2c_driver = { static int __init wm8510_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8510_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8510 I2C driver: %d\n", @@ -755,7 +755,7 @@ module_init(wm8510_modinit); static void __exit wm8510_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8510_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From 008ef947d0c5d14442256a37f6bf6b14015efe26 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:50 -0200 Subject: ASoC: wm8523: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8523.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 139bf9ac940..74d106dc766 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -452,7 +452,7 @@ static const struct regmap_config wm8523_regmap = { .volatile_reg = wm8523_volatile_register, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8523_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -555,7 +555,7 @@ static struct i2c_driver wm8523_i2c_driver = { static int __init wm8523_modinit(void) { int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8523_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8523 I2C driver: %d\n", @@ -568,7 +568,7 @@ module_init(wm8523_modinit); static void __exit wm8523_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8523_i2c_driver); #endif } -- cgit v1.2.3-70-g09d2 From f58c4fc4a3bf9eb699a638634d20ef4d16069366 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:49 -0200 Subject: ASoC: wm8580: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 08a414b57b1..318989acbbe 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -941,7 +941,7 @@ static const struct regmap_config wm8580_regmap = { .volatile_reg = wm8580_volatile, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1003,7 +1003,7 @@ static int __init wm8580_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8580_i2c_driver); if (ret != 0) { pr_err("Failed to register WM8580 I2C driver: %d\n", ret); @@ -1016,7 +1016,7 @@ module_init(wm8580_modinit); static void __exit wm8580_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8580_i2c_driver); #endif } -- cgit v1.2.3-70-g09d2 From 2309d6757900c4a6909fa673724976935b408a25 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:48 -0200 Subject: ASoC: wm8711: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 5b428b060d4..d99f948c513 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -469,7 +469,7 @@ static struct spi_driver wm8711_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8711_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -520,7 +520,7 @@ static struct i2c_driver wm8711_i2c_driver = { static int __init wm8711_modinit(void) { int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8711_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8711 I2C driver: %d\n", @@ -540,7 +540,7 @@ module_init(wm8711_modinit); static void __exit wm8711_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8711_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From 5c1537163ce716e317776565b8210ea06fa2b5de Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:47 -0200 Subject: ASoC: wm8728: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8728.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index c6a292dcded..cd89033e84c 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -320,7 +320,7 @@ static struct spi_driver wm8728_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8728_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -371,7 +371,7 @@ static struct i2c_driver wm8728_i2c_driver = { static int __init wm8728_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8728_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8728 I2C driver: %d\n", @@ -391,7 +391,7 @@ module_init(wm8728_modinit); static void __exit wm8728_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8728_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From 6435e5be652633b87f73460f9c17be361404ee01 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 8 Nov 2013 17:19:55 +0000 Subject: ASoC: wm8940: Convert to table based control and DAPM init Signed-off-by: Mark Brown Acked-by: Charles Keepax Acked-by: Jonathan Cameron --- sound/soc/codecs/wm8940.c | 28 +++++++--------------------- 1 file changed, 7 insertions(+), 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index b1591c61c25..4858b5c039f 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -264,7 +264,7 @@ static const struct snd_soc_dapm_widget wm8940_dapm_widgets[] = { SND_SOC_DAPM_INPUT("AUX"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8940_dapm_routes[] = { /* Mono output mixer */ {"Mono Mixer", "PCM Playback Switch", "DAC"}, {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, @@ -296,21 +296,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"ADC", NULL, "Boost Mixer"}, }; -static int wm8940_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - ret = snd_soc_dapm_new_controls(dapm, wm8940_dapm_widgets, - ARRAY_SIZE(wm8940_dapm_widgets)); - if (ret) - goto error_ret; - ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - -error_ret: - return ret; -} - #define wm8940_reset(c) snd_soc_write(c, WM8940_SOFTRESET, 0); static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai, @@ -716,11 +701,6 @@ static int wm8940_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_add_codec_controls(codec, wm8940_snd_controls, - ARRAY_SIZE(wm8940_snd_controls)); - if (ret) - return ret; - ret = wm8940_add_widgets(codec); return ret; } @@ -736,6 +716,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8940 = { .suspend = wm8940_suspend, .resume = wm8940_resume, .set_bias_level = wm8940_set_bias_level, + .controls = wm8940_snd_controls, + .num_controls = ARRAY_SIZE(wm8940_snd_controls), + .dapm_widgets = wm8940_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8940_dapm_widgets), + .dapm_routes = wm8940_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8940_dapm_routes), .reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8940_reg_defaults, -- cgit v1.2.3-70-g09d2 From fbbf7fea8e806ccc3ce0e059ad1d9671d57b4309 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 8 Nov 2013 17:22:23 +0000 Subject: ASoC: wm8940: Convert to direct regmap API usage This helps move us towards being able to remove the ASoC level I/O code. Signed-off-by: Mark Brown Acked-by: Charles Keepax Acked-by: Jonathan Cameron --- sound/soc/codecs/wm8940.c | 193 +++++++++++++++++++++++++++++----------------- 1 file changed, 121 insertions(+), 72 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 4858b5c039f..b404c26c175 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -28,7 +28,7 @@ #include #include #include -#include +#include #include #include #include @@ -41,78 +41,116 @@ struct wm8940_priv { unsigned int sysclk; - enum snd_soc_control_type control_type; + struct regmap *regmap; }; -static int wm8940_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm8940_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8940_SOFTRESET: - return 1; + return true; default: - return 0; + return false; + } +} + +static bool wm8940_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8940_SOFTRESET: + case WM8940_POWER1: + case WM8940_POWER2: + case WM8940_POWER3: + case WM8940_IFACE: + case WM8940_COMPANDINGCTL: + case WM8940_CLOCK: + case WM8940_ADDCNTRL: + case WM8940_GPIO: + case WM8940_CTLINT: + case WM8940_DAC: + case WM8940_DACVOL: + case WM8940_ADC: + case WM8940_ADCVOL: + case WM8940_NOTCH1: + case WM8940_NOTCH2: + case WM8940_NOTCH3: + case WM8940_NOTCH4: + case WM8940_NOTCH5: + case WM8940_NOTCH6: + case WM8940_NOTCH7: + case WM8940_NOTCH8: + case WM8940_DACLIM1: + case WM8940_DACLIM2: + case WM8940_ALC1: + case WM8940_ALC2: + case WM8940_ALC3: + case WM8940_NOISEGATE: + case WM8940_PLLN: + case WM8940_PLLK1: + case WM8940_PLLK2: + case WM8940_PLLK3: + case WM8940_ALC4: + case WM8940_INPUTCTL: + case WM8940_PGAGAIN: + case WM8940_ADCBOOST: + case WM8940_OUTPUTCTL: + case WM8940_SPKMIX: + case WM8940_SPKVOL: + case WM8940_MONOMIX: + return true; + default: + return false; } } -static u16 wm8940_reg_defaults[] = { - 0x8940, /* Soft Reset */ - 0x0000, /* Power 1 */ - 0x0000, /* Power 2 */ - 0x0000, /* Power 3 */ - 0x0010, /* Interface Control */ - 0x0000, /* Companding Control */ - 0x0140, /* Clock Control */ - 0x0000, /* Additional Controls */ - 0x0000, /* GPIO Control */ - 0x0002, /* Auto Increment Control */ - 0x0000, /* DAC Control */ - 0x00FF, /* DAC Volume */ - 0, - 0, - 0x0100, /* ADC Control */ - 0x00FF, /* ADC Volume */ - 0x0000, /* Notch Filter 1 Control 1 */ - 0x0000, /* Notch Filter 1 Control 2 */ - 0x0000, /* Notch Filter 2 Control 1 */ - 0x0000, /* Notch Filter 2 Control 2 */ - 0x0000, /* Notch Filter 3 Control 1 */ - 0x0000, /* Notch Filter 3 Control 2 */ - 0x0000, /* Notch Filter 4 Control 1 */ - 0x0000, /* Notch Filter 4 Control 2 */ - 0x0032, /* DAC Limit Control 1 */ - 0x0000, /* DAC Limit Control 2 */ - 0, - 0, - 0, - 0, - 0, - 0, - 0x0038, /* ALC Control 1 */ - 0x000B, /* ALC Control 2 */ - 0x0032, /* ALC Control 3 */ - 0x0000, /* Noise Gate */ - 0x0041, /* PLLN */ - 0x000C, /* PLLK1 */ - 0x0093, /* PLLK2 */ - 0x00E9, /* PLLK3 */ - 0, - 0, - 0x0030, /* ALC Control 4 */ - 0, - 0x0002, /* Input Control */ - 0x0050, /* PGA Gain */ - 0, - 0x0002, /* ADC Boost Control */ - 0, - 0x0002, /* Output Control */ - 0x0000, /* Speaker Mixer Control */ - 0, - 0, - 0, - 0x0079, /* Speaker Volume */ - 0, - 0x0000, /* Mono Mixer Control */ +static const struct reg_default wm8940_reg_defaults[] = { + { 0x1, 0x0000 }, /* Power 1 */ + { 0x2, 0x0000 }, /* Power 2 */ + { 0x3, 0x0000 }, /* Power 3 */ + { 0x4, 0x0010 }, /* Interface Control */ + { 0x5, 0x0000 }, /* Companding Control */ + { 0x6, 0x0140 }, /* Clock Control */ + { 0x7, 0x0000 }, /* Additional Controls */ + { 0x8, 0x0000 }, /* GPIO Control */ + { 0x9, 0x0002 }, /* Auto Increment Control */ + { 0xa, 0x0000 }, /* DAC Control */ + { 0xb, 0x00FF }, /* DAC Volume */ + + { 0xe, 0x0100 }, /* ADC Control */ + { 0xf, 0x00FF }, /* ADC Volume */ + { 0x10, 0x0000 }, /* Notch Filter 1 Control 1 */ + { 0x11, 0x0000 }, /* Notch Filter 1 Control 2 */ + { 0x12, 0x0000 }, /* Notch Filter 2 Control 1 */ + { 0x13, 0x0000 }, /* Notch Filter 2 Control 2 */ + { 0x14, 0x0000 }, /* Notch Filter 3 Control 1 */ + { 0x15, 0x0000 }, /* Notch Filter 3 Control 2 */ + { 0x16, 0x0000 }, /* Notch Filter 4 Control 1 */ + { 0x17, 0x0000 }, /* Notch Filter 4 Control 2 */ + { 0x18, 0x0032 }, /* DAC Limit Control 1 */ + { 0x19, 0x0000 }, /* DAC Limit Control 2 */ + + { 0x20, 0x0038 }, /* ALC Control 1 */ + { 0x21, 0x000B }, /* ALC Control 2 */ + { 0x22, 0x0032 }, /* ALC Control 3 */ + { 0x23, 0x0000 }, /* Noise Gate */ + { 0x24, 0x0041 }, /* PLLN */ + { 0x25, 0x000C }, /* PLLK1 */ + { 0x26, 0x0093 }, /* PLLK2 */ + { 0x27, 0x00E9 }, /* PLLK3 */ + + { 0x2a, 0x0030 }, /* ALC Control 4 */ + + { 0x2c, 0x0002 }, /* Input Control */ + { 0x2d, 0x0050 }, /* PGA Gain */ + + { 0x2f, 0x0002 }, /* ADC Boost Control */ + + { 0x31, 0x0002 }, /* Output Control */ + { 0x32, 0x0000 }, /* Speaker Mixer Control */ + + { 0x36, 0x0079 }, /* Speaker Volume */ + + { 0x38, 0x0000 }, /* Mono Mixer Control */ }; static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" }; @@ -431,6 +469,7 @@ static int wm8940_mute(struct snd_soc_dai *dai, int mute) static int wm8940_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8940_priv *wm8940 = snd_soc_codec_get_drvdata(codec); u16 val; u16 pwr_reg = snd_soc_read(codec, WM8940_POWER1) & 0x1F0; int ret = 0; @@ -454,7 +493,7 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(wm8940->regmap); if (ret < 0) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); return ret; @@ -669,12 +708,11 @@ static int wm8940_resume(struct snd_soc_codec *codec) static int wm8940_probe(struct snd_soc_codec *codec) { - struct wm8940_priv *wm8940 = snd_soc_codec_get_drvdata(codec); struct wm8940_setup_data *pdata = codec->dev->platform_data; int ret; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8940->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -722,10 +760,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8940 = { .num_dapm_widgets = ARRAY_SIZE(wm8940_dapm_widgets), .dapm_routes = wm8940_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm8940_dapm_routes), - .reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8940_reg_defaults, - .volatile_register = wm8940_volatile_register, +}; + +static const struct regmap_config wm8940_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM8940_MONOMIX, + .reg_defaults = wm8940_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8940_reg_defaults), + + .readable_reg = wm8940_readable_register, + .volatile_reg = wm8940_volatile_register, }; static int wm8940_i2c_probe(struct i2c_client *i2c, @@ -739,8 +785,11 @@ static int wm8940_i2c_probe(struct i2c_client *i2c, if (wm8940 == NULL) return -ENOMEM; + wm8940->regmap = devm_regmap_init_i2c(i2c, &wm8940_regmap); + if (IS_ERR(wm8940->regmap)) + return PTR_ERR(wm8940->regmap); + i2c_set_clientdata(i2c, wm8940); - wm8940->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8940, &wm8940_dai, 1); -- cgit v1.2.3-70-g09d2 From c3e8494c001ce0bec0ebaa49c6f5eeb2aa5ab36a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:41 -0200 Subject: ASoC: wm8962: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 543c5c2631b..07da601f834 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -74,7 +74,7 @@ struct wm8962_priv { struct regulator_bulk_data supplies[WM8962_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8962_NUM_SUPPLIES]; -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; @@ -3108,7 +3108,7 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(wm8962_mic_detect); -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) static int beep_rates[] = { 500, 1000, 2000, 4000, }; -- cgit v1.2.3-70-g09d2 From e40e0b5da87bb4256a6dc62db7663b8a0c204f1f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 8 Nov 2013 14:01:39 +0000 Subject: ASoC: wm8974: Convert to direct regmap API usage Moves us towards removing the ASoC level I/O functions. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8974.c | 54 +++++++++++++++++++++++++++++------------------ 1 file changed, 33 insertions(+), 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index a2d01d10a5d..15f45c7bd83 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -27,22 +28,22 @@ #include "wm8974.h" -static const u16 wm8974_reg[WM8974_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0050, 0x0000, 0x0140, 0x0000, - 0x0000, 0x0000, 0x0000, 0x00ff, - 0x0000, 0x0000, 0x0100, 0x00ff, - 0x0000, 0x0000, 0x012c, 0x002c, - 0x002c, 0x002c, 0x002c, 0x0000, - 0x0032, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0038, 0x000b, 0x0032, 0x0000, - 0x0008, 0x000c, 0x0093, 0x00e9, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0003, 0x0010, 0x0000, 0x0000, - 0x0000, 0x0002, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0039, 0x0000, - 0x0000, +static const struct reg_default wm8974_reg_defaults[] = { + { 0, 0x0000 }, { 1, 0x0000 }, { 2, 0x0000 }, { 3, 0x0000 }, + { 4, 0x0050 }, { 5, 0x0000 }, { 6, 0x0140 }, { 7, 0x0000 }, + { 8, 0x0000 }, { 9, 0x0000 }, { 10, 0x0000 }, { 11, 0x00ff }, + { 12, 0x0000 }, { 13, 0x0000 }, { 14, 0x0100 }, { 15, 0x00ff }, + { 16, 0x0000 }, { 17, 0x0000 }, { 18, 0x012c }, { 19, 0x002c }, + { 20, 0x002c }, { 21, 0x002c }, { 22, 0x002c }, { 23, 0x0000 }, + { 24, 0x0032 }, { 25, 0x0000 }, { 26, 0x0000 }, { 27, 0x0000 }, + { 28, 0x0000 }, { 29, 0x0000 }, { 30, 0x0000 }, { 31, 0x0000 }, + { 32, 0x0038 }, { 33, 0x000b }, { 34, 0x0032 }, { 35, 0x0000 }, + { 36, 0x0008 }, { 37, 0x000c }, { 38, 0x0093 }, { 39, 0x00e9 }, + { 40, 0x0000 }, { 41, 0x0000 }, { 42, 0x0000 }, { 43, 0x0000 }, + { 44, 0x0003 }, { 45, 0x0010 }, { 46, 0x0000 }, { 47, 0x0000 }, + { 48, 0x0000 }, { 49, 0x0002 }, { 50, 0x0000 }, { 51, 0x0000 }, + { 52, 0x0000 }, { 53, 0x0000 }, { 54, 0x0039 }, { 55, 0x0000 }, + { 56, 0x0000 }, }; #define WM8974_POWER1_BIASEN 0x08 @@ -514,7 +515,7 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec, power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN; if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(dev_get_regmap(codec->dev, NULL)); /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8974_POWER1, power1 | 0x3); @@ -579,11 +580,20 @@ static int wm8974_resume(struct snd_soc_codec *codec) return 0; } +static const struct regmap_config wm8974_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = WM8974_MONOMIX, + .reg_defaults = wm8974_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8974_reg_defaults), +}; + static int wm8974_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -613,9 +623,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8974 = { .suspend = wm8974_suspend, .resume = wm8974_resume, .set_bias_level = wm8974_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8974_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8974_reg, .controls = wm8974_snd_controls, .num_controls = ARRAY_SIZE(wm8974_snd_controls), @@ -628,8 +635,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8974 = { static int wm8974_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct regmap *regmap; int ret; + regmap = devm_regmap_init_i2c(i2c, &wm8974_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8974, &wm8974_dai, 1); -- cgit v1.2.3-70-g09d2 From 6a0773368619644432fbe46f64ded6aa204baab9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 11:16:18 +0000 Subject: ASoC: wm8991: Use a supply to manage input power Instead of using a fake register and events to manage input power use a supply to do the job, saving code and preparing for regmap conversion of the driver. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8991.c | 53 +++++++++++++++-------------------------------- sound/soc/codecs/wm8991.h | 9 -------- 2 files changed, 17 insertions(+), 45 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 3a39df7a382..5078fc8e10f 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -374,30 +374,6 @@ static const struct snd_kcontrol_new wm8991_snd_controls[] = { /* * _DAPM_ Controls */ -static int inmixer_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - u16 reg, fakepower; - - reg = snd_soc_read(w->codec, WM8991_POWER_MANAGEMENT_2); - fakepower = snd_soc_read(w->codec, WM8991_INTDRIVBITS); - - if (fakepower & ((1 << WM8991_INMIXL_PWR_BIT) | - (1 << WM8991_AINLMUX_PWR_BIT))) - reg |= WM8991_AINL_ENA; - else - reg &= ~WM8991_AINL_ENA; - - if (fakepower & ((1 << WM8991_INMIXR_PWR_BIT) | - (1 << WM8991_AINRMUX_PWR_BIT))) - reg |= WM8991_AINR_ENA; - else - reg &= ~WM8991_AINR_ENA; - - snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg); - return 0; -} - static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -655,6 +631,11 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RIN2"), SND_SOC_DAPM_INPUT("Internal ADC Source"), + SND_SOC_DAPM_SUPPLY("INL", WM8991_POWER_MANAGEMENT_2, + WM8991_AINL_ENA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("INR", WM8991_POWER_MANAGEMENT_2, + WM8991_AINR_ENA_BIT, 0, NULL, 0), + /* DACs */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8991_POWER_MANAGEMENT_2, WM8991_ADCL_ENA_BIT, 0), @@ -676,26 +657,22 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { ARRAY_SIZE(wm8991_dapm_rin34_pga_controls)), /* INMIXL */ - SND_SOC_DAPM_MIXER_E("INMIXL", WM8991_INTDRIVBITS, WM8991_INMIXL_PWR_BIT, 0, + SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0, &wm8991_dapm_inmixl_controls[0], - ARRAY_SIZE(wm8991_dapm_inmixl_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8991_dapm_inmixl_controls)), /* AINLMUX */ - SND_SOC_DAPM_MUX_E("AINLMUX", WM8991_INTDRIVBITS, WM8991_AINLMUX_PWR_BIT, 0, - &wm8991_dapm_ainlmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("AINLMUX", SND_SOC_NOPM, 0, 0, + &wm8991_dapm_ainlmux_controls), /* INMIXR */ - SND_SOC_DAPM_MIXER_E("INMIXR", WM8991_INTDRIVBITS, WM8991_INMIXR_PWR_BIT, 0, + SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0, &wm8991_dapm_inmixr_controls[0], - ARRAY_SIZE(wm8991_dapm_inmixr_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8991_dapm_inmixr_controls)), /* AINRMUX */ - SND_SOC_DAPM_MUX_E("AINRMUX", WM8991_INTDRIVBITS, WM8991_AINRMUX_PWR_BIT, 0, - &wm8991_dapm_ainrmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("AINRMUX", SND_SOC_NOPM, 0, 0, + &wm8991_dapm_ainrmux_controls), /* Output Side */ /* DACs */ @@ -797,6 +774,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Right ADC", NULL, "Internal ADC Source"}, /* Input Side */ + {"INMIXL", NULL, "INL"}, + {"AINLMUX", NULL, "INL"}, + {"INMIXR", NULL, "INR"}, + {"AINRMUX", NULL, "INR"}, /* LIN12 PGA */ {"LIN12 PGA", "LIN1 Switch", "LIN1"}, {"LIN12 PGA", "LIN2 Switch", "LIN2"}, diff --git a/sound/soc/codecs/wm8991.h b/sound/soc/codecs/wm8991.h index 07707d8d7e2..08ed383303c 100644 --- a/sound/soc/codecs/wm8991.h +++ b/sound/soc/codecs/wm8991.h @@ -76,7 +76,6 @@ #define WM8991_PLL1 0x3C #define WM8991_PLL2 0x3D #define WM8991_PLL3 0x3E -#define WM8991_INTDRIVBITS 0x3F #define WM8991_REGISTER_COUNT 60 #define WM8991_MAX_REGISTER 0x3F @@ -807,14 +806,6 @@ */ #define WM8991_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */ -/* - * R63 (0x3F) - Internal Driver Bits - */ -#define WM8991_INMIXL_PWR_BIT 0 -#define WM8991_AINLMUX_PWR_BIT 1 -#define WM8991_INMIXR_PWR_BIT 2 -#define WM8991_AINRMUX_PWR_BIT 3 - #define WM8991_MCLK_DIV 0 #define WM8991_DACCLK_DIV 1 #define WM8991_ADCCLK_DIV 2 -- cgit v1.2.3-70-g09d2 From 898249958281ebf52288eb0d3ed6797e1fc4b0bd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 12:28:06 +0000 Subject: ASoC: wm8991: Convert to table based control and widget init Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8991.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 5078fc8e10f..86aa33fd11d 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -764,7 +764,7 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8991_dapm_routes[] = { /* Make DACs turn on when playing even if not mixed into any outputs */ {"Internal DAC Sink", NULL, "Left DAC"}, {"Internal DAC Sink", NULL, "Right DAC"}, @@ -1278,13 +1278,6 @@ static int wm8991_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - snd_soc_add_codec_controls(codec, wm8991_snd_controls, - ARRAY_SIZE(wm8991_snd_controls)); - - snd_soc_dapm_new_controls(&codec->dapm, wm8991_dapm_widgets, - ARRAY_SIZE(wm8991_dapm_widgets)); - snd_soc_dapm_add_routes(&codec->dapm, audio_map, - ARRAY_SIZE(audio_map)); return 0; } @@ -1333,6 +1326,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8991 = { .suspend = wm8991_suspend, .resume = wm8991_resume, .set_bias_level = wm8991_set_bias_level, + .controls = wm8991_snd_controls, + .num_controls = ARRAY_SIZE(wm8991_snd_controls), + .dapm_widgets = wm8991_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8991_dapm_widgets), + .dapm_routes = wm8991_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8991_dapm_routes), .reg_cache_size = WM8991_MAX_REGISTER + 1, .reg_word_size = sizeof(u16), .reg_cache_default = wm8991_reg_defs -- cgit v1.2.3-70-g09d2 From a86652e51a8776bc0fe811e32ec3118f03c7e3bb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 13:28:55 +0000 Subject: ASoC: wm8991: Convert to direct regmap API usage Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8991.c | 166 ++++++++++++++++++++++++++-------------------- 1 file changed, 94 insertions(+), 72 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 86aa33fd11d..7006f9732af 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -31,76 +32,85 @@ #include "wm8991.h" struct wm8991_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; unsigned int pcmclk; }; -static const u16 wm8991_reg_defs[] = { - 0x8991, /* R0 - Reset */ - 0x0000, /* R1 - Power Management (1) */ - 0x6000, /* R2 - Power Management (2) */ - 0x0000, /* R3 - Power Management (3) */ - 0x4050, /* R4 - Audio Interface (1) */ - 0x4000, /* R5 - Audio Interface (2) */ - 0x01C8, /* R6 - Clocking (1) */ - 0x0000, /* R7 - Clocking (2) */ - 0x0040, /* R8 - Audio Interface (3) */ - 0x0040, /* R9 - Audio Interface (4) */ - 0x0004, /* R10 - DAC CTRL */ - 0x00C0, /* R11 - Left DAC Digital Volume */ - 0x00C0, /* R12 - Right DAC Digital Volume */ - 0x0000, /* R13 - Digital Side Tone */ - 0x0100, /* R14 - ADC CTRL */ - 0x00C0, /* R15 - Left ADC Digital Volume */ - 0x00C0, /* R16 - Right ADC Digital Volume */ - 0x0000, /* R17 */ - 0x0000, /* R18 - GPIO CTRL 1 */ - 0x1000, /* R19 - GPIO1 & GPIO2 */ - 0x1010, /* R20 - GPIO3 & GPIO4 */ - 0x1010, /* R21 - GPIO5 & GPIO6 */ - 0x8000, /* R22 - GPIOCTRL 2 */ - 0x0800, /* R23 - GPIO_POL */ - 0x008B, /* R24 - Left Line Input 1&2 Volume */ - 0x008B, /* R25 - Left Line Input 3&4 Volume */ - 0x008B, /* R26 - Right Line Input 1&2 Volume */ - 0x008B, /* R27 - Right Line Input 3&4 Volume */ - 0x0000, /* R28 - Left Output Volume */ - 0x0000, /* R29 - Right Output Volume */ - 0x0066, /* R30 - Line Outputs Volume */ - 0x0022, /* R31 - Out3/4 Volume */ - 0x0079, /* R32 - Left OPGA Volume */ - 0x0079, /* R33 - Right OPGA Volume */ - 0x0003, /* R34 - Speaker Volume */ - 0x0003, /* R35 - ClassD1 */ - 0x0000, /* R36 */ - 0x0100, /* R37 - ClassD3 */ - 0x0000, /* R38 */ - 0x0000, /* R39 - Input Mixer1 */ - 0x0000, /* R40 - Input Mixer2 */ - 0x0000, /* R41 - Input Mixer3 */ - 0x0000, /* R42 - Input Mixer4 */ - 0x0000, /* R43 - Input Mixer5 */ - 0x0000, /* R44 - Input Mixer6 */ - 0x0000, /* R45 - Output Mixer1 */ - 0x0000, /* R46 - Output Mixer2 */ - 0x0000, /* R47 - Output Mixer3 */ - 0x0000, /* R48 - Output Mixer4 */ - 0x0000, /* R49 - Output Mixer5 */ - 0x0000, /* R50 - Output Mixer6 */ - 0x0180, /* R51 - Out3/4 Mixer */ - 0x0000, /* R52 - Line Mixer1 */ - 0x0000, /* R53 - Line Mixer2 */ - 0x0000, /* R54 - Speaker Mixer */ - 0x0000, /* R55 - Additional Control */ - 0x0000, /* R56 - AntiPOP1 */ - 0x0000, /* R57 - AntiPOP2 */ - 0x0000, /* R58 - MICBIAS */ - 0x0000, /* R59 */ - 0x0008, /* R60 - PLL1 */ - 0x0031, /* R61 - PLL2 */ - 0x0026, /* R62 - PLL3 */ +static const struct reg_default wm8991_reg_defaults[] = { + { 1, 0x0000 }, /* R1 - Power Management (1) */ + { 2, 0x6000 }, /* R2 - Power Management (2) */ + { 3, 0x0000 }, /* R3 - Power Management (3) */ + { 4, 0x4050 }, /* R4 - Audio Interface (1) */ + { 5, 0x4000 }, /* R5 - Audio Interface (2) */ + { 6, 0x01C8 }, /* R6 - Clocking (1) */ + { 7, 0x0000 }, /* R7 - Clocking (2) */ + { 8, 0x0040 }, /* R8 - Audio Interface (3) */ + { 9, 0x0040 }, /* R9 - Audio Interface (4) */ + { 10, 0x0004 }, /* R10 - DAC CTRL */ + { 11, 0x00C0 }, /* R11 - Left DAC Digital Volume */ + { 12, 0x00C0 }, /* R12 - Right DAC Digital Volume */ + { 13, 0x0000 }, /* R13 - Digital Side Tone */ + { 14, 0x0100 }, /* R14 - ADC CTRL */ + { 15, 0x00C0 }, /* R15 - Left ADC Digital Volume */ + { 16, 0x00C0 }, /* R16 - Right ADC Digital Volume */ + + { 18, 0x0000 }, /* R18 - GPIO CTRL 1 */ + { 19, 0x1000 }, /* R19 - GPIO1 & GPIO2 */ + { 20, 0x1010 }, /* R20 - GPIO3 & GPIO4 */ + { 21, 0x1010 }, /* R21 - GPIO5 & GPIO6 */ + { 22, 0x8000 }, /* R22 - GPIOCTRL 2 */ + { 23, 0x0800 }, /* R23 - GPIO_POL */ + { 24, 0x008B }, /* R24 - Left Line Input 1&2 Volume */ + { 25, 0x008B }, /* R25 - Left Line Input 3&4 Volume */ + { 26, 0x008B }, /* R26 - Right Line Input 1&2 Volume */ + { 27, 0x008B }, /* R27 - Right Line Input 3&4 Volume */ + { 28, 0x0000 }, /* R28 - Left Output Volume */ + { 29, 0x0000 }, /* R29 - Right Output Volume */ + { 30, 0x0066 }, /* R30 - Line Outputs Volume */ + { 31, 0x0022 }, /* R31 - Out3/4 Volume */ + { 32, 0x0079 }, /* R32 - Left OPGA Volume */ + { 33, 0x0079 }, /* R33 - Right OPGA Volume */ + { 34, 0x0003 }, /* R34 - Speaker Volume */ + { 35, 0x0003 }, /* R35 - ClassD1 */ + + { 37, 0x0100 }, /* R37 - ClassD3 */ + + { 39, 0x0000 }, /* R39 - Input Mixer1 */ + { 40, 0x0000 }, /* R40 - Input Mixer2 */ + { 41, 0x0000 }, /* R41 - Input Mixer3 */ + { 42, 0x0000 }, /* R42 - Input Mixer4 */ + { 43, 0x0000 }, /* R43 - Input Mixer5 */ + { 44, 0x0000 }, /* R44 - Input Mixer6 */ + { 45, 0x0000 }, /* R45 - Output Mixer1 */ + { 46, 0x0000 }, /* R46 - Output Mixer2 */ + { 47, 0x0000 }, /* R47 - Output Mixer3 */ + { 48, 0x0000 }, /* R48 - Output Mixer4 */ + { 49, 0x0000 }, /* R49 - Output Mixer5 */ + { 50, 0x0000 }, /* R50 - Output Mixer6 */ + { 51, 0x0180 }, /* R51 - Out3/4 Mixer */ + { 52, 0x0000 }, /* R52 - Line Mixer1 */ + { 53, 0x0000 }, /* R53 - Line Mixer2 */ + { 54, 0x0000 }, /* R54 - Speaker Mixer */ + { 55, 0x0000 }, /* R55 - Additional Control */ + { 56, 0x0000 }, /* R56 - AntiPOP1 */ + { 57, 0x0000 }, /* R57 - AntiPOP2 */ + { 58, 0x0000 }, /* R58 - MICBIAS */ + + { 60, 0x0008 }, /* R60 - PLL1 */ + { 61, 0x0031 }, /* R61 - PLL2 */ + { 62, 0x0026 }, /* R62 - PLL3 */ }; +static bool wm8991_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8991_RESET: + return true; + default: + return false; + } +} + #define wm8991_reset(c) snd_soc_write(c, WM8991_RESET, 0) static const unsigned int rec_mix_tlv[] = { @@ -1110,6 +1120,7 @@ static int wm8991_mute(struct snd_soc_dai *dai, int mute) static int wm8991_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8991_priv *wm8991 = snd_soc_codec_get_drvdata(codec); u16 val; switch (level) { @@ -1125,7 +1136,7 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8991->regmap); /* Enable all output discharge bits */ snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE | WM8991_DIS_RLINE | WM8991_DIS_OUT3 | @@ -1213,7 +1224,7 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ snd_soc_write(codec, WM8991_ANTIPOP2, 0x0); - codec->cache_sync = 1; + regcache_mark_dirty(wm8991->regmap); break; } @@ -1247,7 +1258,7 @@ static int wm8991_probe(struct snd_soc_codec *codec) wm8991 = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8991->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; @@ -1332,9 +1343,17 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8991 = { .num_dapm_widgets = ARRAY_SIZE(wm8991_dapm_widgets), .dapm_routes = wm8991_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm8991_dapm_routes), - .reg_cache_size = WM8991_MAX_REGISTER + 1, - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8991_reg_defs +}; + +static const struct regmap_config wm8991_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM8991_PLL3, + .volatile_reg = wm8991_volatile, + .reg_defaults = wm8991_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8991_reg_defaults), + .cache_type = REGCACHE_RBTREE, }; static int wm8991_i2c_probe(struct i2c_client *i2c, @@ -1347,7 +1366,10 @@ static int wm8991_i2c_probe(struct i2c_client *i2c, if (!wm8991) return -ENOMEM; - wm8991->control_type = SND_SOC_I2C; + wm8991->regmap = devm_regmap_init_i2c(i2c, &wm8991_regmap); + if (IS_ERR(wm8991->regmap)) + return PTR_ERR(wm8991->regmap); + i2c_set_clientdata(i2c, wm8991); ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.2.3-70-g09d2 From e4634804cacce6fe1ec34d92786f764fcb75cb97 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 13:32:36 +0000 Subject: ASoC: wm8991: Move basic initialisation to I2C level probe This is better practice, though some of this stuff ought not to be here at all. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8991.c | 50 +++++++++++++++++++++++------------------------ 1 file changed, 25 insertions(+), 25 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 7006f9732af..5fdcf788e99 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -111,8 +111,6 @@ static bool wm8991_volatile(struct device *dev, unsigned int reg) } } -#define wm8991_reset(c) snd_soc_write(c, WM8991_RESET, 0) - static const unsigned int rec_mix_tlv[] = { TLV_DB_RANGE_HEAD(1), 0, 7, TLV_DB_LINEAR_ITEM(-1500, 600), @@ -1264,31 +1262,8 @@ static int wm8991_probe(struct snd_soc_codec *codec) return ret; } - ret = wm8991_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - return ret; - } - wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_update_bits(codec, WM8991_AUDIO_INTERFACE_4, - WM8991_ALRCGPIO1, WM8991_ALRCGPIO1); - - snd_soc_update_bits(codec, WM8991_GPIO1_GPIO2, - WM8991_GPIO1_SEL_MASK, 1); - - snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_1, - WM8991_VREF_ENA | WM8991_VMID_MODE_MASK, - WM8991_VREF_ENA | WM8991_VMID_MODE_MASK); - - snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_2, - WM8991_OPCLK_ENA, WM8991_OPCLK_ENA); - - snd_soc_write(codec, WM8991_DAC_CTRL, 0); - snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); - snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - return 0; } @@ -1372,6 +1347,31 @@ static int wm8991_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8991); + ret = regmap_write(wm8991->regmap, WM8991_RESET, 0); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + return ret; + } + + regmap_update_bits(wm8991->regmap, WM8991_AUDIO_INTERFACE_4, + WM8991_ALRCGPIO1, WM8991_ALRCGPIO1); + + regmap_update_bits(wm8991->regmap, WM8991_GPIO1_GPIO2, + WM8991_GPIO1_SEL_MASK, 1); + + regmap_update_bits(wm8991->regmap, WM8991_POWER_MANAGEMENT_1, + WM8991_VREF_ENA | WM8991_VMID_MODE_MASK, + WM8991_VREF_ENA | WM8991_VMID_MODE_MASK); + + regmap_update_bits(wm8991->regmap, WM8991_POWER_MANAGEMENT_2, + WM8991_OPCLK_ENA, WM8991_OPCLK_ENA); + + regmap_write(wm8991->regmap, WM8991_DAC_CTRL, 0); + regmap_write(wm8991->regmap, WM8991_LEFT_OUTPUT_VOLUME, + 0x50 | (1<<8)); + regmap_write(wm8991->regmap, WM8991_RIGHT_OUTPUT_VOLUME, + 0x50 | (1<<8)); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8991, &wm8991_dai, 1); -- cgit v1.2.3-70-g09d2 From a0a05916cf67a007f4ee0071fd0fa04e45137a38 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 13:34:48 +0000 Subject: ASoC: wm8991: Verify device ID during probe() Just in case. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8991.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 5fdcf788e99..dba0306c42a 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1335,6 +1335,7 @@ static int wm8991_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8991_priv *wm8991; + unsigned int val; int ret; wm8991 = devm_kzalloc(&i2c->dev, sizeof(*wm8991), GFP_KERNEL); @@ -1347,6 +1348,16 @@ static int wm8991_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8991); + ret = regmap_read(wm8991->regmap, WM8991_RESET, &val); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read device ID: %d\n", ret); + return ret; + } + if (val != 0x8991) { + dev_err(&i2c->dev, "Device with ID %x is not a WM8991\n", val); + return -EINVAL; + } + ret = regmap_write(wm8991->regmap, WM8991_RESET, 0); if (ret < 0) { dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); -- cgit v1.2.3-70-g09d2 From 4f534777c130180f9338f0fb96090d43464b7ddf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 15:14:48 +0000 Subject: ASoC: ak4641: Convert to direct regmap API usage We're trying to remove the ASoC level I/O functions. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4641.c | 38 +++++++++++++++++++++++++------------- 1 file changed, 25 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 49cc5f6d6db..94cbe508dd3 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -30,6 +31,7 @@ /* codec private data */ struct ak4641_priv { + struct regmap *regmap; unsigned int sysclk; int deemph; int playback_fs; @@ -38,12 +40,12 @@ struct ak4641_priv { /* * ak4641 register cache */ -static const u8 ak4641_reg[AK4641_CACHEREGNUM] = { - 0x00, 0x80, 0x00, 0x80, - 0x02, 0x00, 0x11, 0x05, - 0x00, 0x00, 0x36, 0x10, - 0x00, 0x00, 0x57, 0x00, - 0x88, 0x88, 0x08, 0x08 +static const struct reg_default ak4641_reg_defaults[] = { + { 0, 0x00 }, { 1, 0x80 }, { 2, 0x00 }, { 3, 0x80 }, + { 4, 0x02 }, { 5, 0x00 }, { 6, 0x11 }, { 7, 0x05 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x36 }, { 11, 0x10 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x57 }, { 15, 0x00 }, + { 16, 0x88 }, { 17, 0x88 }, { 18, 0x08 }, { 19, 0x08 } }; static const int deemph_settings[] = {44100, 0, 48000, 32000}; @@ -396,6 +398,7 @@ static int ak4641_mute(struct snd_soc_dai *dai, int mute) static int ak4641_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); struct ak4641_platform_data *pdata = codec->dev->platform_data; int ret; @@ -417,7 +420,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec, gpio_set_value(pdata->gpio_npdn, 1); mdelay(1); - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(ak4641->regmap); if (ret) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -433,7 +436,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec, gpio_set_value(pdata->gpio_npdn, 0); if (pdata && gpio_is_valid(pdata->gpio_power)) gpio_set_value(pdata->gpio_power, 0); - codec->cache_sync = 1; + regcache_mark_dirty(ak4641->regmap); break; } codec->dapm.bias_level = level; @@ -518,7 +521,7 @@ static int ak4641_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -550,12 +553,17 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { .dapm_routes = ak4641_audio_map, .num_dapm_routes = ARRAY_SIZE(ak4641_audio_map), .set_bias_level = ak4641_set_bias_level, - .reg_cache_size = ARRAY_SIZE(ak4641_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = ak4641_reg, - .reg_cache_step = 1, }; +static const struct regmap_config ak4641_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AK4641_BTIF, + .reg_defaults = ak4641_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4641_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; static int ak4641_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -569,6 +577,10 @@ static int ak4641_i2c_probe(struct i2c_client *i2c, if (!ak4641) return -ENOMEM; + ak4641->regmap = devm_regmap_init_i2c(i2c, &ak4641_regmap); + if (IS_ERR(ak4641->regmap)) + return PTR_ERR(ak4641->regmap); + if (pdata) { if (gpio_is_valid(pdata->gpio_power)) { ret = gpio_request_one(pdata->gpio_power, -- cgit v1.2.3-70-g09d2 From 9a3e1b8c503590a2a89164586c0e65415bb90979 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 19 Nov 2013 16:04:01 +0000 Subject: ASoC: wm5110: Hook up ADSP2 cores Signed-off-by: Mark Brown Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 + sound/soc/codecs/wm5110.c | 119 +++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 119 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b33b45dfcee..983d087aa92 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -163,8 +163,10 @@ config SND_SOC_WM_HUBS config SND_SOC_WM_ADSP tristate default y if SND_SOC_WM5102=y + default y if SND_SOC_WM5110=y default y if SND_SOC_WM2200=y default m if SND_SOC_WM5102=m + default m if SND_SOC_WM5110=m default m if SND_SOC_WM2200=m config SND_SOC_AB8500_CODEC diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c6dbc1ddaf7..e5f2804086a 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -30,13 +30,51 @@ #include #include "arizona.h" +#include "wm_adsp.h" #include "wm5110.h" +#define WM5110_NUM_ADSP 4 + struct wm5110_priv { struct arizona_priv core; struct arizona_fll fll[2]; }; +static const struct wm_adsp_region wm5110_dsp1_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x100000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x180000 }, + { .type = WMFW_ADSP2_XM, .base = 0x190000 }, + { .type = WMFW_ADSP2_YM, .base = 0x1a8000 }, +}; + +static const struct wm_adsp_region wm5110_dsp2_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x200000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x280000 }, + { .type = WMFW_ADSP2_XM, .base = 0x290000 }, + { .type = WMFW_ADSP2_YM, .base = 0x2a8000 }, +}; + +static const struct wm_adsp_region wm5110_dsp3_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x300000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x380000 }, + { .type = WMFW_ADSP2_XM, .base = 0x390000 }, + { .type = WMFW_ADSP2_YM, .base = 0x3a8000 }, +}; + +static const struct wm_adsp_region wm5110_dsp4_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x400000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x480000 }, + { .type = WMFW_ADSP2_XM, .base = 0x490000 }, + { .type = WMFW_ADSP2_YM, .base = 0x4a8000 }, +}; + +static const struct wm_adsp_region *wm5110_dsp_regions[] = { + wm5110_dsp1_regions, + wm5110_dsp2_regions, + wm5110_dsp3_regions, + wm5110_dsp4_regions, +}; + static const struct reg_default wm5110_sysclk_revd_patch[] = { { 0x3093, 0x1001 }, { 0x30E3, 0x1301 }, @@ -395,6 +433,22 @@ ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP1L, ARIZONA_DSP1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP1R, ARIZONA_DSP1RMIX_INPUT_1_SOURCE); +ARIZONA_DSP_AUX_ENUMS(DSP1, ARIZONA_DSP1AUX1MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DSP2L, ARIZONA_DSP2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP2R, ARIZONA_DSP2RMIX_INPUT_1_SOURCE); +ARIZONA_DSP_AUX_ENUMS(DSP2, ARIZONA_DSP2AUX1MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DSP3L, ARIZONA_DSP3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP3R, ARIZONA_DSP3RMIX_INPUT_1_SOURCE); +ARIZONA_DSP_AUX_ENUMS(DSP3, ARIZONA_DSP3AUX1MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DSP4L, ARIZONA_DSP4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP4R, ARIZONA_DSP4RMIX_INPUT_1_SOURCE); +ARIZONA_DSP_AUX_ENUMS(DSP4, ARIZONA_DSP4AUX1MIX_INPUT_1_SOURCE); + ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); @@ -587,6 +641,11 @@ SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0, SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, NULL, 0), +WM_ADSP2("DSP1", 0), +WM_ADSP2("DSP2", 1), +WM_ADSP2("DSP3", 2), +WM_ADSP2("DSP4", 3), + SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &wm5110_aec_loopback_mux), @@ -809,6 +868,11 @@ ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"), ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"), ARIZONA_MUX_WIDGETS(ASRC2R, "ASRC2R"), +ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), +ARIZONA_DSP_WIDGETS(DSP2, "DSP2"), +ARIZONA_DSP_WIDGETS(DSP3, "DSP3"), +ARIZONA_DSP_WIDGETS(DSP4, "DSP4"), + SND_SOC_DAPM_OUTPUT("HPOUT1L"), SND_SOC_DAPM_OUTPUT("HPOUT1R"), SND_SOC_DAPM_OUTPUT("HPOUT2L"), @@ -881,7 +945,31 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"), { name, "ASRC1L", "ASRC1L" }, \ { name, "ASRC1R", "ASRC1R" }, \ { name, "ASRC2L", "ASRC2L" }, \ - { name, "ASRC2R", "ASRC2R" } + { name, "ASRC2R", "ASRC2R" }, \ + { name, "DSP1.1", "DSP1" }, \ + { name, "DSP1.2", "DSP1" }, \ + { name, "DSP1.3", "DSP1" }, \ + { name, "DSP1.4", "DSP1" }, \ + { name, "DSP1.5", "DSP1" }, \ + { name, "DSP1.6", "DSP1" }, \ + { name, "DSP2.1", "DSP2" }, \ + { name, "DSP2.2", "DSP2" }, \ + { name, "DSP2.3", "DSP2" }, \ + { name, "DSP2.4", "DSP2" }, \ + { name, "DSP2.5", "DSP2" }, \ + { name, "DSP2.6", "DSP2" }, \ + { name, "DSP3.1", "DSP3" }, \ + { name, "DSP3.2", "DSP3" }, \ + { name, "DSP3.3", "DSP3" }, \ + { name, "DSP3.4", "DSP3" }, \ + { name, "DSP3.5", "DSP3" }, \ + { name, "DSP3.6", "DSP3" }, \ + { name, "DSP4.1", "DSP4" }, \ + { name, "DSP4.2", "DSP4" }, \ + { name, "DSP4.3", "DSP4" }, \ + { name, "DSP4.4", "DSP4" }, \ + { name, "DSP4.5", "DSP4" }, \ + { name, "DSP4.6", "DSP4" } static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF2 Capture", NULL, "DBVDD2" }, @@ -1087,6 +1175,11 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), + ARIZONA_DSP_ROUTES("DSP1"), + ARIZONA_DSP_ROUTES("DSP2"), + ARIZONA_DSP_ROUTES("DSP3"), + ARIZONA_DSP_ROUTES("DSP4"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, @@ -1292,6 +1385,10 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_spk(codec); arizona_init_gpio(codec); + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); + if (ret != 0) + return ret; + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); priv->core.arizona->dapm = &codec->dapm; @@ -1346,7 +1443,7 @@ static int wm5110_probe(struct platform_device *pdev) { struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); struct wm5110_priv *wm5110; - int i; + int i, ret; wm5110 = devm_kzalloc(&pdev->dev, sizeof(struct wm5110_priv), GFP_KERNEL); @@ -1357,6 +1454,24 @@ static int wm5110_probe(struct platform_device *pdev) wm5110->core.arizona = arizona; wm5110->core.num_inputs = 8; + for (i = 0; i < WM5110_NUM_ADSP; i++) { + wm5110->core.adsp[i].part = "wm5110"; + wm5110->core.adsp[i].num = i + 1; + wm5110->core.adsp[i].type = WMFW_ADSP2; + wm5110->core.adsp[i].dev = arizona->dev; + wm5110->core.adsp[i].regmap = arizona->regmap; + + wm5110->core.adsp[i].base = ARIZONA_DSP1_CONTROL_1 + + (0x100 * i); + wm5110->core.adsp[i].mem = wm5110_dsp_regions[i]; + wm5110->core.adsp[i].num_mems + = ARRAY_SIZE(wm5110_dsp1_regions); + + ret = wm_adsp2_init(&wm5110->core.adsp[i], false); + if (ret != 0) + return ret; + } + for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++) wm5110->fll[i].vco_mult = 3; -- cgit v1.2.3-70-g09d2 From 0da2e5baf4233e2744a7fc691932638c39d9b245 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Nov 2013 16:04:02 +0000 Subject: ASoC: wm5110: Add basic support for ISRCs Add support for the ISRCs that matches the current support on the w5102. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 178 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 178 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index e5f2804086a..0e63d8ce533 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -277,6 +277,10 @@ SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), +SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), +SOC_VALUE_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]), + ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP2L", ARIZONA_DSP2LMIX_INPUT_1_SOURCE), @@ -501,6 +505,36 @@ ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT3, ARIZONA_ISRC1INT3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT4, ARIZONA_ISRC1INT4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC3, ARIZONA_ISRC1DEC3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC4, ARIZONA_ISRC1DEC4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT3, ARIZONA_ISRC2INT3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT4, ARIZONA_ISRC2INT4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC3, ARIZONA_ISRC2DEC3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC4, ARIZONA_ISRC2DEC4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC3INT1, ARIZONA_ISRC3INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3INT2, ARIZONA_ISRC3INT2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3INT3, ARIZONA_ISRC3INT3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3INT4, ARIZONA_ISRC3INT4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC3DEC1, ARIZONA_ISRC3DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3DEC2, ARIZONA_ISRC3DEC2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3DEC3, ARIZONA_ISRC3DEC3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3DEC4, ARIZONA_ISRC3DEC4MIX_INPUT_1_SOURCE); + static const char *wm5110_aec_loopback_texts[] = { "HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "HPOUT3L", "HPOUT3R", "SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R", "SPKDAT2L", "SPKDAT2R", @@ -646,6 +680,60 @@ WM_ADSP2("DSP2", 1), WM_ADSP2("DSP3", 2), WM_ADSP2("DSP4", 3), +SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT3", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT4", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC3", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC4", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT3", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT4", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC3", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC4", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC3INT1", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3INT2", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3INT3", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3INT4", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC3DEC1", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3DEC2", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3DEC3", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3DEC4", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC3_ENA_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &wm5110_aec_loopback_mux), @@ -873,6 +961,36 @@ ARIZONA_DSP_WIDGETS(DSP2, "DSP2"), ARIZONA_DSP_WIDGETS(DSP3, "DSP3"), ARIZONA_DSP_WIDGETS(DSP4, "DSP4"), +ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), +ARIZONA_MUX_WIDGETS(ISRC1DEC3, "ISRC1DEC3"), +ARIZONA_MUX_WIDGETS(ISRC1DEC4, "ISRC1DEC4"), + +ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), +ARIZONA_MUX_WIDGETS(ISRC1INT3, "ISRC1INT3"), +ARIZONA_MUX_WIDGETS(ISRC1INT4, "ISRC1INT4"), + +ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), +ARIZONA_MUX_WIDGETS(ISRC2DEC3, "ISRC2DEC3"), +ARIZONA_MUX_WIDGETS(ISRC2DEC4, "ISRC2DEC4"), + +ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), +ARIZONA_MUX_WIDGETS(ISRC2INT3, "ISRC2INT3"), +ARIZONA_MUX_WIDGETS(ISRC2INT4, "ISRC2INT4"), + +ARIZONA_MUX_WIDGETS(ISRC3DEC1, "ISRC3DEC1"), +ARIZONA_MUX_WIDGETS(ISRC3DEC2, "ISRC3DEC2"), +ARIZONA_MUX_WIDGETS(ISRC3DEC3, "ISRC3DEC3"), +ARIZONA_MUX_WIDGETS(ISRC3DEC4, "ISRC3DEC4"), + +ARIZONA_MUX_WIDGETS(ISRC3INT1, "ISRC3INT1"), +ARIZONA_MUX_WIDGETS(ISRC3INT2, "ISRC3INT2"), +ARIZONA_MUX_WIDGETS(ISRC3INT3, "ISRC3INT3"), +ARIZONA_MUX_WIDGETS(ISRC3INT4, "ISRC3INT4"), + SND_SOC_DAPM_OUTPUT("HPOUT1L"), SND_SOC_DAPM_OUTPUT("HPOUT1R"), SND_SOC_DAPM_OUTPUT("HPOUT2L"), @@ -946,6 +1064,30 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"), { name, "ASRC1R", "ASRC1R" }, \ { name, "ASRC2L", "ASRC2L" }, \ { name, "ASRC2R", "ASRC2R" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1DEC3", "ISRC1DEC3" }, \ + { name, "ISRC1DEC4", "ISRC1DEC4" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC1INT3", "ISRC1INT3" }, \ + { name, "ISRC1INT4", "ISRC1INT4" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2DEC3", "ISRC2DEC3" }, \ + { name, "ISRC2DEC4", "ISRC2DEC4" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" }, \ + { name, "ISRC2INT3", "ISRC2INT3" }, \ + { name, "ISRC2INT4", "ISRC2INT4" }, \ + { name, "ISRC3DEC1", "ISRC3DEC1" }, \ + { name, "ISRC3DEC2", "ISRC3DEC2" }, \ + { name, "ISRC3DEC3", "ISRC3DEC3" }, \ + { name, "ISRC3DEC4", "ISRC3DEC4" }, \ + { name, "ISRC3INT1", "ISRC3INT1" }, \ + { name, "ISRC3INT2", "ISRC3INT2" }, \ + { name, "ISRC3INT3", "ISRC3INT3" }, \ + { name, "ISRC3INT4", "ISRC3INT4" }, \ { name, "DSP1.1", "DSP1" }, \ { name, "DSP1.2", "DSP1" }, \ { name, "DSP1.3", "DSP1" }, \ @@ -1180,6 +1322,36 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_DSP_ROUTES("DSP3"), ARIZONA_DSP_ROUTES("DSP4"), + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), + ARIZONA_MUX_ROUTES("ISRC1INT3", "ISRC1INT3"), + ARIZONA_MUX_ROUTES("ISRC1INT4", "ISRC1INT4"), + + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), + ARIZONA_MUX_ROUTES("ISRC1DEC3", "ISRC1DEC3"), + ARIZONA_MUX_ROUTES("ISRC1DEC4", "ISRC1DEC4"), + + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), + ARIZONA_MUX_ROUTES("ISRC2INT3", "ISRC2INT3"), + ARIZONA_MUX_ROUTES("ISRC2INT4", "ISRC2INT4"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), + ARIZONA_MUX_ROUTES("ISRC2DEC3", "ISRC2DEC3"), + ARIZONA_MUX_ROUTES("ISRC2DEC4", "ISRC2DEC4"), + + ARIZONA_MUX_ROUTES("ISRC3INT1", "ISRC3INT1"), + ARIZONA_MUX_ROUTES("ISRC3INT2", "ISRC3INT2"), + ARIZONA_MUX_ROUTES("ISRC3INT3", "ISRC3INT3"), + ARIZONA_MUX_ROUTES("ISRC3INT4", "ISRC3INT4"), + + ARIZONA_MUX_ROUTES("ISRC3DEC1", "ISRC3DEC1"), + ARIZONA_MUX_ROUTES("ISRC3DEC2", "ISRC3DEC2"), + ARIZONA_MUX_ROUTES("ISRC3DEC3", "ISRC3DEC3"), + ARIZONA_MUX_ROUTES("ISRC3DEC4", "ISRC3DEC4"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, @@ -1482,6 +1654,12 @@ static int wm5110_probe(struct platform_device *pdev) ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, &wm5110->fll[1]); + /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */ + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2, + ARIZONA_SAMPLE_RATE_2_MASK, 0x11); + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3, + ARIZONA_SAMPLE_RATE_3_MASK, 0x12); + for (i = 0; i < ARRAY_SIZE(wm5110_dai); i++) arizona_init_dai(&wm5110->core, i); -- cgit v1.2.3-70-g09d2 From 93818c9a12dd38f2b32f960f979815ac2e15a176 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 13:42:51 +0000 Subject: ASoC: wm8990: Convet to module_i2c_driver() The device is I2C only (or at least current support is). Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8990.c | 24 +----------------------- 1 file changed, 1 insertion(+), 23 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 4f05fb88bdd..33bec567241 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1378,7 +1378,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .volatile_register = wm8990_volatile_register, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8990_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1420,29 +1419,8 @@ static struct i2c_driver wm8990_i2c_driver = { .remove = wm8990_i2c_remove, .id_table = wm8990_i2c_id, }; -#endif -static int __init wm8990_modinit(void) -{ - int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&wm8990_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8990 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(wm8990_modinit); - -static void __exit wm8990_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&wm8990_i2c_driver); -#endif -} -module_exit(wm8990_exit); +module_i2c_driver(wm8990_i2c_driver); MODULE_DESCRIPTION("ASoC WM8990 driver"); MODULE_AUTHOR("Liam Girdwood"); -- cgit v1.2.3-70-g09d2 From f6b415b6065041c0970426cc8ac81a980b2998f2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 13:44:56 +0000 Subject: ASoC: wm8990: Convert to table based control and DAPM init Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8990.c | 24 +++++++----------------- 1 file changed, 7 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 33bec567241..6ee1cf1d353 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -787,7 +787,7 @@ SND_SOC_DAPM_OUTPUT("RON"), SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8990_dapm_routes[] = { /* Make DACs turn on when playing even if not mixed into any outputs */ {"Internal DAC Sink", NULL, "Left DAC"}, {"Internal DAC Sink", NULL, "Right DAC"}, @@ -912,18 +912,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"RON", NULL, "RONMIX"}, }; -static int wm8990_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8990_dapm_widgets, - ARRAY_SIZE(wm8990_dapm_widgets)); - /* set up the WM8990 audio map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - /* PLL divisors */ struct _pll_div { u32 div2; @@ -1352,10 +1340,6 @@ static int wm8990_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - snd_soc_add_codec_controls(codec, wm8990_snd_controls, - ARRAY_SIZE(wm8990_snd_controls)); - wm8990_add_widgets(codec); - return 0; } @@ -1376,6 +1360,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm8990_reg, .volatile_register = wm8990_volatile_register, + .controls = wm8990_snd_controls, + .num_controls = ARRAY_SIZE(wm8990_snd_controls), + .dapm_widgets = wm8990_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8990_dapm_widgets), + .dapm_routes = wm8990_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8990_dapm_routes), }; static int wm8990_i2c_probe(struct i2c_client *i2c, -- cgit v1.2.3-70-g09d2 From d2fd5fe7ee3bc231e21aeb9ee120e0e61a79f8be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 14:25:04 +0000 Subject: ASoC: wm8990: Use supplies to manage input power Instead of using a fake register use a supply widget to manage the power for the inputs, this is more idiomatic and supports regmap conversion. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8990.c | 54 ++++++++++++++--------------------------------- sound/soc/codecs/wm8990.h | 9 -------- 2 files changed, 16 insertions(+), 47 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 6ee1cf1d353..2261fe1b68c 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -376,32 +376,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, * _DAPM_ Controls */ -static int inmixer_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - u16 reg, fakepower; - - reg = snd_soc_read(w->codec, WM8990_POWER_MANAGEMENT_2); - fakepower = snd_soc_read(w->codec, WM8990_INTDRIVBITS); - - if (fakepower & ((1 << WM8990_INMIXL_PWR_BIT) | - (1 << WM8990_AINLMUX_PWR_BIT))) { - reg |= WM8990_AINL_ENA; - } else { - reg &= ~WM8990_AINL_ENA; - } - - if (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) | - (1 << WM8990_AINRMUX_PWR_BIT))) { - reg |= WM8990_AINR_ENA; - } else { - reg &= ~WM8990_AINR_ENA; - } - snd_soc_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg); - - return 0; -} - static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -656,6 +630,11 @@ SND_SOC_DAPM_INPUT("RIN1"), SND_SOC_DAPM_INPUT("RIN2"), SND_SOC_DAPM_INPUT("Internal ADC Source"), +SND_SOC_DAPM_SUPPLY("INL", WM8990_POWER_MANAGEMENT_2, WM8990_AINL_ENA_BIT, 0, + NULL, 0), +SND_SOC_DAPM_SUPPLY("INR", WM8990_POWER_MANAGEMENT_2, WM8990_AINR_ENA_BIT, 0, + NULL, 0), + /* DACs */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8990_POWER_MANAGEMENT_2, WM8990_ADCL_ENA_BIT, 0), @@ -677,26 +656,20 @@ SND_SOC_DAPM_MIXER("RIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN34_ENA_BIT, ARRAY_SIZE(wm8990_dapm_rin34_pga_controls)), /* INMIXL */ -SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0, +SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0, &wm8990_dapm_inmixl_controls[0], - ARRAY_SIZE(wm8990_dapm_inmixl_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8990_dapm_inmixl_controls)), /* AINLMUX */ -SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, - &wm8990_dapm_ainlmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AINLMUX", SND_SOC_NOPM, 0, 0, &wm8990_dapm_ainlmux_controls), /* INMIXR */ -SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0, +SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0, &wm8990_dapm_inmixr_controls[0], - ARRAY_SIZE(wm8990_dapm_inmixr_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8990_dapm_inmixr_controls)), /* AINRMUX */ -SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, - &wm8990_dapm_ainrmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AINRMUX", SND_SOC_NOPM, 0, 0, &wm8990_dapm_ainrmux_controls), /* Output Side */ /* DACs */ @@ -796,6 +769,11 @@ static const struct snd_soc_dapm_route wm8990_dapm_routes[] = { {"Left ADC", NULL, "Internal ADC Source"}, {"Right ADC", NULL, "Internal ADC Source"}, + {"AINLMUX", NULL, "INL"}, + {"INMIXL", NULL, "INL"}, + {"AINRMUX", NULL, "INR"}, + {"INMIXR", NULL, "INR"}, + /* Input Side */ /* LIN12 PGA */ {"LIN12 PGA", "LIN1 Switch", "LIN1"}, diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h index 77c98a4bfe9..0e9c78040c4 100644 --- a/sound/soc/codecs/wm8990.h +++ b/sound/soc/codecs/wm8990.h @@ -78,7 +78,6 @@ #define WM8990_PLL1 0x3C #define WM8990_PLL2 0x3D #define WM8990_PLL3 0x3E -#define WM8990_INTDRIVBITS 0x3F #define WM8990_EXT_ACCESS_ENA 0x75 #define WM8990_EXT_CTL1 0x7a @@ -818,14 +817,6 @@ */ #define WM8990_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */ -/* - * R63 (0x3F) - Internal Driver Bits - */ -#define WM8990_INMIXL_PWR_BIT 0 -#define WM8990_AINLMUX_PWR_BIT 1 -#define WM8990_INMIXR_PWR_BIT 2 -#define WM8990_AINRMUX_PWR_BIT 3 - #define WM8990_MCLK_DIV 0 #define WM8990_DACCLK_DIV 1 #define WM8990_ADCCLK_DIV 2 -- cgit v1.2.3-70-g09d2 From 0112b62b12e18b883e1027689acab8eaa8830bac Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Nov 2013 14:36:23 +0000 Subject: ASoC: wm8990: Convert to direct regmap API usage Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8990.c | 156 ++++++++++++++++++++++++---------------------- 1 file changed, 81 insertions(+), 75 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 2261fe1b68c..0ccd4d8d043 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -30,13 +31,12 @@ /* codec private data */ struct wm8990_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; unsigned int sysclk; unsigned int pcmclk; }; -static int wm8990_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm8990_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8990_RESET: @@ -46,71 +46,69 @@ static int wm8990_volatile_register(struct snd_soc_codec *codec, } } -static const u16 wm8990_reg[] = { - 0x8990, /* R0 - Reset */ - 0x0000, /* R1 - Power Management (1) */ - 0x6000, /* R2 - Power Management (2) */ - 0x0000, /* R3 - Power Management (3) */ - 0x4050, /* R4 - Audio Interface (1) */ - 0x4000, /* R5 - Audio Interface (2) */ - 0x01C8, /* R6 - Clocking (1) */ - 0x0000, /* R7 - Clocking (2) */ - 0x0040, /* R8 - Audio Interface (3) */ - 0x0040, /* R9 - Audio Interface (4) */ - 0x0004, /* R10 - DAC CTRL */ - 0x00C0, /* R11 - Left DAC Digital Volume */ - 0x00C0, /* R12 - Right DAC Digital Volume */ - 0x0000, /* R13 - Digital Side Tone */ - 0x0100, /* R14 - ADC CTRL */ - 0x00C0, /* R15 - Left ADC Digital Volume */ - 0x00C0, /* R16 - Right ADC Digital Volume */ - 0x0000, /* R17 */ - 0x0000, /* R18 - GPIO CTRL 1 */ - 0x1000, /* R19 - GPIO1 & GPIO2 */ - 0x1010, /* R20 - GPIO3 & GPIO4 */ - 0x1010, /* R21 - GPIO5 & GPIO6 */ - 0x8000, /* R22 - GPIOCTRL 2 */ - 0x0800, /* R23 - GPIO_POL */ - 0x008B, /* R24 - Left Line Input 1&2 Volume */ - 0x008B, /* R25 - Left Line Input 3&4 Volume */ - 0x008B, /* R26 - Right Line Input 1&2 Volume */ - 0x008B, /* R27 - Right Line Input 3&4 Volume */ - 0x0000, /* R28 - Left Output Volume */ - 0x0000, /* R29 - Right Output Volume */ - 0x0066, /* R30 - Line Outputs Volume */ - 0x0022, /* R31 - Out3/4 Volume */ - 0x0079, /* R32 - Left OPGA Volume */ - 0x0079, /* R33 - Right OPGA Volume */ - 0x0003, /* R34 - Speaker Volume */ - 0x0003, /* R35 - ClassD1 */ - 0x0000, /* R36 */ - 0x0100, /* R37 - ClassD3 */ - 0x0079, /* R38 - ClassD4 */ - 0x0000, /* R39 - Input Mixer1 */ - 0x0000, /* R40 - Input Mixer2 */ - 0x0000, /* R41 - Input Mixer3 */ - 0x0000, /* R42 - Input Mixer4 */ - 0x0000, /* R43 - Input Mixer5 */ - 0x0000, /* R44 - Input Mixer6 */ - 0x0000, /* R45 - Output Mixer1 */ - 0x0000, /* R46 - Output Mixer2 */ - 0x0000, /* R47 - Output Mixer3 */ - 0x0000, /* R48 - Output Mixer4 */ - 0x0000, /* R49 - Output Mixer5 */ - 0x0000, /* R50 - Output Mixer6 */ - 0x0180, /* R51 - Out3/4 Mixer */ - 0x0000, /* R52 - Line Mixer1 */ - 0x0000, /* R53 - Line Mixer2 */ - 0x0000, /* R54 - Speaker Mixer */ - 0x0000, /* R55 - Additional Control */ - 0x0000, /* R56 - AntiPOP1 */ - 0x0000, /* R57 - AntiPOP2 */ - 0x0000, /* R58 - MICBIAS */ - 0x0000, /* R59 */ - 0x0008, /* R60 - PLL1 */ - 0x0031, /* R61 - PLL2 */ - 0x0026, /* R62 - PLL3 */ - 0x0000, /* R63 - Driver internal */ +static const struct reg_default wm8990_reg_defaults[] = { + { 1, 0x0000 }, /* R1 - Power Management (1) */ + { 2, 0x6000 }, /* R2 - Power Management (2) */ + { 3, 0x0000 }, /* R3 - Power Management (3) */ + { 4, 0x4050 }, /* R4 - Audio Interface (1) */ + { 5, 0x4000 }, /* R5 - Audio Interface (2) */ + { 6, 0x01C8 }, /* R6 - Clocking (1) */ + { 7, 0x0000 }, /* R7 - Clocking (2) */ + { 8, 0x0040 }, /* R8 - Audio Interface (3) */ + { 9, 0x0040 }, /* R9 - Audio Interface (4) */ + { 10, 0x0004 }, /* R10 - DAC CTRL */ + { 11, 0x00C0 }, /* R11 - Left DAC Digital Volume */ + { 12, 0x00C0 }, /* R12 - Right DAC Digital Volume */ + { 13, 0x0000 }, /* R13 - Digital Side Tone */ + { 14, 0x0100 }, /* R14 - ADC CTRL */ + { 15, 0x00C0 }, /* R15 - Left ADC Digital Volume */ + { 16, 0x00C0 }, /* R16 - Right ADC Digital Volume */ + + { 18, 0x0000 }, /* R18 - GPIO CTRL 1 */ + { 19, 0x1000 }, /* R19 - GPIO1 & GPIO2 */ + { 20, 0x1010 }, /* R20 - GPIO3 & GPIO4 */ + { 21, 0x1010 }, /* R21 - GPIO5 & GPIO6 */ + { 22, 0x8000 }, /* R22 - GPIOCTRL 2 */ + { 23, 0x0800 }, /* R23 - GPIO_POL */ + { 24, 0x008B }, /* R24 - Left Line Input 1&2 Volume */ + { 25, 0x008B }, /* R25 - Left Line Input 3&4 Volume */ + { 26, 0x008B }, /* R26 - Right Line Input 1&2 Volume */ + { 27, 0x008B }, /* R27 - Right Line Input 3&4 Volume */ + { 28, 0x0000 }, /* R28 - Left Output Volume */ + { 29, 0x0000 }, /* R29 - Right Output Volume */ + { 30, 0x0066 }, /* R30 - Line Outputs Volume */ + { 31, 0x0022 }, /* R31 - Out3/4 Volume */ + { 32, 0x0079 }, /* R32 - Left OPGA Volume */ + { 33, 0x0079 }, /* R33 - Right OPGA Volume */ + { 34, 0x0003 }, /* R34 - Speaker Volume */ + { 35, 0x0003 }, /* R35 - ClassD1 */ + + { 37, 0x0100 }, /* R37 - ClassD3 */ + { 38, 0x0079 }, /* R38 - ClassD4 */ + { 39, 0x0000 }, /* R39 - Input Mixer1 */ + { 40, 0x0000 }, /* R40 - Input Mixer2 */ + { 41, 0x0000 }, /* R41 - Input Mixer3 */ + { 42, 0x0000 }, /* R42 - Input Mixer4 */ + { 43, 0x0000 }, /* R43 - Input Mixer5 */ + { 44, 0x0000 }, /* R44 - Input Mixer6 */ + { 45, 0x0000 }, /* R45 - Output Mixer1 */ + { 46, 0x0000 }, /* R46 - Output Mixer2 */ + { 47, 0x0000 }, /* R47 - Output Mixer3 */ + { 48, 0x0000 }, /* R48 - Output Mixer4 */ + { 49, 0x0000 }, /* R49 - Output Mixer5 */ + { 50, 0x0000 }, /* R50 - Output Mixer6 */ + { 51, 0x0180 }, /* R51 - Out3/4 Mixer */ + { 52, 0x0000 }, /* R52 - Line Mixer1 */ + { 53, 0x0000 }, /* R53 - Line Mixer2 */ + { 54, 0x0000 }, /* R54 - Speaker Mixer */ + { 55, 0x0000 }, /* R55 - Additional Control */ + { 56, 0x0000 }, /* R56 - AntiPOP1 */ + { 57, 0x0000 }, /* R57 - AntiPOP2 */ + { 58, 0x0000 }, /* R58 - MICBIAS */ + + { 60, 0x0008 }, /* R60 - PLL1 */ + { 61, 0x0031 }, /* R61 - PLL2 */ + { 62, 0x0026 }, /* R62 - PLL3 */ }; #define wm8990_reset(c) snd_soc_write(c, WM8990_RESET, 0) @@ -1114,6 +1112,7 @@ static int wm8990_mute(struct snd_soc_dai *dai, int mute) static int wm8990_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8990_priv *wm8990 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { @@ -1128,7 +1127,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(wm8990->regmap); if (ret < 0) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); return ret; @@ -1226,7 +1225,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ snd_soc_write(codec, WM8990_ANTIPOP2, 0x0); - codec->cache_sync = 1; + regcache_mark_dirty(wm8990->regmap); break; } @@ -1295,7 +1294,7 @@ static int wm8990_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret); return ret; @@ -1334,10 +1333,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .suspend = wm8990_suspend, .resume = wm8990_resume, .set_bias_level = wm8990_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8990_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8990_reg, - .volatile_register = wm8990_volatile_register, .controls = wm8990_snd_controls, .num_controls = ARRAY_SIZE(wm8990_snd_controls), .dapm_widgets = wm8990_dapm_widgets, @@ -1346,6 +1341,17 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .num_dapm_routes = ARRAY_SIZE(wm8990_dapm_routes), }; +static const struct regmap_config wm8990_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM8990_PLL3, + .volatile_reg = wm8990_volatile_register, + .reg_defaults = wm8990_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8990_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + static int wm8990_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { -- cgit v1.2.3-70-g09d2 From d44008b358588cf6fcc74716b50584a8e59cbe65 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:48:31 -0200 Subject: ASoC: wm8995: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index da2899e6c40..4300caff178 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2293,7 +2293,7 @@ static struct spi_driver wm8995_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8995_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2350,7 +2350,7 @@ static int __init wm8995_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8995_i2c_driver); if (ret) { printk(KERN_ERR "Failed to register wm8995 I2C driver: %d\n", @@ -2371,7 +2371,7 @@ module_init(wm8995_modinit); static void __exit wm8995_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8995_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From c6aeb7de226dd08ad9b343fc6cbaf2ff36f69c84 Mon Sep 17 00:00:00 2001 From: Florian Meier Date: Fri, 22 Nov 2013 16:24:08 +0100 Subject: ASoC: Add support for BCM2835 This driver adds support for digital audio (I2S) for the BCM2835 SoC that is used by the Raspberry Pi. External audio codecs can be connected to the Raspberry Pi via P5 header. It relies on cyclic DMA engine support for BCM2835. Signed-off-by: Florian Meier Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/bcm2835-i2s.txt | 25 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/bcm/Kconfig | 10 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/bcm2835-i2s.c | 886 +++++++++++++++++++++ 6 files changed, 928 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/bcm2835-i2s.txt create mode 100644 sound/soc/bcm/Kconfig create mode 100644 sound/soc/bcm/Makefile create mode 100644 sound/soc/bcm/bcm2835-i2s.c (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt b/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt new file mode 100644 index 00000000000..65783de0aed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt @@ -0,0 +1,25 @@ +* Broadcom BCM2835 SoC I2S/PCM module + +Required properties: +- compatible: "brcm,bcm2835-i2s" +- reg: A list of base address and size entries: + * The first entry should cover the PCM registers + * The second entry should cover the PCM clock registers +- dmas: List of DMA controller phandle and DMA request line ordered pairs. +- dma-names: Identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + + One of the DMA channels will be responsible for transmission (should be + named "tx") and one for reception (should be named "rx"). + +Example: + +bcm2835_i2s: i2s@7e203000 { + compatible = "brcm,bcm2835-i2s"; + reg = <0x7e203000 0x20>, + <0x7e101098 0x02>; + + dmas = <&dma 2>, + <&dma 3>; + dma-names = "tx", "rx"; +}; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5138b849305..a5e3a70c0d3 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM # All the supported SoCs source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" +source "sound/soc/bcm/Kconfig" source "sound/soc/blackfin/Kconfig" source "sound/soc/cirrus/Kconfig" source "sound/soc/davinci/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8b9e70105dd..b52d4aad071 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,6 +10,7 @@ obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ +obj-$(CONFIG_SND_SOC) += bcm/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += cirrus/ obj-$(CONFIG_SND_SOC) += davinci/ diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig new file mode 100644 index 00000000000..3d82a29ce3a --- /dev/null +++ b/sound/soc/bcm/Kconfig @@ -0,0 +1,10 @@ +config SND_BCM2835_SOC_I2S + tristate "SoC Audio support for the Broadcom BCM2835 I2S module" + depends on ARCH_BCM2835 || COMPILE_TEST + select SND_SOC_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to + the BCM2835 I2S interface. You will also need + to select the audio interfaces to support below. diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile new file mode 100644 index 00000000000..bc816b71e5a --- /dev/null +++ b/sound/soc/bcm/Makefile @@ -0,0 +1,5 @@ +# BCM2835 Platform Support +snd-soc-bcm2835-i2s-objs := bcm2835-i2s.o + +obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o + diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c new file mode 100644 index 00000000000..f49b007c8b4 --- /dev/null +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -0,0 +1,886 @@ +/* + * ALSA SoC I2S Audio Layer for Broadcom BCM2835 SoC + * + * Author: Florian Meier + * Copyright 2013 + * + * Based on + * Raspberry Pi PCM I2S ALSA Driver + * Copyright (c) by Phil Poole 2013 + * + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * Vladimir Barinov, + * Copyright (C) 2007 MontaVista Software, Inc., + * + * OMAP ALSA SoC DAI driver using McBSP port + * Copyright (C) 2008 Nokia Corporation + * Contact: Jarkko Nikula + * Peter Ujfalusi + * + * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver + * Author: Timur Tabi + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +/* Clock registers */ +#define BCM2835_CLK_PCMCTL_REG 0x00 +#define BCM2835_CLK_PCMDIV_REG 0x04 + +/* Clock register settings */ +#define BCM2835_CLK_PASSWD (0x5a000000) +#define BCM2835_CLK_PASSWD_MASK (0xff000000) +#define BCM2835_CLK_MASH(v) ((v) << 9) +#define BCM2835_CLK_FLIP BIT(8) +#define BCM2835_CLK_BUSY BIT(7) +#define BCM2835_CLK_KILL BIT(5) +#define BCM2835_CLK_ENAB BIT(4) +#define BCM2835_CLK_SRC(v) (v) + +#define BCM2835_CLK_SHIFT (12) +#define BCM2835_CLK_DIVI(v) ((v) << BCM2835_CLK_SHIFT) +#define BCM2835_CLK_DIVF(v) (v) +#define BCM2835_CLK_DIVF_MASK (0xFFF) + +enum { + BCM2835_CLK_MASH_0 = 0, + BCM2835_CLK_MASH_1, + BCM2835_CLK_MASH_2, + BCM2835_CLK_MASH_3, +}; + +enum { + BCM2835_CLK_SRC_GND = 0, + BCM2835_CLK_SRC_OSC, + BCM2835_CLK_SRC_DBG0, + BCM2835_CLK_SRC_DBG1, + BCM2835_CLK_SRC_PLLA, + BCM2835_CLK_SRC_PLLC, + BCM2835_CLK_SRC_PLLD, + BCM2835_CLK_SRC_HDMI, +}; + +/* Most clocks are not useable (freq = 0) */ +static const unsigned int bcm2835_clk_freq[BCM2835_CLK_SRC_HDMI+1] = { + [BCM2835_CLK_SRC_GND] = 0, + [BCM2835_CLK_SRC_OSC] = 19200000, + [BCM2835_CLK_SRC_DBG0] = 0, + [BCM2835_CLK_SRC_DBG1] = 0, + [BCM2835_CLK_SRC_PLLA] = 0, + [BCM2835_CLK_SRC_PLLC] = 0, + [BCM2835_CLK_SRC_PLLD] = 500000000, + [BCM2835_CLK_SRC_HDMI] = 0, +}; + +/* I2S registers */ +#define BCM2835_I2S_CS_A_REG 0x00 +#define BCM2835_I2S_FIFO_A_REG 0x04 +#define BCM2835_I2S_MODE_A_REG 0x08 +#define BCM2835_I2S_RXC_A_REG 0x0c +#define BCM2835_I2S_TXC_A_REG 0x10 +#define BCM2835_I2S_DREQ_A_REG 0x14 +#define BCM2835_I2S_INTEN_A_REG 0x18 +#define BCM2835_I2S_INTSTC_A_REG 0x1c +#define BCM2835_I2S_GRAY_REG 0x20 + +/* I2S register settings */ +#define BCM2835_I2S_STBY BIT(25) +#define BCM2835_I2S_SYNC BIT(24) +#define BCM2835_I2S_RXSEX BIT(23) +#define BCM2835_I2S_RXF BIT(22) +#define BCM2835_I2S_TXE BIT(21) +#define BCM2835_I2S_RXD BIT(20) +#define BCM2835_I2S_TXD BIT(19) +#define BCM2835_I2S_RXR BIT(18) +#define BCM2835_I2S_TXW BIT(17) +#define BCM2835_I2S_CS_RXERR BIT(16) +#define BCM2835_I2S_CS_TXERR BIT(15) +#define BCM2835_I2S_RXSYNC BIT(14) +#define BCM2835_I2S_TXSYNC BIT(13) +#define BCM2835_I2S_DMAEN BIT(9) +#define BCM2835_I2S_RXTHR(v) ((v) << 7) +#define BCM2835_I2S_TXTHR(v) ((v) << 5) +#define BCM2835_I2S_RXCLR BIT(4) +#define BCM2835_I2S_TXCLR BIT(3) +#define BCM2835_I2S_TXON BIT(2) +#define BCM2835_I2S_RXON BIT(1) +#define BCM2835_I2S_EN (1) + +#define BCM2835_I2S_CLKDIS BIT(28) +#define BCM2835_I2S_PDMN BIT(27) +#define BCM2835_I2S_PDME BIT(26) +#define BCM2835_I2S_FRXP BIT(25) +#define BCM2835_I2S_FTXP BIT(24) +#define BCM2835_I2S_CLKM BIT(23) +#define BCM2835_I2S_CLKI BIT(22) +#define BCM2835_I2S_FSM BIT(21) +#define BCM2835_I2S_FSI BIT(20) +#define BCM2835_I2S_FLEN(v) ((v) << 10) +#define BCM2835_I2S_FSLEN(v) (v) + +#define BCM2835_I2S_CHWEX BIT(15) +#define BCM2835_I2S_CHEN BIT(14) +#define BCM2835_I2S_CHPOS(v) ((v) << 4) +#define BCM2835_I2S_CHWID(v) (v) +#define BCM2835_I2S_CH1(v) ((v) << 16) +#define BCM2835_I2S_CH2(v) (v) + +#define BCM2835_I2S_TX_PANIC(v) ((v) << 24) +#define BCM2835_I2S_RX_PANIC(v) ((v) << 16) +#define BCM2835_I2S_TX(v) ((v) << 8) +#define BCM2835_I2S_RX(v) (v) + +#define BCM2835_I2S_INT_RXERR BIT(3) +#define BCM2835_I2S_INT_TXERR BIT(2) +#define BCM2835_I2S_INT_RXR BIT(1) +#define BCM2835_I2S_INT_TXW BIT(0) + +/* I2S DMA interface */ +/* FIXME: Needs IOMMU support */ +#define BCM2835_VCMMU_SHIFT (0x7E000000 - 0x20000000) + +/* General device struct */ +struct bcm2835_i2s_dev { + struct device *dev; + struct snd_dmaengine_dai_dma_data dma_data[2]; + unsigned int fmt; + unsigned int bclk_ratio; + + struct regmap *i2s_regmap; + struct regmap *clk_regmap; +}; + +static void bcm2835_i2s_start_clock(struct bcm2835_i2s_dev *dev) +{ + /* Start the clock if in master mode */ + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD | BCM2835_CLK_ENAB); + break; + default: + break; + } +} + +static void bcm2835_i2s_stop_clock(struct bcm2835_i2s_dev *dev) +{ + uint32_t clkreg; + int timeout = 1000; + + /* Stop clock */ + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD); + + /* Wait for the BUSY flag going down */ + while (--timeout) { + regmap_read(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, &clkreg); + if (!(clkreg & BCM2835_CLK_BUSY)) + break; + } + + if (!timeout) { + /* KILL the clock */ + dev_err(dev->dev, "I2S clock didn't stop. Kill the clock!\n"); + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_KILL | BCM2835_CLK_PASSWD_MASK, + BCM2835_CLK_KILL | BCM2835_CLK_PASSWD); + } +} + +static void bcm2835_i2s_clear_fifos(struct bcm2835_i2s_dev *dev, + bool tx, bool rx) +{ + int timeout = 1000; + uint32_t syncval; + uint32_t csreg; + uint32_t i2s_active_state; + uint32_t clkreg; + uint32_t clk_active_state; + uint32_t off; + uint32_t clr; + + off = tx ? BCM2835_I2S_TXON : 0; + off |= rx ? BCM2835_I2S_RXON : 0; + + clr = tx ? BCM2835_I2S_TXCLR : 0; + clr |= rx ? BCM2835_I2S_RXCLR : 0; + + /* Backup the current state */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + i2s_active_state = csreg & (BCM2835_I2S_RXON | BCM2835_I2S_TXON); + + regmap_read(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, &clkreg); + clk_active_state = clkreg & BCM2835_CLK_ENAB; + + /* Start clock if not running */ + if (!clk_active_state) { + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD | BCM2835_CLK_ENAB); + } + + /* Stop I2S module */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, off, 0); + + /* + * Clear the FIFOs + * Requires at least 2 PCM clock cycles to take effect + */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, clr, clr); + + /* Wait for 2 PCM clock cycles */ + + /* + * Toggle the SYNC flag. After 2 PCM clock cycles it can be read back + * FIXME: This does not seem to work for slave mode! + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &syncval); + syncval &= BCM2835_I2S_SYNC; + + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_SYNC, ~syncval); + + /* Wait for the SYNC flag changing it's state */ + while (--timeout) { + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + if ((csreg & BCM2835_I2S_SYNC) != syncval) + break; + } + + if (!timeout) + dev_err(dev->dev, "I2S SYNC error!\n"); + + /* Stop clock if it was not running before */ + if (!clk_active_state) + bcm2835_i2s_stop_clock(dev); + + /* Restore I2S state */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_RXON | BCM2835_I2S_TXON, i2s_active_state); +} + +static int bcm2835_i2s_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + dev->fmt = fmt; + return 0; +} + +static int bcm2835_i2s_set_dai_bclk_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + dev->bclk_ratio = ratio; + return 0; +} + +static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + unsigned int sampling_rate = params_rate(params); + unsigned int data_length, data_delay, bclk_ratio; + unsigned int ch1pos, ch2pos, mode, format; + unsigned int mash = BCM2835_CLK_MASH_1; + unsigned int divi, divf, target_frequency; + int clk_src = -1; + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + bool bit_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBS_CFM); + + bool frame_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBM_CFS); + uint32_t csreg; + + /* + * If a stream is already enabled, + * the registers are already set properly. + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + + if (csreg & (BCM2835_I2S_TXON | BCM2835_I2S_RXON)) + return 0; + + /* + * Adjust the data length according to the format. + * We prefill the half frame length with an integer + * divider of 2400 as explained at the clock settings. + * Maybe it is overwritten there, if the Integer mode + * does not apply. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + data_length = 16; + bclk_ratio = 40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + data_length = 32; + bclk_ratio = 80; + break; + default: + return -EINVAL; + } + + /* If bclk_ratio already set, use that one. */ + if (dev->bclk_ratio) + bclk_ratio = dev->bclk_ratio; + + /* + * Clock Settings + * + * The target frequency of the bit clock is + * sampling rate * frame length + * + * Integer mode: + * Sampling rates that are multiples of 8000 kHz + * can be driven by the oscillator of 19.2 MHz + * with an integer divider as long as the frame length + * is an integer divider of 19200000/8000=2400 as set up above. + * This is no longer possible if the sampling rate + * is too high (e.g. 192 kHz), because the oscillator is too slow. + * + * MASH mode: + * For all other sampling rates, it is not possible to + * have an integer divider. Approximate the clock + * with the MASH module that induces a slight frequency + * variance. To minimize that it is best to have the fastest + * clock here. That is PLLD with 500 MHz. + */ + target_frequency = sampling_rate * bclk_ratio; + clk_src = BCM2835_CLK_SRC_OSC; + mash = BCM2835_CLK_MASH_0; + + if (bcm2835_clk_freq[clk_src] % target_frequency == 0 + && bit_master && frame_master) { + divi = bcm2835_clk_freq[clk_src] / target_frequency; + divf = 0; + } else { + uint64_t dividend; + + if (!dev->bclk_ratio) { + /* + * Overwrite bclk_ratio, because the + * above trick is not needed or can + * not be used. + */ + bclk_ratio = 2 * data_length; + } + + target_frequency = sampling_rate * bclk_ratio; + + clk_src = BCM2835_CLK_SRC_PLLD; + mash = BCM2835_CLK_MASH_1; + + dividend = bcm2835_clk_freq[clk_src]; + dividend <<= BCM2835_CLK_SHIFT; + do_div(dividend, target_frequency); + divi = dividend >> BCM2835_CLK_SHIFT; + divf = dividend & BCM2835_CLK_DIVF_MASK; + } + + /* Set clock divider */ + regmap_write(dev->clk_regmap, BCM2835_CLK_PCMDIV_REG, BCM2835_CLK_PASSWD + | BCM2835_CLK_DIVI(divi) + | BCM2835_CLK_DIVF(divf)); + + /* Setup clock, but don't start it yet */ + regmap_write(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, BCM2835_CLK_PASSWD + | BCM2835_CLK_MASH(mash) + | BCM2835_CLK_SRC(clk_src)); + + /* Setup the frame format */ + format = BCM2835_I2S_CHEN; + + if (data_length > 24) + format |= BCM2835_I2S_CHWEX; + + format |= BCM2835_I2S_CHWID((data_length-8)&0xf); + + switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + data_delay = 1; + break; + default: + /* + * TODO + * Others are possible but are not implemented at the moment. + */ + dev_err(dev->dev, "%s:bad format\n", __func__); + return -EINVAL; + } + + ch1pos = data_delay; + ch2pos = bclk_ratio / 2 + data_delay; + + switch (params_channels(params)) { + case 2: + format = BCM2835_I2S_CH1(format) | BCM2835_I2S_CH2(format); + format |= BCM2835_I2S_CH1(BCM2835_I2S_CHPOS(ch1pos)); + format |= BCM2835_I2S_CH2(BCM2835_I2S_CHPOS(ch2pos)); + break; + default: + return -EINVAL; + } + + /* + * Set format for both streams. + * We cannot set another frame length + * (and therefore word length) anyway, + * so the format will be the same. + */ + regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, format); + regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, format); + + /* Setup the I2S mode */ + mode = 0; + + if (data_length <= 16) { + /* + * Use frame packed mode (2 channels per 32 bit word) + * We cannot set another frame length in the second stream + * (and therefore word length) anyway, + * so the format will be the same. + */ + mode |= BCM2835_I2S_FTXP | BCM2835_I2S_FRXP; + } + + mode |= BCM2835_I2S_FLEN(bclk_ratio - 1); + mode |= BCM2835_I2S_FSLEN(bclk_ratio / 2); + + /* Master or slave? */ + switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* CPU is master */ + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* + * CODEC is bit clock master + * CPU is frame master + */ + mode |= BCM2835_I2S_CLKM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + /* + * CODEC is frame master + * CPU is bit clock master + */ + mode |= BCM2835_I2S_FSM; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* CODEC is master */ + mode |= BCM2835_I2S_CLKM; + mode |= BCM2835_I2S_FSM; + break; + default: + dev_err(dev->dev, "%s:bad master\n", __func__); + return -EINVAL; + } + + /* + * Invert clocks? + * + * The BCM approach seems to be inverted to the classical I2S approach. + */ + switch (dev->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* None. Therefore, both for BCM */ + mode |= BCM2835_I2S_CLKI; + mode |= BCM2835_I2S_FSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Both. Therefore, none for BCM */ + break; + case SND_SOC_DAIFMT_NB_IF: + /* + * Invert only frame sync. Therefore, + * invert only bit clock for BCM + */ + mode |= BCM2835_I2S_CLKI; + break; + case SND_SOC_DAIFMT_IB_NF: + /* + * Invert only bit clock. Therefore, + * invert only frame sync for BCM + */ + mode |= BCM2835_I2S_FSI; + break; + default: + return -EINVAL; + } + + regmap_write(dev->i2s_regmap, BCM2835_I2S_MODE_A_REG, mode); + + /* Setup the DMA parameters */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_RXTHR(1) + | BCM2835_I2S_TXTHR(1) + | BCM2835_I2S_DMAEN, 0xffffffff); + + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_DREQ_A_REG, + BCM2835_I2S_TX_PANIC(0x10) + | BCM2835_I2S_RX_PANIC(0x30) + | BCM2835_I2S_TX(0x30) + | BCM2835_I2S_RX(0x20), 0xffffffff); + + /* Clear FIFOs */ + bcm2835_i2s_clear_fifos(dev, true, true); + + return 0; +} + +static int bcm2835_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + uint32_t cs_reg; + + bcm2835_i2s_start_clock(dev); + + /* + * Clear both FIFOs if the one that should be started + * is not empty at the moment. This should only happen + * after overrun. Otherwise, hw_params would have cleared + * the FIFO. + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &cs_reg); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK + && !(cs_reg & BCM2835_I2S_TXE)) + bcm2835_i2s_clear_fifos(dev, true, false); + else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE + && (cs_reg & BCM2835_I2S_RXD)) + bcm2835_i2s_clear_fifos(dev, false, true); + + return 0; +} + +static void bcm2835_i2s_stop(struct bcm2835_i2s_dev *dev, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + uint32_t mask; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = BCM2835_I2S_RXON; + else + mask = BCM2835_I2S_TXON; + + regmap_update_bits(dev->i2s_regmap, + BCM2835_I2S_CS_A_REG, mask, 0); + + /* Stop also the clock when not SND_SOC_DAIFMT_CONT */ + if (!dai->active && !(dev->fmt & SND_SOC_DAIFMT_CONT)) + bcm2835_i2s_stop_clock(dev); +} + +static int bcm2835_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + uint32_t mask; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + bcm2835_i2s_start_clock(dev); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = BCM2835_I2S_RXON; + else + mask = BCM2835_I2S_TXON; + + regmap_update_bits(dev->i2s_regmap, + BCM2835_I2S_CS_A_REG, mask, mask); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + bcm2835_i2s_stop(dev, substream, dai); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int bcm2835_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + if (dai->active) + return 0; + + /* Should this still be running stop it */ + bcm2835_i2s_stop_clock(dev); + + /* Enable PCM block */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_EN, BCM2835_I2S_EN); + + /* + * Disable STBY. + * Requires at least 4 PCM clock cycles to take effect. + */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_STBY, BCM2835_I2S_STBY); + + return 0; +} + +static void bcm2835_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + bcm2835_i2s_stop(dev, substream, dai); + + /* If both streams are stopped, disable module and clock */ + if (dai->active) + return; + + /* Disable the module */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_EN, 0); + + /* + * Stopping clock is necessary, because stop does + * not stop the clock when SND_SOC_DAIFMT_CONT + */ + bcm2835_i2s_stop_clock(dev); +} + +static const struct snd_soc_dai_ops bcm2835_i2s_dai_ops = { + .startup = bcm2835_i2s_startup, + .shutdown = bcm2835_i2s_shutdown, + .prepare = bcm2835_i2s_prepare, + .trigger = bcm2835_i2s_trigger, + .hw_params = bcm2835_i2s_hw_params, + .set_fmt = bcm2835_i2s_set_dai_fmt, + .set_bclk_ratio = bcm2835_i2s_set_dai_bclk_ratio +}; + +static int bcm2835_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, + &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK], + &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]); + + return 0; +} + +static struct snd_soc_dai_driver bcm2835_i2s_dai = { + .name = "bcm2835-i2s", + .probe = bcm2835_i2s_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .ops = &bcm2835_i2s_dai_ops, + .symmetric_rates = 1 +}; + +static bool bcm2835_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_I2S_CS_A_REG: + case BCM2835_I2S_FIFO_A_REG: + case BCM2835_I2S_INTSTC_A_REG: + case BCM2835_I2S_GRAY_REG: + return true; + default: + return false; + }; +} + +static bool bcm2835_i2s_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_I2S_FIFO_A_REG: + return true; + default: + return false; + }; +} + +static bool bcm2835_clk_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_CLK_PCMCTL_REG: + return true; + default: + return false; + }; +} + +static const struct regmap_config bcm2835_regmap_config[] = { + { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = BCM2835_I2S_GRAY_REG, + .precious_reg = bcm2835_i2s_precious_reg, + .volatile_reg = bcm2835_i2s_volatile_reg, + .cache_type = REGCACHE_RBTREE, + }, + { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = BCM2835_CLK_PCMDIV_REG, + .volatile_reg = bcm2835_clk_volatile_reg, + .cache_type = REGCACHE_RBTREE, + }, +}; + +static const struct snd_soc_component_driver bcm2835_i2s_component = { + .name = "bcm2835-i2s-comp", +}; + +static int bcm2835_i2s_probe(struct platform_device *pdev) +{ + struct bcm2835_i2s_dev *dev; + int i; + int ret; + struct regmap *regmap[2]; + struct resource *mem[2]; + + /* Request both ioareas */ + for (i = 0; i <= 1; i++) { + void __iomem *base; + + mem[i] = platform_get_resource(pdev, IORESOURCE_MEM, i); + base = devm_ioremap_resource(&pdev->dev, mem[i]); + if (IS_ERR(base)) + return PTR_ERR(base); + + regmap[i] = devm_regmap_init_mmio(&pdev->dev, base, + &bcm2835_regmap_config[i]); + if (IS_ERR(regmap[i])) + return PTR_ERR(regmap[i]); + } + + dev = devm_kzalloc(&pdev->dev, sizeof(*dev), + GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dev->i2s_regmap = regmap[0]; + dev->clk_regmap = regmap[1]; + + /* Set the DMA address */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = + (dma_addr_t)mem[0]->start + BCM2835_I2S_FIFO_A_REG + + BCM2835_VCMMU_SHIFT; + + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = + (dma_addr_t)mem[0]->start + BCM2835_I2S_FIFO_A_REG + + BCM2835_VCMMU_SHIFT; + + /* Set the bus width */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr_width = + DMA_SLAVE_BUSWIDTH_4_BYTES; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr_width = + DMA_SLAVE_BUSWIDTH_4_BYTES; + + /* Set burst */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].maxburst = 2; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].maxburst = 2; + + /* BCLK ratio - use default */ + dev->bclk_ratio = 0; + + /* Store the pdev */ + dev->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, dev); + + ret = devm_snd_soc_register_component(&pdev->dev, + &bcm2835_i2s_component, &bcm2835_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + return ret; + } + + ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct of_device_id bcm2835_i2s_of_match[] = { + { .compatible = "brcm,bcm2835-i2s", }, + {}, +}; + +static int bcm2835_i2s_remove(struct platform_device *pdev) +{ + snd_dmaengine_pcm_unregister(&pdev->dev); + return 0; +} + +static struct platform_driver bcm2835_i2s_driver = { + .probe = bcm2835_i2s_probe, + .remove = bcm2835_i2s_remove, + .driver = { + .name = "bcm2835-i2s", + .owner = THIS_MODULE, + .of_match_table = bcm2835_i2s_of_match, + }, +}; + +module_platform_driver(bcm2835_i2s_driver); + +MODULE_ALIAS("platform:bcm2835-i2s"); +MODULE_DESCRIPTION("BCM2835 I2S interface"); +MODULE_AUTHOR("Florian Meier "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-70-g09d2 From 4d9127faa864e7068d7e06527dfdf099ad06f64a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 20 Nov 2013 15:37:42 -0200 Subject: ASoC: soc-io: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 4f11d23f206..aa886cca3ec 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -99,14 +99,14 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, config.val_bits = data_bits; switch (control) { -#if defined(CONFIG_REGMAP_I2C) || defined(CONFIG_REGMAP_I2C_MODULE) +#if IS_ENABLED(CONFIG_REGMAP_I2C) case SND_SOC_I2C: codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev), &config); break; #endif -#if defined(CONFIG_REGMAP_SPI) || defined(CONFIG_REGMAP_SPI_MODULE) +#if IS_ENABLED(CONFIG_REGMAP_SPI) case SND_SOC_SPI: codec->control_data = regmap_init_spi(to_spi_device(codec->dev), &config); -- cgit v1.2.3-70-g09d2 From 516ea4b584332f511d3bf1b98ceabd974b1a2313 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:38 -0200 Subject: ASoC: cs4271: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f6e953454bc..ce05fd93dc7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -675,7 +675,7 @@ static struct spi_driver cs4271_spi_driver = { }; #endif /* defined(CONFIG_SPI_MASTER) */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static const struct i2c_device_id cs4271_i2c_id[] = { {"cs4271", 0}, {} @@ -728,7 +728,7 @@ static struct i2c_driver cs4271_i2c_driver = { .probe = cs4271_i2c_probe, .remove = cs4271_i2c_remove, }; -#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ +#endif /* IS_ENABLED(CONFIG_I2C) */ /* * We only register our serial bus driver here without @@ -741,7 +741,7 @@ static int __init cs4271_modinit(void) { int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&cs4271_i2c_driver); if (ret) { pr_err("Failed to register CS4271 I2C driver: %d\n", ret); @@ -767,7 +767,7 @@ static void __exit cs4271_modexit(void) spi_unregister_driver(&cs4271_spi_driver); #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&cs4271_i2c_driver); #endif } -- cgit v1.2.3-70-g09d2 From 25c1a63f43ca40f1581c076b7f7618297ef1cbba Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:40 -0200 Subject: ASoC: da7210: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 9c123145650..8166dcb2e4a 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1188,7 +1188,7 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static struct reg_default da7210_regmap_i2c_patch[] = { @@ -1362,7 +1362,7 @@ static struct spi_driver da7210_spi_driver = { static int __init da7210_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&da7210_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) @@ -1378,7 +1378,7 @@ module_init(da7210_modinit); static void __exit da7210_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&da7210_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From e42644c64c0de145ab2041ceb904e44421dd6794 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:42 -0200 Subject: ASoC: ssm2602: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 492644e67ac..480074d8647 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -730,7 +730,7 @@ static struct spi_driver ssm2602_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) /* * ssm2602 2 wire address is determined by GPIO5 * state during powerup. @@ -797,7 +797,7 @@ static int __init ssm2602_modinit(void) return ret; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&ssm2602_i2c_driver); if (ret) return ret; @@ -813,7 +813,7 @@ static void __exit ssm2602_exit(void) spi_unregister_driver(&ssm2602_spi_driver); #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&ssm2602_i2c_driver); #endif } -- cgit v1.2.3-70-g09d2 From b65ab73e5d624eb4a88bc6094a3627007cb92500 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:43 -0200 Subject: ASoC: wm8731: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 456bb8c6d75..6117107ea56 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -732,7 +732,7 @@ static struct spi_driver wm8731_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8731_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -791,7 +791,7 @@ static struct i2c_driver wm8731_i2c_driver = { static int __init wm8731_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8731_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8731 I2C driver: %d\n", @@ -811,7 +811,7 @@ module_init(wm8731_modinit); static void __exit wm8731_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8731_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From 26090a834b49673945458b185be0afa03c2737fe Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:45 -0200 Subject: ASoC: wm8741: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8741.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index b18813cc7ba..2895c8d3b5e 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -500,7 +500,7 @@ static const struct regmap_config wm8741_regmap = { .readable_reg = wm8741_readable, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -617,7 +617,7 @@ static int __init wm8741_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8741_i2c_driver); if (ret != 0) pr_err("Failed to register WM8741 I2C driver: %d\n", ret); @@ -639,7 +639,7 @@ static void __exit wm8741_exit(void) #if defined(CONFIG_SPI_MASTER) spi_unregister_driver(&wm8741_spi_driver); #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8741_i2c_driver); #endif } -- cgit v1.2.3-70-g09d2 From 9ea6fbc66d15c83089e177b445872a9ba40f125d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:46 -0200 Subject: ASoC: wm8750: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8750.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 50d5ff61623..78616a638a5 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -816,7 +816,7 @@ static struct spi_driver wm8750_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8750_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -868,7 +868,7 @@ static struct i2c_driver wm8750_i2c_driver = { static int __init wm8750_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8750_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8750 I2C driver: %d\n", @@ -888,7 +888,7 @@ module_init(wm8750_modinit); static void __exit wm8750_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8750_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From 2c4864334c4d9a23fa810638ad27e80ea0ceb9a4 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:47 -0200 Subject: ASoC: wm8753: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d96ebf52d95..be85da93a26 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1596,7 +1596,7 @@ static struct spi_driver wm8753_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8753_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1653,7 +1653,7 @@ static struct i2c_driver wm8753_i2c_driver = { static int __init wm8753_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8753_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8753 I2C driver: %d\n", @@ -1673,7 +1673,7 @@ module_init(wm8753_modinit); static void __exit wm8753_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8753_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From b888edbc68fbace3101cb092c6910476e85ae922 Mon Sep 17 00:00:00 2001 From: wangbiao Date: Fri, 22 Nov 2013 10:44:30 +0800 Subject: ASoC: wm8994: Move DCS done IRQ request later once code return from request_threaded_irq, irq was setup enabled by default, but completion var dcs_done not got initialized yet, if then a dcs done irq was raised, system will got hung as the sync mechanism is invalid now. so this patch move dcs done irq request to the end of initialization of completion. Signed-off-by: wang, biao Signed-off-by: Zhang, Di Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 86426a117b0..b9be9cbc460 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4077,12 +4077,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, wm8994_temp_shut, "Thermal shutdown", codec); - ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE, - wm_hubs_dcs_done, "DC servo done", - &wm8994->hubs); - if (ret == 0) - wm8994->hubs.dcs_done_irq = true; - switch (control->type) { case WM8994: if (wm8994->micdet_irq) { @@ -4313,6 +4307,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } wm_hubs_add_analogue_routes(codec, 0, 0); + ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE, + wm_hubs_dcs_done, "DC servo done", + &wm8994->hubs); + if (ret == 0) + wm8994->hubs.dcs_done_irq = true; snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); switch (control->type) { -- cgit v1.2.3-70-g09d2 From 1769267bb01303ac73b48535454461819ef1dcc2 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:48 -0200 Subject: ASoC: wm8776: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 942d58e455f..ef824672523 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -532,7 +532,7 @@ static struct spi_driver wm8776_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8776_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -584,7 +584,7 @@ static struct i2c_driver wm8776_i2c_driver = { static int __init wm8776_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8776_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8776 I2C driver: %d\n", @@ -604,7 +604,7 @@ module_init(wm8776_modinit); static void __exit wm8776_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8776_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From f3f9a60f7947b6bd2f970d5680dd3df624405027 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:49 -0200 Subject: ASoC: wm8804: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e119c..9bc8206a680 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -739,7 +739,7 @@ static struct spi_driver wm8804_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8804_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -791,7 +791,7 @@ static int __init wm8804_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8804_i2c_driver); if (ret) { printk(KERN_ERR "Failed to register wm8804 I2C driver: %d\n", @@ -811,7 +811,7 @@ module_init(wm8804_modinit); static void __exit wm8804_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8804_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From f25cf34969823ab7197ce9ff2521c33f0141075b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:50 -0200 Subject: ASoC: wm8900: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 734209e252c..e98bc7038a0 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1288,7 +1288,7 @@ static struct spi_driver wm8900_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8900_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1338,7 +1338,7 @@ static struct i2c_driver wm8900_i2c_driver = { static int __init wm8900_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8900_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8900 I2C driver: %d\n", @@ -1358,7 +1358,7 @@ module_init(wm8900_modinit); static void __exit wm8900_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8900_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From 50c9697320434d1489d087cbf38f7907a9894609 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:52 -0200 Subject: ASoC: wm8985: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 18f2babe109..271b517911a 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1148,7 +1148,7 @@ static struct spi_driver wm8985_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8985_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1201,7 +1201,7 @@ static int __init wm8985_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8985_i2c_driver); if (ret) { printk(KERN_ERR "Failed to register wm8985 I2C driver: %d\n", @@ -1221,7 +1221,7 @@ module_init(wm8985_modinit); static void __exit wm8985_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8985_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From 63587116811bd23d22693b50447a2a356602e70b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:53 -0200 Subject: ASoC: wm8988: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 39b9acceb59..a55e1c2c382 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -912,7 +912,7 @@ static struct spi_driver wm8988_spi_driver = { }; #endif /* CONFIG_SPI_MASTER */ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8988_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -964,7 +964,7 @@ static struct i2c_driver wm8988_i2c_driver = { static int __init wm8988_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8988_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8988 I2C driver: %d\n", @@ -984,7 +984,7 @@ module_init(wm8988_modinit); static void __exit wm8988_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8988_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From 3f3002692ce8fa1e9b257183ea1a36baacfdcfcf Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:48:32 -0200 Subject: AsoC: wm9081: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 630b3d776ec..0982c1d38ec 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1326,7 +1326,7 @@ static const struct regmap_config wm9081_regmap = { .cache_type = REGCACHE_RBTREE, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm9081_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { -- cgit v1.2.3-70-g09d2 From 9a199b8e9933edf83585bac2c9030870e014381b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:48:32 -0200 Subject: ASoC: wm9081: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 630b3d776ec..0982c1d38ec 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1326,7 +1326,7 @@ static const struct regmap_config wm9081_regmap = { .cache_type = REGCACHE_RBTREE, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm9081_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { -- cgit v1.2.3-70-g09d2 From 7ae10ed2ee757f2ce19188e540eaa44f337c7cd2 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:39 -0200 Subject: ASoC: cs42l52: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 8b427c97708..19ee10b6d6c 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -50,7 +50,7 @@ struct cs42l52_private { u8 mclksel; u32 mclk; u8 flags; -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; @@ -953,7 +953,7 @@ static int cs42l52_resume(struct snd_soc_codec *codec) return 0; } -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) static int beep_rates[] = { 261, 522, 585, 667, 706, 774, 889, 1000, 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 -- cgit v1.2.3-70-g09d2 From a3d36bc2aba531328f7311ef57dec7687283ec57 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Wed, 13 Nov 2013 16:05:40 -0600 Subject: ASoC: cs42l52: Reorganize MICA/B Config and Select This patch reworks the MICA an MICB config for single-ended or differential and the selection of which MIC for the single config Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- include/sound/cs42l52.h | 6 ------ sound/soc/codecs/cs42l52.c | 25 ++++--------------------- 2 files changed, 4 insertions(+), 27 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index 7c2be4a5189..daa91f327e4 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -22,12 +22,6 @@ struct cs42l52_platform_data { /* MICB mode selection 0=Single 1=Differential */ unsigned int micb_cfg; - /* MICA Select 0=MIC1A 1=MIC2A */ - unsigned int mica_sel; - - /* MICB Select 0=MIC2A 1=MIC2B */ - unsigned int micb_sel; - /* Charge Pump Freq. Check datasheet Pg73 */ unsigned int chgfreq; diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 19ee10b6d6c..18010639d0c 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -233,7 +233,7 @@ static const struct soc_enum mic_bias_level_enum = SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0, ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); -static const char * const cs42l52_mic_text[] = { "Single", "Differential" }; +static const char * const cs42l52_mic_text[] = { "MIC1", "MIC2" }; static const struct soc_enum mica_enum = SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5, @@ -243,12 +243,6 @@ static const struct soc_enum micb_enum = SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5, ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); -static const struct snd_kcontrol_new mica_mux = - SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum); - -static const struct snd_kcontrol_new micb_mux = - SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum); - static const char * const digital_output_mux_text[] = {"ADC", "DSP"}; static const struct soc_enum digital_output_mux_enum = @@ -425,6 +419,9 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), + SOC_ENUM("MICA Select", mica_enum), + SOC_ENUM("MICB Select", micb_enum), + SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL, CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv), @@ -550,9 +547,6 @@ static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIFOUTR", NULL, 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux), - SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux), - SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1), SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1), SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0), @@ -1239,17 +1233,6 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, cs42l52->pdata.micb_cfg << CS42L52_MIC_CTL_TYPE_SHIFT); - if (cs42l52->pdata.mica_sel) - regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL, - CS42L52_MIC_CTL_MIC_SEL_MASK, - cs42l52->pdata.mica_sel << - CS42L52_MIC_CTL_MIC_SEL_SHIFT); - if (cs42l52->pdata.micb_sel) - regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL, - CS42L52_MIC_CTL_MIC_SEL_MASK, - cs42l52->pdata.micb_sel << - CS42L52_MIC_CTL_MIC_SEL_SHIFT); - if (cs42l52->pdata.chgfreq) regmap_update_bits(cs42l52->regmap, CS42L52_CHARGE_PUMP, CS42L52_CHARGE_PUMP_MASK, -- cgit v1.2.3-70-g09d2 From 44b2ed54036ecec36ad27adf356f0274a72e5f05 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 14 Nov 2013 11:46:11 -0600 Subject: ASoC: cs42l52: Make MICA/B mixer dependent on mic config MICA/B Single-Ended input selection depends on mica/b config so lets make the mixer controls for them only show for selected mic's Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- include/sound/cs42l52.h | 8 ++++---- sound/soc/codecs/cs42l52.c | 37 ++++++++++++++++++++++++++++++------- 2 files changed, 34 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index daa91f327e4..bbabf84bdb4 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -16,11 +16,11 @@ struct cs42l52_platform_data { /* MICBIAS Level. Check datasheet Pg48 */ unsigned int micbias_lvl; - /* MICA mode selection 0=Single 1=Differential */ - unsigned int mica_cfg; + /* MICA mode selection Differential or Single-ended */ + bool mica_diff_cfg; - /* MICB mode selection 0=Single 1=Differential */ - unsigned int micb_cfg; + /* MICB mode selection Differential or Single-ended */ + bool micb_diff_cfg; /* Charge Pump Freq. Check datasheet Pg73 */ unsigned int chgfreq; diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 18010639d0c..78d2dd669e8 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -419,9 +419,6 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), - SOC_ENUM("MICA Select", mica_enum), - SOC_ENUM("MICB Select", micb_enum), - SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL, CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv), @@ -528,6 +525,30 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { }; +static const struct snd_kcontrol_new cs42l52_mica_controls[] = { + SOC_ENUM("MICA Select", mica_enum), +}; + +static const struct snd_kcontrol_new cs42l52_micb_controls[] = { + SOC_ENUM("MICB Select", micb_enum), +}; + +static int cs42l52_add_mic_controls(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + struct cs42l52_platform_data *pdata = &cs42l52->pdata; + + if (!pdata->mica_diff_cfg) + snd_soc_add_codec_controls(codec, cs42l52_mica_controls, + ARRAY_SIZE(cs42l52_mica_controls)); + + if (!pdata->micb_diff_cfg) + snd_soc_add_codec_controls(codec, cs42l52_micb_controls, + ARRAY_SIZE(cs42l52_micb_controls)); + + return 0; +} + static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = { SND_SOC_DAPM_INPUT("AIN1L"), @@ -1104,6 +1125,8 @@ static int cs42l52_probe(struct snd_soc_codec *codec) } regcache_cache_only(cs42l52->regmap, true); + cs42l52_add_mic_controls(codec); + cs42l52_init_beep(codec); cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1221,16 +1244,16 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, reg & 0xFF); /* Set Platform Data */ - if (cs42l52->pdata.mica_cfg) + if (cs42l52->pdata.mica_diff_cfg) regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL, CS42L52_MIC_CTL_TYPE_MASK, - cs42l52->pdata.mica_cfg << + cs42l52->pdata.mica_diff_cfg << CS42L52_MIC_CTL_TYPE_SHIFT); - if (cs42l52->pdata.micb_cfg) + if (cs42l52->pdata.micb_diff_cfg) regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL, CS42L52_MIC_CTL_TYPE_MASK, - cs42l52->pdata.micb_cfg << + cs42l52->pdata.micb_diff_cfg << CS42L52_MIC_CTL_TYPE_SHIFT); if (cs42l52->pdata.chgfreq) -- cgit v1.2.3-70-g09d2 From 391fc59db87615e07e8a6ab5fbffe3cc04f2b19c Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 15 Nov 2013 09:35:33 -0600 Subject: ASoC: cs42l52: Add devicetree support for CS42L52 This patch adds device tree support for the CS42L52 Codec Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 45 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 42 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 78d2dd669e8..4a47a634f37 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -17,7 +17,7 @@ #include #include #include -#include +#include #include #include #include @@ -1193,6 +1193,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, int ret; unsigned int devid = 0; unsigned int reg; + u32 val32; cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private), GFP_KERNEL); @@ -1206,9 +1207,39 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); return ret; } - - if (pdata) + if (pdata) { + cs42l52->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs42l52_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + if (of_property_read_bool(i2c_client->dev.of_node, + "cirrus,mica-differential-cfg")) + pdata->mica_diff_cfg = true; + + if (of_property_read_bool(i2c_client->dev.of_node, + "cirrus,micb-differential-cfg")) + pdata->micb_diff_cfg = true; + + if (of_property_read_u32(i2c_client->dev.of_node, + "cirrus,micbias-lvl", &val32) >= 0) + pdata->micbias_lvl = val32; + + if (of_property_read_u32(i2c_client->dev.of_node, + "cirrus,chgfreq-divisor", &val32) >= 0) + pdata->chgfreq_divisor = val32; + + pdata->reset_gpio = + of_get_named_gpio(i2c_client->dev.of_node, + "cirrus,reset-gpio", 0); + } cs42l52->pdata = *pdata; + } if (cs42l52->pdata.reset_gpio) { ret = gpio_request_one(cs42l52->pdata.reset_gpio, @@ -1280,6 +1311,13 @@ static int cs42l52_i2c_remove(struct i2c_client *client) return 0; } +static const struct of_device_id cs42l52_of_match[] = { + { .compatible = "cirrus,cs42l52", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs42l52_of_match); + + static const struct i2c_device_id cs42l52_id[] = { { "cs42l52", 0 }, { } @@ -1290,6 +1328,7 @@ static struct i2c_driver cs42l52_i2c_driver = { .driver = { .name = "cs42l52", .owner = THIS_MODULE, + .of_match_table = cs42l52_of_match, }, .id_table = cs42l52_id, .probe = cs42l52_i2c_probe, -- cgit v1.2.3-70-g09d2 From 21585ee848078b12d0d1a513e93936bf96b444a0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 28 Nov 2013 08:50:32 +0100 Subject: ASoC: Add resource managed snd_dmaengine_pcm_register() For many drivers using the generic dmaengine PCM driver one of the few (or the only) things left to do in the drivers remove function is to unregister the PCM device. This patch adds a resource managed version of snd_dmaengine_pcm_register() which makes it possible to simplify the remove function as well as the error path in the probe function for those drivers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/dmaengine_pcm.h | 4 ++++ sound/soc/soc-devres.c | 41 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 45 insertions(+) (limited to 'sound/soc') diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 15017311f2e..4ef986cab18 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -140,6 +140,10 @@ int snd_dmaengine_pcm_register(struct device *dev, unsigned int flags); void snd_dmaengine_pcm_unregister(struct device *dev); +int devm_snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, + unsigned int flags); + int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index b1d732255c0..999861942d2 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -12,6 +12,7 @@ #include #include #include +#include static void devm_component_release(struct device *dev, void *res) { @@ -84,3 +85,43 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) return ret; } EXPORT_SYMBOL_GPL(devm_snd_soc_register_card); + +#ifdef CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM + +static void devm_dmaengine_pcm_release(struct device *dev, void *res) +{ + snd_dmaengine_pcm_unregister(*(struct device **)res); +} + +/** + * devm_snd_dmaengine_pcm_register - resource managed dmaengine PCM registration + * @dev: The parent device for the PCM device + * @config: Platform specific PCM configuration + * @flags: Platform specific quirks + * + * Register a dmaengine based PCM device with automatic unregistration when the + * device is unregistered. + */ +int devm_snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, unsigned int flags) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_dmaengine_pcm_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_dmaengine_pcm_register(dev, config, flags); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_dmaengine_pcm_register); + +#endif -- cgit v1.2.3-70-g09d2 From d71b3ef44f9e5cfda2499768f6420b784845af06 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 28 Nov 2013 08:50:36 +0100 Subject: ASoC: spear: Use devm_snd_dmaengine_pcm_register Makes the code slightly shorter. Signed-off-by: Lars-Peter Clausen Acked-by: Rajeev Kumar Signed-off-by: Mark Brown --- sound/soc/spear/spear_pcm.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 4707f2b862c..9a02141666e 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -49,18 +49,12 @@ static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = { static int spear_soc_platform_probe(struct platform_device *pdev) { - return snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(&pdev->dev, &spear_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } -static int spear_soc_platform_remove(struct platform_device *pdev) -{ - snd_dmaengine_pcm_unregister(&pdev->dev); - return 0; -} - static struct platform_driver spear_pcm_driver = { .driver = { .name = "spear-pcm-audio", @@ -68,7 +62,6 @@ static struct platform_driver spear_pcm_driver = { }, .probe = spear_soc_platform_probe, - .remove = spear_soc_platform_remove, }; module_platform_driver(spear_pcm_driver); -- cgit v1.2.3-70-g09d2 From d733dc0828cfb230171ae7420a6e8c344ec8473a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 28 Nov 2013 16:37:51 +0000 Subject: ASoC: wm_adsp: Stop region iteration when the desired region is found When locating the memory region relating to a coefficient block written through a bin file we keep processing the list of regions even after we have found the region we require. This patch adds a break, so we don't process redundant list items. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 46ec0e9744d..b42f9af163c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1286,6 +1286,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) reg = wm_adsp_region_to_reg(mem, reg); reg += offset; + break; } } -- cgit v1.2.3-70-g09d2 From 17b9a2b78586c42916a2bfc55ea6c0ef962b2c1e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:42:49 -0800 Subject: ASoC: rcar: remove unused register settings AUDIO_CLK_SEL4/5 are not used Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 3 --- sound/soc/sh/rcar/rsnd.h | 3 --- 2 files changed, 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 61212ee97c2..b94d4cefa8b 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -208,9 +208,6 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) RSND_GEN1_S_REG(gen, ADG, SSICKR, 0x08), RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL3, 0x18), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL4, 0x1c), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL5, 0x20), RSND_GEN1_M_REG(gen, SSI, SSICR, 0x00, 0x40), RSND_GEN1_M_REG(gen, SSI, SSISR, 0x04, 0x40), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index b5ac3a2afc5..63a9d7081c6 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -49,9 +49,6 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL0, RSND_REG_AUDIO_CLK_SEL1, RSND_REG_AUDIO_CLK_SEL2, - RSND_REG_AUDIO_CLK_SEL3, - RSND_REG_AUDIO_CLK_SEL4, - RSND_REG_AUDIO_CLK_SEL5, /* SSI */ RSND_REG_SSICR, -- cgit v1.2.3-70-g09d2 From c1e6cc5e577d1d446a645aea02d28a924f20b834 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:43:01 -0800 Subject: ASoC: rcar: separate regmap init common field The repmap initialization difference between Gen1/Gen2 is only register offset. This patch separates rsnd_gen1_regmap_init() into common part and Gen1 specific part. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 49 ++++++++++++++++++++++++++++--------------------- 1 file changed, 28 insertions(+), 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index b94d4cefa8b..0ebea44e890 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -115,6 +115,33 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, mask, data); } +static int rsnd_gen_regmap_init(struct rsnd_priv *priv, + struct rsnd_gen *gen, + struct reg_field *regf) +{ + int i; + struct device *dev = rsnd_priv_to_dev(priv); + struct regmap_config regc; + + memset(®c, 0, sizeof(regc)); + regc.reg_bits = 32; + regc.val_bits = 32; + + gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, ®c); + if (IS_ERR(gen->regmap)) { + dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap)); + return PTR_ERR(gen->regmap); + } + + for (i = 0; i < RSND_REG_MAX; i++) { + gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]); + if (IS_ERR(gen->regs[i])) + return PTR_ERR(gen->regs[i]); + + } + + return 0; +} /* * Gen2 * will be filled in the future @@ -189,9 +216,6 @@ static int rsnd_gen1_path_exit(struct rsnd_priv *priv, static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) { - int i; - struct device *dev = rsnd_priv_to_dev(priv); - struct regmap_config regc; struct reg_field regf[RSND_REG_MAX] = { RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_SEL, 0x00), RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL0, 0x08), @@ -216,24 +240,7 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) RSND_GEN1_M_REG(gen, SSI, SSIWSR, 0x20, 0x40), }; - memset(®c, 0, sizeof(regc)); - regc.reg_bits = 32; - regc.val_bits = 32; - - gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, ®c); - if (IS_ERR(gen->regmap)) { - dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap)); - return PTR_ERR(gen->regmap); - } - - for (i = 0; i < RSND_REG_MAX; i++) { - gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]); - if (IS_ERR(gen->regs[i])) - return PTR_ERR(gen->regs[i]); - - } - - return 0; + return rsnd_gen_regmap_init(priv, gen, regf); } static int rsnd_gen1_probe(struct platform_device *pdev, -- cgit v1.2.3-70-g09d2 From 42ee5d22e3d2550a49bc5d3e6f19c92da9a19446 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:43:13 -0800 Subject: ASoC: rcar: add rsnd_is_accessible_reg() Current rcar driver is supporting Gen1, and Gen2 will be supported soon. Then, some registers are used from Gen1 only, or from Gen2 only. To avoid NULL pointer access, this patch adds register accessible check function. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 0ebea44e890..970439d5ec0 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -86,12 +86,28 @@ static struct regmap_bus rsnd_regmap_bus = { .val_format_endian_default = REGMAP_ENDIAN_NATIVE, }; +static int rsnd_is_accessible_reg(struct rsnd_priv *priv, + struct rsnd_gen *gen, enum rsnd_reg reg) +{ + if (!gen->regs[reg]) { + struct device *dev = rsnd_priv_to_dev(priv); + + dev_err(dev, "unsupported register access %x\n", reg); + return 0; + } + + return 1; +} + u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg) { struct rsnd_gen *gen = rsnd_priv_to_gen(priv); u32 val; + if (!rsnd_is_accessible_reg(priv, gen, reg)) + return 0; + regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); return val; @@ -103,6 +119,9 @@ void rsnd_write(struct rsnd_priv *priv, { struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + if (!rsnd_is_accessible_reg(priv, gen, reg)) + return; + regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); } @@ -111,6 +130,9 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, { struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + if (!rsnd_is_accessible_reg(priv, gen, reg)) + return; + regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod), mask, data); } @@ -134,6 +156,10 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, } for (i = 0; i < RSND_REG_MAX; i++) { + gen->regs[i] = NULL; + if (!regf[i].reg) + continue; + gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]); if (IS_ERR(gen->regs[i])) return PTR_ERR(gen->regs[i]); -- cgit v1.2.3-70-g09d2 From 994a9df1e3ea3244e94309e4f893e9a5121116c9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:43:23 -0800 Subject: ASoC: rcar: remove .path_init/exit from rsnd_gen_ops rsnd_gen_ops has .path_init/exit callback function which cares SRU/SSI (if Gen1) SCU/SSIU/SSI (if Gen2) path settings. But, the differences between Gen1/Gen2 are cared in ssi.c/scu.c, and the path itself is same in Gen1/Gen2. This patch removes .path_init/exit callback. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 54 ++++++++++++++----------------------------------- 1 file changed, 15 insertions(+), 39 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 970439d5ec0..4f2eaa3262d 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -16,12 +16,6 @@ struct rsnd_gen_ops { struct rsnd_priv *priv); void (*remove)(struct platform_device *pdev, struct rsnd_priv *priv); - int (*path_init)(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io); - int (*path_exit)(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io); }; struct rsnd_gen { @@ -168,17 +162,10 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, return 0; } -/* - * Gen2 - * will be filled in the future - */ -/* - * Gen1 - */ -static int rsnd_gen1_path_init(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) +int rsnd_gen_path_init(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) { struct rsnd_mod *mod; int ret; @@ -216,9 +203,9 @@ static int rsnd_gen1_path_init(struct rsnd_priv *priv, return ret; } -static int rsnd_gen1_path_exit(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) +int rsnd_gen_path_exit(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) { struct rsnd_mod *mod, *n; int ret = 0; @@ -232,6 +219,15 @@ static int rsnd_gen1_path_exit(struct rsnd_priv *priv, return ret; } +/* + * Gen2 + * will be filled in the future + */ + +/* + * Gen1 + */ + /* single address mapping */ #define RSND_GEN1_S_REG(gen, reg, id, offset) \ RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, 0, 9) @@ -319,31 +315,11 @@ static void rsnd_gen1_remove(struct platform_device *pdev, static struct rsnd_gen_ops rsnd_gen1_ops = { .probe = rsnd_gen1_probe, .remove = rsnd_gen1_remove, - .path_init = rsnd_gen1_path_init, - .path_exit = rsnd_gen1_path_exit, }; /* * Gen */ -int rsnd_gen_path_init(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) -{ - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - - return gen->ops->path_init(priv, rdai, io); -} - -int rsnd_gen_path_exit(struct rsnd_priv *priv, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) -{ - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - - return gen->ops->path_exit(priv, rdai, io); -} - int rsnd_gen_probe(struct platform_device *pdev, struct rcar_snd_info *info, struct rsnd_priv *priv) -- cgit v1.2.3-70-g09d2 From 531eaf491e25ce215bbcf434e23e77f53fc98171 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:43:34 -0800 Subject: ASoC: rcar: remove rcar_gen_ops Current rcar driver gen.c is using rcar_gen_ops which was made with the assumption that Gen1 and Gen2 need different behavior. but it was not needed. This patch removes unnecessary complex method. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 33 ++++++--------------------------- 1 file changed, 6 insertions(+), 27 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 4f2eaa3262d..a29e36eb1a3 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -10,14 +10,6 @@ */ #include "rsnd.h" -struct rsnd_gen_ops { - int (*probe)(struct platform_device *pdev, - struct rcar_snd_info *info, - struct rsnd_priv *priv); - void (*remove)(struct platform_device *pdev, - struct rsnd_priv *priv); -}; - struct rsnd_gen { void __iomem *base[RSND_BASE_MAX]; @@ -307,16 +299,6 @@ static int rsnd_gen1_probe(struct platform_device *pdev, } -static void rsnd_gen1_remove(struct platform_device *pdev, - struct rsnd_priv *priv) -{ -} - -static struct rsnd_gen_ops rsnd_gen1_ops = { - .probe = rsnd_gen1_probe, - .remove = rsnd_gen1_remove, -}; - /* * Gen */ @@ -326,6 +308,7 @@ int rsnd_gen_probe(struct platform_device *pdev, { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen; + int ret; gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL); if (!gen) { @@ -333,23 +316,19 @@ int rsnd_gen_probe(struct platform_device *pdev, return -ENOMEM; } - if (rsnd_is_gen1(priv)) - gen->ops = &rsnd_gen1_ops; + priv->gen = gen; - if (!gen->ops) { + if (rsnd_is_gen1(priv)) { + ret = rsnd_gen1_probe(pdev, info, priv); + } else { dev_err(dev, "unknown generation R-Car sound device\n"); return -ENODEV; } - priv->gen = gen; - - return gen->ops->probe(pdev, info, priv); + return ret; } void rsnd_gen_remove(struct platform_device *pdev, struct rsnd_priv *priv) { - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - - gen->ops->remove(pdev, priv); } -- cgit v1.2.3-70-g09d2 From 507d466c733e28d132a7be87040a3da126df7947 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Nov 2013 18:43:45 -0800 Subject: ASoC: rcar: add Gen2 sound support This patch adds Gen2 sound support for Renesas R-Car. But, it is supporting PIO transfer only at this point Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 91 +++++++++++++++++++++++++++++++++++++++++++++--- sound/soc/sh/rcar/rsnd.h | 3 +- sound/soc/sh/rcar/ssi.c | 3 ++ 3 files changed, 91 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index a29e36eb1a3..bf066f73ef0 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -213,9 +213,88 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, /* * Gen2 - * will be filled in the future */ +/* single address mapping */ +#define RSND_GEN2_S_REG(gen, reg, id, offset) \ + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 9) + +/* multi address mapping */ +#define RSND_GEN2_M_REG(gen, reg, id, offset, _id_offset) \ + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 9) + +static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) +{ + struct reg_field regf[RSND_REG_MAX] = { + RSND_GEN2_S_REG(gen, SSIU, SSI_MODE0, 0x800), + RSND_GEN2_S_REG(gen, SSIU, SSI_MODE1, 0x804), + /* FIXME: it needs SSI_MODE2/3 in the future */ + RSND_GEN2_M_REG(gen, SSIU, INT_ENABLE, 0x18, 0x80), + + RSND_GEN2_S_REG(gen, ADG, BRRA, 0x00), + RSND_GEN2_S_REG(gen, ADG, BRRB, 0x04), + RSND_GEN2_S_REG(gen, ADG, SSICKR, 0x08), + RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), + RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), + RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL2, 0x14), + + RSND_GEN2_M_REG(gen, SSI, SSICR, 0x00, 0x40), + RSND_GEN2_M_REG(gen, SSI, SSISR, 0x04, 0x40), + RSND_GEN2_M_REG(gen, SSI, SSITDR, 0x08, 0x40), + RSND_GEN2_M_REG(gen, SSI, SSIRDR, 0x0c, 0x40), + RSND_GEN2_M_REG(gen, SSI, SSIWSR, 0x20, 0x40), + }; + + return rsnd_gen_regmap_init(priv, gen, regf); +} + +static int rsnd_gen2_probe(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + struct resource *scu_res; + struct resource *adg_res; + struct resource *ssiu_res; + struct resource *ssi_res; + int ret; + + /* + * map address + */ + scu_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SCU); + adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_ADG); + ssiu_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSIU); + ssi_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSI); + + gen->base[RSND_GEN2_SCU] = devm_ioremap_resource(dev, scu_res); + gen->base[RSND_GEN2_ADG] = devm_ioremap_resource(dev, adg_res); + gen->base[RSND_GEN2_SSIU] = devm_ioremap_resource(dev, ssiu_res); + gen->base[RSND_GEN2_SSI] = devm_ioremap_resource(dev, ssi_res); + if (IS_ERR(gen->base[RSND_GEN2_SCU]) || + IS_ERR(gen->base[RSND_GEN2_ADG]) || + IS_ERR(gen->base[RSND_GEN2_SSIU]) || + IS_ERR(gen->base[RSND_GEN2_SSI])) + return -ENODEV; + + ret = rsnd_gen2_regmap_init(priv, gen); + if (ret < 0) + return ret; + + dev_dbg(dev, "Gen2 device probed\n"); + dev_dbg(dev, "SRU : %08x => %p\n", scu_res->start, + gen->base[RSND_GEN2_SCU]); + dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + gen->base[RSND_GEN2_ADG]); + dev_dbg(dev, "SSIU : %08x => %p\n", ssiu_res->start, + gen->base[RSND_GEN2_SSIU]); + dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + gen->base[RSND_GEN2_SSI]); + + return 0; +} + /* * Gen1 */ @@ -318,12 +397,14 @@ int rsnd_gen_probe(struct platform_device *pdev, priv->gen = gen; - if (rsnd_is_gen1(priv)) { + ret = -ENODEV; + if (rsnd_is_gen1(priv)) ret = rsnd_gen1_probe(pdev, info, priv); - } else { + else if (rsnd_is_gen2(priv)) + ret = rsnd_gen2_probe(pdev, info, priv); + + if (ret < 0) dev_err(dev, "unknown generation R-Car sound device\n"); - return -ENODEV; - } return ret; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 63a9d7081c6..bff7b9e5306 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -31,7 +31,7 @@ * see gen1/gen2 for detail */ enum rsnd_reg { - /* SRU/SCU */ + /* SRU/SCU/SSIU */ RSND_REG_SRC_ROUTE_SEL, RSND_REG_SRC_TMG_SEL0, RSND_REG_SRC_TMG_SEL1, @@ -41,6 +41,7 @@ enum rsnd_reg { RSND_REG_SSI_MODE1, RSND_REG_BUSIF_MODE, RSND_REG_BUSIF_ADINR, + RSND_REG_INT_ENABLE, /* ADG */ RSND_REG_BRRA, diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index b71cf9d7dd3..477465f9507 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -457,6 +457,9 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod, /* enable PIO IRQ */ ssi->cr_etc = UIEN | OIEN | DIEN; + /* enable PIO interrupt */ + rsnd_mod_write(&ssi->mod, INT_ENABLE, 0x0f000000); + rsnd_ssi_hw_start(ssi, rdai, io); dev_dbg(dev, "%s.%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); -- cgit v1.2.3-70-g09d2 From 69ae8489076fa0fa98609155434f3c286c7364a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Nov 2013 11:42:18 +0000 Subject: ASoC: cs42l52: Fix build Reported-by: Stephen Rothwell Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4a47a634f37..0bac6d5a4ac 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1232,7 +1232,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, if (of_property_read_u32(i2c_client->dev.of_node, "cirrus,chgfreq-divisor", &val32) >= 0) - pdata->chgfreq_divisor = val32; + pdata->chgfreq = val32; pdata->reset_gpio = of_get_named_gpio(i2c_client->dev.of_node, -- cgit v1.2.3-70-g09d2 From 4ded61eb3ea87e9c563e09662be3ed5e942ff2a2 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 2 Dec 2013 00:41:24 -0200 Subject: ASoC: imx-spdif: Remove error message upon devm_kzalloc() failure No need to have a specific OOM message, since there is generic MM out of memory message in place. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-spdif.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 8499d5292f0..980dd1fc911 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -35,7 +35,6 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { - dev_err(&pdev->dev, "failed to allocate memory\n"); ret = -ENOMEM; goto end; } -- cgit v1.2.3-70-g09d2 From a22f33b00346f26d29483cdacdbf26df7947ef23 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 30 Nov 2013 18:00:45 +0100 Subject: ASoC: generic-dmaengine-pcm: Set BATCH flag when residue reporting is not supported For dmaengine drivers which do not support transfer residue reporting we update the PCM pointer with period granularity. Set the SNDRV_PCM_INFO_BATCH flag in this case to let userspace know about this. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index cbc9c96ce1f..87e86357124 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -137,6 +137,9 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea hw.buffer_bytes_max = SIZE_MAX; hw.fifo_size = dma_data->fifo_size; + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) + hw.info |= SNDRV_PCM_INFO_BATCH; + ret = dma_get_slave_caps(chan, &dma_caps); if (ret == 0) { if (dma_caps.cmd_pause) -- cgit v1.2.3-70-g09d2 From 62e5f676f6a063e1ab0d6b8fcaef2feb026ee00e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 30 Nov 2013 17:38:58 +0100 Subject: ASoC: Set SNDRV_PCM_INFO_JOINT_DUPLEX for PCMs with symmetry constraints If there are symmetry constraints between the playback and the capture channel set the SNDRV_PCM_INFO_JOINT_DUPLEX flag to let userspace know about this. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 170ff969575..f3592f14283 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -180,6 +180,21 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, return 0; } +static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver; + struct snd_soc_dai_driver *codec_driver = rtd->codec_dai->driver; + struct snd_soc_dai_link *link = rtd->dai_link; + + return cpu_driver->symmetric_rates || codec_driver->symmetric_rates || + link->symmetric_rates || cpu_driver->symmetric_channels || + codec_driver->symmetric_channels || link->symmetric_channels || + cpu_driver->symmetric_samplebits || + codec_driver->symmetric_samplebits || + link->symmetric_samplebits; +} + /* * List of sample sizes that might go over the bus for parameter * application. There ought to be a wildcard sample size for things @@ -309,6 +324,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) &cpu_dai_drv->capture); } + if (soc_pcm_has_symmetry(substream)) + runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; + ret = -EINVAL; snd_pcm_limit_hw_rates(runtime); if (!runtime->hw.rates) { -- cgit v1.2.3-70-g09d2 From 3990c516de66af940c5c366a81531787aefe81ae Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 28 Nov 2013 08:50:33 +0100 Subject: ASoC: bcm2835-i2s: Use devm_snd_dmaengine_pcm_register() Makes the code slightly shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/bcm/bcm2835-i2s.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index f49b007c8b4..2685fe4f842 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -848,7 +848,7 @@ static int bcm2835_i2s_probe(struct platform_device *pdev) return ret; } - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); return ret; @@ -862,15 +862,8 @@ static const struct of_device_id bcm2835_i2s_of_match[] = { {}, }; -static int bcm2835_i2s_remove(struct platform_device *pdev) -{ - snd_dmaengine_pcm_unregister(&pdev->dev); - return 0; -} - static struct platform_driver bcm2835_i2s_driver = { .probe = bcm2835_i2s_probe, - .remove = bcm2835_i2s_remove, .driver = { .name = "bcm2835-i2s", .owner = THIS_MODULE, -- cgit v1.2.3-70-g09d2 From 237eeb1c044fdd0f406a8484ee31884e34b9dfc5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 28 Nov 2013 08:50:34 +0100 Subject: ASoC: ep93xx: Use devm_snd_dmaengine_pcm_register() Makes the code slightly shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-pcm.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index cfe517e6800..fdb8b8feb4e 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -78,19 +78,13 @@ static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { static int ep93xx_soc_platform_probe(struct platform_device *pdev) { - return snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(&pdev->dev, &ep93xx_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } -static int ep93xx_soc_platform_remove(struct platform_device *pdev) -{ - snd_dmaengine_pcm_unregister(&pdev->dev); - return 0; -} - static struct platform_driver ep93xx_pcm_driver = { .driver = { .name = "ep93xx-pcm-audio", @@ -98,7 +92,6 @@ static struct platform_driver ep93xx_pcm_driver = { }, .probe = ep93xx_soc_platform_probe, - .remove = ep93xx_soc_platform_remove, }; module_platform_driver(ep93xx_pcm_driver); -- cgit v1.2.3-70-g09d2 From 7e6d18ac7ea1372b462778ff7c416ceaabe71b66 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 28 Nov 2013 08:50:35 +0100 Subject: ASoC: fsl: Use devm_snd_dmaengine_pcm_register() Makes the code shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 8 -------- sound/soc/fsl/fsl_ssi.c | 4 ---- sound/soc/fsl/imx-pcm-dma.c | 9 ++------- sound/soc/fsl/imx-pcm.h | 5 ----- sound/soc/fsl/imx-ssi.c | 3 --- 5 files changed, 2 insertions(+), 27 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 55193a5596c..4d075f1abe7 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1181,13 +1181,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) return ret; } -static int fsl_spdif_remove(struct platform_device *pdev) -{ - imx_pcm_dma_exit(pdev); - - return 0; -} - static const struct of_device_id fsl_spdif_dt_ids[] = { { .compatible = "fsl,imx35-spdif", }, {} @@ -1201,7 +1194,6 @@ static struct platform_driver fsl_spdif_driver = { .of_match_table = fsl_spdif_dt_ids, }, .probe = fsl_spdif_probe, - .remove = fsl_spdif_remove, }; module_platform_driver(fsl_spdif_driver); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fb8f52a0e7b..3df0318e71d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1109,8 +1109,6 @@ done: return 0; error_dai: - if (ssi_private->ssi_on_imx) - imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); error_dev: @@ -1132,8 +1130,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (!ssi_private->new_binding) platform_device_unregister(ssi_private->pdev); - if (ssi_private->ssi_on_imx) - imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); if (ssi_private->ssi_on_imx) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index aee23077080..c5e47f866b4 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -61,16 +61,11 @@ static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { int imx_pcm_dma_init(struct platform_device *pdev) { - return snd_dmaengine_pcm_register(&pdev->dev, &imx_dmaengine_pcm_config, + return devm_snd_dmaengine_pcm_register(&pdev->dev, + &imx_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_COMPAT); } EXPORT_SYMBOL_GPL(imx_pcm_dma_init); -void imx_pcm_dma_exit(struct platform_device *pdev) -{ - snd_dmaengine_pcm_unregister(&pdev->dev); -} -EXPORT_SYMBOL_GPL(imx_pcm_dma_exit); - MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 5d5b73303e1..c79cb27473b 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -40,16 +40,11 @@ struct imx_pcm_fiq_params { #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) int imx_pcm_dma_init(struct platform_device *pdev); -void imx_pcm_dma_exit(struct platform_device *pdev); #else static inline int imx_pcm_dma_init(struct platform_device *pdev) { return -ENODEV; } - -static inline void imx_pcm_dma_exit(struct platform_device *pdev) -{ -} #endif #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f5f248c91c1..cc7376f87f8 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -624,9 +624,6 @@ static int imx_ssi_remove(struct platform_device *pdev) { struct imx_ssi *ssi = platform_get_drvdata(pdev); - if (!ssi->dma_init) - imx_pcm_dma_exit(pdev); - if (!ssi->fiq_init) imx_pcm_fiq_exit(pdev); -- cgit v1.2.3-70-g09d2 From 2650bc4f6d0c36f1219d2070485cc2980a88fab3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 28 Nov 2013 08:50:37 +0100 Subject: ASoC: mxs: Use devm_snd_dmaengine_pcm_register() Makes the code shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-pcm.c | 8 +------- sound/soc/mxs/mxs-pcm.h | 1 - sound/soc/mxs/mxs-saif.c | 8 -------- 3 files changed, 1 insertion(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index b16abbbf776..04a6b0d6094 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -56,16 +56,10 @@ static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { int mxs_pcm_platform_register(struct device *dev) { - return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, + return devm_snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); -void mxs_pcm_platform_unregister(struct device *dev) -{ - snd_dmaengine_pcm_unregister(dev); -} -EXPORT_SYMBOL_GPL(mxs_pcm_platform_unregister); - MODULE_LICENSE("GPL"); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index bc685b67cac..035ea0436ca 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -20,6 +20,5 @@ #define _MXS_PCM_H int mxs_pcm_platform_register(struct device *dev); -void mxs_pcm_platform_unregister(struct device *dev); #endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 54e622acac3..92db74dc317 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -804,13 +804,6 @@ static int mxs_saif_probe(struct platform_device *pdev) return 0; } -static int mxs_saif_remove(struct platform_device *pdev) -{ - mxs_pcm_platform_unregister(&pdev->dev); - - return 0; -} - static const struct of_device_id mxs_saif_dt_ids[] = { { .compatible = "fsl,imx28-saif", }, { /* sentinel */ } @@ -819,7 +812,6 @@ MODULE_DEVICE_TABLE(of, mxs_saif_dt_ids); static struct platform_driver mxs_saif_driver = { .probe = mxs_saif_probe, - .remove = mxs_saif_remove, .driver = { .name = "mxs-saif", -- cgit v1.2.3-70-g09d2 From a8ca52b7911378864e6defb42be9166c248a3749 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Nov 2013 17:27:02 +0000 Subject: ASoC: ak4642: Convert to table based control init Improves error handling and saves code. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2f861c9b1d6..7fe1e9030c4 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -469,9 +469,6 @@ static int ak4642_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_add_codec_controls(codec, ak4642_snd_controls, - ARRAY_SIZE(ak4642_snd_controls)); - ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -491,6 +488,8 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .reg_cache_default = ak4642_reg, /* ak4642 reg */ .reg_cache_size = ARRAY_SIZE(ak4642_reg), /* ak4642 reg */ .reg_word_size = sizeof(u8), + .controls = ak4642_snd_controls, + .num_controls = ARRAY_SIZE(ak4642_snd_controls), .dapm_widgets = ak4642_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), .dapm_routes = ak4642_intercon, @@ -505,6 +504,8 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { .reg_cache_default = ak4648_reg, /* ak4648 reg */ .reg_cache_size = ARRAY_SIZE(ak4648_reg), /* ak4648 reg */ .reg_word_size = sizeof(u8), + .controls = ak4642_snd_controls, + .num_controls = ARRAY_SIZE(ak4642_snd_controls), .dapm_widgets = ak4642_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), .dapm_routes = ak4642_intercon, -- cgit v1.2.3-70-g09d2 From 4574cd94a717eff3021a3e187dd48846adbd21ea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Nov 2013 18:03:49 +0000 Subject: ASoC: ak4642: Convert to direct regmap API usage This moves us towards being able to remove the ASoC level I/O code which duplicates regmap functionality. Currently the only difference between the supported devices in the driver is the regmap so we can replace the CODEC driver selections with regmap selection instead. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 113 ++++++++++++++++++++++++---------------------- 1 file changed, 59 insertions(+), 54 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7fe1e9030c4..5af23746861 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include @@ -198,30 +199,30 @@ static const struct snd_soc_dapm_route ak4642_intercon[] = { /* * ak4642 register cache */ -static const u8 ak4642_reg[] = { - 0x00, 0x00, 0x01, 0x00, - 0x02, 0x00, 0x00, 0x00, - 0xe1, 0xe1, 0x18, 0x00, - 0xe1, 0x18, 0x11, 0x08, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, +static const struct reg_default ak4642_reg[] = { + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, + { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, + { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 }, + { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0x08 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 }, + { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, }; -static const u8 ak4648_reg[] = { - 0x00, 0x00, 0x01, 0x00, - 0x02, 0x00, 0x00, 0x00, - 0xe1, 0xe1, 0x18, 0x00, - 0xe1, 0x18, 0x11, 0xb8, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - 0x00, 0x88, 0x88, 0x08, +static const struct reg_default ak4648_reg[] = { + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, + { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, + { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 }, + { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0xb8 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 }, + { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 }, }; static int ak4642_dai_startup(struct snd_pcm_substream *substream, @@ -454,7 +455,10 @@ static struct snd_soc_dai_driver ak4642_dai = { static int ak4642_resume(struct snd_soc_codec *codec) { - snd_soc_cache_sync(codec); + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_mark_dirty(regmap); + regcache_sync(regmap); return 0; } @@ -463,7 +467,7 @@ static int ak4642_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -485,9 +489,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .remove = ak4642_remove, .resume = ak4642_resume, .set_bias_level = ak4642_set_bias_level, - .reg_cache_default = ak4642_reg, /* ak4642 reg */ - .reg_cache_size = ARRAY_SIZE(ak4642_reg), /* ak4642 reg */ - .reg_word_size = sizeof(u8), .controls = ak4642_snd_controls, .num_controls = ARRAY_SIZE(ak4642_snd_controls), .dapm_widgets = ak4642_dapm_widgets, @@ -496,20 +497,20 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), }; -static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { - .probe = ak4642_probe, - .remove = ak4642_remove, - .resume = ak4642_resume, - .set_bias_level = ak4642_set_bias_level, - .reg_cache_default = ak4648_reg, /* ak4648 reg */ - .reg_cache_size = ARRAY_SIZE(ak4648_reg), /* ak4648 reg */ - .reg_word_size = sizeof(u8), - .controls = ak4642_snd_controls, - .num_controls = ARRAY_SIZE(ak4642_snd_controls), - .dapm_widgets = ak4642_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), - .dapm_routes = ak4642_intercon, - .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), +static const struct regmap_config ak4642_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(ak4642_reg) + 1, + .reg_defaults = ak4642_reg, + .num_reg_defaults = ARRAY_SIZE(ak4642_reg), +}; + +static const struct regmap_config ak4648_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(ak4648_reg) + 1, + .reg_defaults = ak4648_reg, + .num_reg_defaults = ARRAY_SIZE(ak4648_reg), }; #if IS_ENABLED(CONFIG_I2C) @@ -518,26 +519,30 @@ static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct device_node *np = i2c->dev.of_node; - const struct snd_soc_codec_driver *driver; + const struct regmap_config *regmap_config = NULL; + struct regmap *regmap; - driver = NULL; if (np) { const struct of_device_id *of_id; of_id = of_match_device(ak4642_of_match, &i2c->dev); if (of_id) - driver = of_id->data; + regmap_config = of_id->data; } else { - driver = (struct snd_soc_codec_driver *)id->driver_data; + regmap_config = (const struct regmap_config *)id->driver_data; } - if (!driver) { - dev_err(&i2c->dev, "no driver\n"); + if (!regmap_config) { + dev_err(&i2c->dev, "Unknown device type\n"); return -EINVAL; } + regmap = devm_regmap_init_i2c(i2c, regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + return snd_soc_register_codec(&i2c->dev, - driver, &ak4642_dai, 1); + &soc_codec_dev_ak4642, &ak4642_dai, 1); } static int ak4642_i2c_remove(struct i2c_client *client) @@ -547,17 +552,17 @@ static int ak4642_i2c_remove(struct i2c_client *client) } static struct of_device_id ak4642_of_match[] = { - { .compatible = "asahi-kasei,ak4642", .data = &soc_codec_dev_ak4642}, - { .compatible = "asahi-kasei,ak4643", .data = &soc_codec_dev_ak4642}, - { .compatible = "asahi-kasei,ak4648", .data = &soc_codec_dev_ak4648}, + { .compatible = "asahi-kasei,ak4642", .data = &ak4642_regmap}, + { .compatible = "asahi-kasei,ak4643", .data = &ak4642_regmap}, + { .compatible = "asahi-kasei,ak4648", .data = &ak4648_regmap}, {}, }; MODULE_DEVICE_TABLE(of, ak4642_of_match); static const struct i2c_device_id ak4642_i2c_id[] = { - { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 }, - { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 }, - { "ak4648", (kernel_ulong_t)&soc_codec_dev_ak4648 }, + { "ak4642", (kernel_ulong_t)&ak4642_regmap }, + { "ak4643", (kernel_ulong_t)&ak4642_regmap }, + { "ak4648", (kernel_ulong_t)&ak4648_regmap }, { } }; MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); -- cgit v1.2.3-70-g09d2 From 3621dbbc27ff347f2e4476013054bab18ebd906c Mon Sep 17 00:00:00 2001 From: Oskar Schirmer Date: Sat, 16 Nov 2013 07:52:25 +0000 Subject: ASoC: fsl: imx-ssi: omit ssi counter to avoid harm in unbalanced situation Unbalanced calls to imx_ssi_trigger() may result in endless SSI activity and thus provoke eternal sound. While on the first glance, the switch statement looks pretty symmetric, the SUSPEND/RESUME pair is not: the suspend case comes along snd_pcm_suspend_all(), which for fsl/imx-pcm-fiq is called only at snd_soc_suspend(), but the resume case originates straight from the SNDRV_PCM_IOCTL_RESUME. This way userland may provoke an unbalanced resume, which might cause the ssi->enabled counter to increase and never return to zero again, so eventually SSI_SCR_SSIEN is never disabled. As the information on whether to enable the SSI or not is contained in the two bits for TE/RE, we save all the software mirroring of hardware state here and simply use the hardware register itself to keep the state of whether someone is currently playing or capturing. This is essentially the same stuff as in sound/soc/fsl/imx-pcm-fiq.c which I send a patch for three days ago. Astonishing enough this highly fragile scheme is used twice in parallel to serve the very same control function, synchronously: Once out of sync you are lost until reboot. Note, that these fixes wont prevent state machine distortion on alsa level to cut sound or the like. It just makes sure we have a chance to synchronise again later on. Signed-off-by: Oskar Schirmer Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 5 ++--- sound/soc/fsl/imx-ssi.h | 1 - 2 files changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index cc7376f87f8..6336757e967 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -304,8 +304,7 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, scr |= SSI_SCR_RE; sier |= sier_bits; - if (++ssi->enabled == 1) - scr |= SSI_SCR_SSIEN; + scr |= SSI_SCR_SSIEN; break; @@ -318,7 +317,7 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, scr &= ~SSI_SCR_RE; sier &= ~sier_bits; - if (--ssi->enabled == 0) + if (!(scr & (SSI_SCR_TE | SSI_SCR_RE))) scr &= ~SSI_SCR_SSIEN; break; diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index 560c40fc9eb..be6562365b6 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -213,7 +213,6 @@ struct imx_ssi { int fiq_init; int dma_init; - int enabled; }; #endif /* _IMX_SSI_H */ -- cgit v1.2.3-70-g09d2 From fa558c2801fc173758c742d836f3cc4621851557 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Nov 2013 15:25:02 +0900 Subject: ASoC: simple-card: add Device Tree support Support for loading the simple-card module via DeviceTree. It requests CPU/CODEC information. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/simple-card.txt | 68 +++++++++ sound/soc/generic/simple-card.c | 156 ++++++++++++++++++++- 2 files changed, 218 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt new file mode 100644 index 00000000000..769a346f890 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -0,0 +1,68 @@ +Simple-Card: + +Simple-Card specifies audio DAI connection of SoC <-> codec. + +Required properties: + +- compatible : "simple-audio-card" + +Optional properties: + +- simple-audio-card,format : CPU/CODEC common audio format. + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" +Required subnodes: + +- simple-audio-card,cpu : CPU sub-node +- simple-audio-card,codec : CODEC sub-node + +Required CPU/CODEC subnodes properties: + +- sound-dai : phandle and port of CPU/CODEC + +Optional CPU/CODEC subnodes properties: + +- format : CPU/CODEC specific audio format if needed. + see simple-audio-card,format +- frame-master : bool property. add this if subnode is frame master +- bitclock-master : bool property. add this if subnode is bitclock master +- bitclock-inversion : bool property. add this if subnode has clock inversion +- frame-inversion : bool property. add this if subnode has frame inversion +- clocks / system-clock-frequency : specify subnode's clock if needed. + it can be specified via "clocks" if system has + clock node (= common clock), or "system-clock-frequency" + (if system doens't support common clock) + +Example: + +sound { + compatible = "simple-audio-card"; + simple-audio-card,format = "left_j"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + simple-audio-card,codec { + sound-dai = <&ak4648>; + bitclock-master; + frame-master; + clocks = <&osc>; + }; +}; + +&i2c0 { + ak4648: ak4648@12 { + #sound-dai-cells = <0>; + compatible = "asahi-kasei,ak4648"; + reg = <0x12>; + }; +}; + +sh_fsi2: sh_fsi2@ec230000 { + #sound-dai-cells = <1>; + compatible = "renesas,sh_fsi2"; + reg = <0xec230000 0x400>; + interrupt-parent = <&gic>; + interrupts = <0 146 0x4>; +}; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index b2fbb7075a6..7a9b6b4898c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -8,7 +8,8 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ - +#include +#include #include #include #include @@ -57,11 +58,144 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int +asoc_simple_card_sub_parse_of(struct device_node *np, + struct asoc_simple_dai *dai, + struct device_node **node) +{ + struct clk *clk; + int ret; + + /* + * get node via "sound-dai = <&phandle port>" + * it will be used as xxx_of_node on soc_bind_dai_link() + */ + *node = of_parse_phandle(np, "sound-dai", 0); + if (!*node) + return -ENODEV; + + /* get dai->name */ + ret = snd_soc_of_get_dai_name(np, &dai->name); + if (ret < 0) + goto parse_error; + + /* + * bitclock-inversion, frame-inversion + * bitclock-master, frame-master + * and specific "format" if it has + */ + dai->fmt = snd_soc_of_parse_daifmt(np, NULL); + + /* + * dai->sysclk come from + * "clocks = <&xxx>" (if system has common clock) + * or "system-clock-frequency = " + */ + clk = of_clk_get(np, 0); + if (IS_ERR(clk)) + of_property_read_u32(np, + "system-clock-frequency", + &dai->sysclk); + else + dai->sysclk = clk_get_rate(clk); + + ret = 0; + +parse_error: + of_node_put(*node); + + return ret; +} + +static int asoc_simple_card_parse_of(struct device_node *node, + struct asoc_simple_card_info *info, + struct device *dev, + struct device_node **of_cpu, + struct device_node **of_codec, + struct device_node **of_platform) +{ + struct device_node *np; + char *name; + int ret = 0; + + /* get CPU/CODEC common format via simple-audio-card,format */ + info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & + (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); + + /* CPU sub-node */ + ret = -EINVAL; + np = of_get_child_by_name(node, "simple-audio-card,cpu"); + if (np) + ret = asoc_simple_card_sub_parse_of(np, + &info->cpu_dai, + of_cpu); + if (ret < 0) + return ret; + + /* CODEC sub-node */ + ret = -EINVAL; + np = of_get_child_by_name(node, "simple-audio-card,codec"); + if (np) + ret = asoc_simple_card_sub_parse_of(np, + &info->codec_dai, + of_codec); + if (ret < 0) + return ret; + + /* card name is created from CPU/CODEC dai name */ + name = devm_kzalloc(dev, + strlen(info->cpu_dai.name) + + strlen(info->codec_dai.name) + 2, + GFP_KERNEL); + sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); + info->name = info->card = name; + + /* simple-card assumes platform == cpu */ + *of_platform = *of_cpu; + + dev_dbg(dev, "card-name : %s\n", info->card); + dev_dbg(dev, "platform : %04x\n", info->daifmt); + dev_dbg(dev, "cpu : %s / %04x / %d\n", + info->cpu_dai.name, + info->cpu_dai.fmt, + info->cpu_dai.sysclk); + dev_dbg(dev, "codec : %s / %04x / %d\n", + info->codec_dai.name, + info->codec_dai.fmt, + info->codec_dai.sysclk); + + return 0; +} + static int asoc_simple_card_probe(struct platform_device *pdev) { - struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; + struct asoc_simple_card_info *cinfo; + struct device_node *np = pdev->dev.of_node; + struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; + cinfo = NULL; + of_cpu = NULL; + of_codec = NULL; + of_platform = NULL; + if (np && of_device_is_available(np)) { + cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); + if (cinfo) { + int ret; + ret = asoc_simple_card_parse_of(np, cinfo, dev, + &of_cpu, + &of_codec, + &of_platform); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "parse error %d\n", ret); + return ret; + } + } + } else { + cinfo = pdev->dev.platform_data; + } + if (!cinfo) { dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; @@ -69,10 +203,10 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (!cinfo->name || !cinfo->card || - !cinfo->codec || - !cinfo->platform || - !cinfo->cpu_dai.name || - !cinfo->codec_dai.name) { + !cinfo->codec_dai.name || + !(cinfo->codec || of_codec) || + !(cinfo->platform || of_platform) || + !(cinfo->cpu_dai.name || of_cpu)) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } @@ -86,6 +220,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_link.platform_name = cinfo->platform; cinfo->snd_link.codec_name = cinfo->codec; cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; + cinfo->snd_link.cpu_of_node = of_cpu; + cinfo->snd_link.codec_of_node = of_codec; + cinfo->snd_link.platform_of_node = of_platform; cinfo->snd_link.init = asoc_simple_card_dai_init; /* @@ -107,10 +244,17 @@ static int asoc_simple_card_remove(struct platform_device *pdev) return snd_soc_unregister_card(&cinfo->snd_card); } +static const struct of_device_id asoc_simple_of_match[] = { + { .compatible = "simple-audio-card", }, + {}, +}; +MODULE_DEVICE_TABLE(of, asoc_simple_of_match); + static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", .owner = THIS_MODULE, + .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, .remove = asoc_simple_card_remove, -- cgit v1.2.3-70-g09d2 From 2f54d2a1cf7e62f56b1b0bcf44bd704f65359f38 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Nov 2013 17:24:55 +0000 Subject: ASoC: ak4642: Convert to module_i2c_driver() The device does not support anything other than I2C (at least with the current driver) so save code. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 22 +--------------------- 1 file changed, 1 insertion(+), 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 5af23746861..1f646c6e90c 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -513,7 +513,6 @@ static const struct regmap_config ak4648_regmap = { .num_reg_defaults = ARRAY_SIZE(ak4648_reg), }; -#if IS_ENABLED(CONFIG_I2C) static struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -577,27 +576,8 @@ static struct i2c_driver ak4642_i2c_driver = { .remove = ak4642_i2c_remove, .id_table = ak4642_i2c_id, }; -#endif -static int __init ak4642_modinit(void) -{ - int ret = 0; -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&ak4642_i2c_driver); -#endif - return ret; - -} -module_init(ak4642_modinit); - -static void __exit ak4642_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&ak4642_i2c_driver); -#endif - -} -module_exit(ak4642_exit); +module_i2c_driver(ak4642_i2c_driver); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto "); -- cgit v1.2.3-70-g09d2 From 2924a9981006ad01efb46c754689fa7d03e3eb4f Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Mon, 2 Dec 2013 23:29:03 +0800 Subject: ASoC: fsl_ssi: Add monaural audio support for non-ac97 interface The normal mode of SSI allows it to send/receive data to/from the first slot of each period. So we can use this normal mode to trick I2S signal by puting/getting data to/from the first slot only (the left channel) so as to support monaural audio playback and recording. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 3df0318e71d..90ff1071e29 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -143,6 +143,7 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + u8 i2s_mode; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -354,14 +355,13 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u8 i2s_mode; u8 wm; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; if (ssi_private->imx_ac97) - i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; else - i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; /* * Section 16.5 of the MPC8610 reference manual says that the SSI needs @@ -378,7 +378,7 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | - i2s_mode | + ssi_private->i2s_mode | (synchronous ? CCSR_SSI_SCR_SYN : 0)); write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -508,6 +508,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int channels = params_channels(hw_params); unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); @@ -537,6 +538,11 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, else write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if (!ssi_private->imx_ac97) + write_ssi_mask(&ssi->scr, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : ssi_private->i2s_mode); + return 0; } @@ -649,14 +655,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { static struct snd_soc_dai_driver fsl_ssi_dai_template = { .probe = fsl_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, -- cgit v1.2.3-70-g09d2 From 07a9483aaca5d3b5de8ee824ee576321d3f8b4c6 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 3 Dec 2013 18:38:07 +0800 Subject: ASoC: fsl_ssi: Implement symmetric_channels and symmetric_samplebits Since we introduced symmetric_channels and symmetric_samplebits, we implement these two features to fsl_ssi so as to drop some no-more-needed code and make the driver neat and clean. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 79 ++++++------------------------------------------- 1 file changed, 9 insertions(+), 70 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 90ff1071e29..f9f4569417e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -119,8 +119,6 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI - * @first_stream: pointer to the stream that was opened first - * @second_stream: pointer to second stream * @playback: the number of playback streams opened * @capture: the number of capture streams opened * @cpu_dai: the CPU DAI for this device @@ -132,8 +130,6 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; - struct snd_pcm_substream *first_stream; - struct snd_pcm_substream *second_stream; unsigned int fifo_depth; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; @@ -438,54 +434,13 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; - /* - * If this is the first stream opened, then request the IRQ - * and initialize the SSI registers. + /* First, we only do fsl_ssi_setup() when SSI is going to be active. + * Second, fsl_ssi_setup was already called by ac97_init earlier if + * the driver is in ac97 mode. */ - if (!ssi_private->first_stream) { - ssi_private->first_stream = substream; - - /* - * fsl_ssi_setup was already called by ac97_init earlier if - * the driver is in ac97 mode. - */ - if (!ssi_private->imx_ac97) - fsl_ssi_setup(ssi_private); - } else { - if (synchronous) { - struct snd_pcm_runtime *first_runtime = - ssi_private->first_stream->runtime; - /* - * This is the second stream open, and we're in - * synchronous mode, so we need to impose sample - * sample size constraints. This is because STCCR is - * used for playback and capture in synchronous mode, - * so there's no way to specify different word - * lengths. - * - * Note that this can cause a race condition if the - * second stream is opened before the first stream is - * fully initialized. We provide some protection by - * checking to make sure the first stream is - * initialized, but it's not perfect. ALSA sometimes - * re-initializes the driver with a different sample - * rate or size. If the second stream is opened - * before the first stream has received its final - * parameters, then the second stream may be - * constrained to the wrong sample rate or size. - */ - if (first_runtime->sample_bits) { - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - first_runtime->sample_bits, - first_runtime->sample_bits); - } - } - - ssi_private->second_stream = substream; - } + if (!dai->active && !ssi_private->imx_ac97) + fsl_ssi_setup(ssi_private); return 0; } @@ -615,23 +570,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -/** - * fsl_ssi_shutdown: shutdown the SSI - * - * Shutdown the SSI if there are no other substreams open. - */ -static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - - if (ssi_private->first_stream == substream) - ssi_private->first_stream = ssi_private->second_stream; - - ssi_private->second_stream = NULL; -} - static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai); @@ -647,7 +585,6 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, - .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, }; @@ -722,7 +659,6 @@ static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd, static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = { .startup = fsl_ssi_startup, - .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_ac97_trigger, }; @@ -947,8 +883,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) } /* Are the RX and the TX clocks locked? */ - if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) + if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; + ssi_private->cpu_dai_drv.symmetric_samplebits = 1; + } /* Determine the FIFO depth. */ iprop = of_get_property(np, "fsl,fifo-depth", NULL); -- cgit v1.2.3-70-g09d2 From a010ff628c0953e6c914ecd09678363848617a88 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 2 Dec 2013 21:26:22 +0100 Subject: ASoC: ssm2602: Use core for applying symmetry constraints Let the core take care of applying sample rate and sample bits constraints instead of open-coding this in the driver. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 50 ++-------------------------------------------- 1 file changed, 2 insertions(+), 48 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 480074d8647..c6dd4856188 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -53,8 +53,6 @@ enum ssm2602_type { struct ssm2602_priv { unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; struct regmap *regmap; @@ -277,11 +275,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); unsigned int iface; - if (substream == ssm2602->slave_substream) { - dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n"); - return 0; - } - if (srate < 0) return srate; @@ -314,33 +307,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct snd_pcm_runtime *master_runtime; - - /* The DAI has shared clocks so if we already have a playback or - * capture going then constrain this substream to match it. - * TODO: the ssm2602 allows pairs of non-matching PB/REC rates - */ - if (ssm2602->master_substream) { - master_runtime = ssm2602->master_substream->runtime; - dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n", - master_runtime->sample_bits, - master_runtime->rate); - - if (master_runtime->rate != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - if (master_runtime->sample_bits != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - - ssm2602->slave_substream = substream; - } else - ssm2602->master_substream = substream; if (ssm2602->sysclk_constraints) { snd_pcm_hw_constraint_list(substream->runtime, 0, @@ -351,19 +317,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, return 0; } -static void ssm2602_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - - if (ssm2602->master_substream == substream) - ssm2602->master_substream = ssm2602->slave_substream; - - ssm2602->slave_substream = NULL; -} - - static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec); @@ -530,7 +483,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops ssm2602_dai_ops = { .startup = ssm2602_startup, .hw_params = ssm2602_hw_params, - .shutdown = ssm2602_shutdown, .digital_mute = ssm2602_mute, .set_sysclk = ssm2602_set_dai_sysclk, .set_fmt = ssm2602_set_dai_fmt, @@ -551,6 +503,8 @@ static struct snd_soc_dai_driver ssm2602_dai = { .rates = SSM2602_RATES, .formats = SSM2602_FORMATS,}, .ops = &ssm2602_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, }; static int ssm2602_suspend(struct snd_soc_codec *codec) -- cgit v1.2.3-70-g09d2 From b84c9ce809c91b3c613c967abcee90ebd6582092 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 3 Dec 2013 18:53:02 +0100 Subject: ASoC: jz4740-i2s: Use managed resources Makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 85 ++++++++----------------------------------- 1 file changed, 15 insertions(+), 70 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 4c849a49c72..a0b6a85d89c 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -432,91 +432,36 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { static int jz4740_i2s_dev_probe(struct platform_device *pdev) { struct jz4740_i2s *i2s; + struct resource *mem; int ret; - i2s = kzalloc(sizeof(*i2s), GFP_KERNEL); - + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) return -ENOMEM; - i2s->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!i2s->mem) { - ret = -ENOENT; - goto err_free; - } - - i2s->mem = request_mem_region(i2s->mem->start, resource_size(i2s->mem), - pdev->name); - if (!i2s->mem) { - ret = -EBUSY; - goto err_free; - } - - i2s->base = ioremap_nocache(i2s->mem->start, resource_size(i2s->mem)); - if (!i2s->base) { - ret = -EBUSY; - goto err_release_mem_region; - } + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->base)) + return PTR_ERR(i2s->base); - i2s->phys_base = i2s->mem->start; + i2s->phys_base = mem->start; - i2s->clk_aic = clk_get(&pdev->dev, "aic"); - if (IS_ERR(i2s->clk_aic)) { - ret = PTR_ERR(i2s->clk_aic); - goto err_iounmap; - } + i2s->clk_aic = devm_clk_get(&pdev->dev, "aic"); + if (IS_ERR(i2s->clk_aic)) + return PTR_ERR(i2s->clk_aic); - i2s->clk_i2s = clk_get(&pdev->dev, "i2s"); - if (IS_ERR(i2s->clk_i2s)) { - ret = PTR_ERR(i2s->clk_i2s); - goto err_clk_put_aic; - } + i2s->clk_i2s = devm_clk_get(&pdev->dev, "i2s"); + if (IS_ERR(i2s->clk_i2s)) + return PTR_ERR(i2s->clk_i2s); platform_set_drvdata(pdev, i2s); - ret = snd_soc_register_component(&pdev->dev, &jz4740_i2s_component, - &jz4740_i2s_dai, 1); - - if (ret) { - dev_err(&pdev->dev, "Failed to register DAI\n"); - goto err_clk_put_i2s; - } - - return 0; - -err_clk_put_i2s: - clk_put(i2s->clk_i2s); -err_clk_put_aic: - clk_put(i2s->clk_aic); -err_iounmap: - iounmap(i2s->base); -err_release_mem_region: - release_mem_region(i2s->mem->start, resource_size(i2s->mem)); -err_free: - kfree(i2s); - - return ret; -} - -static int jz4740_i2s_dev_remove(struct platform_device *pdev) -{ - struct jz4740_i2s *i2s = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - - clk_put(i2s->clk_i2s); - clk_put(i2s->clk_aic); - iounmap(i2s->base); - release_mem_region(i2s->mem->start, resource_size(i2s->mem)); - - kfree(i2s); - - return 0; + return devm_snd_soc_register_component(&pdev->dev, + &jz4740_i2s_component, &jz4740_i2s_dai, 1); } static struct platform_driver jz4740_i2s_driver = { .probe = jz4740_i2s_dev_probe, - .remove = jz4740_i2s_dev_remove, .driver = { .name = "jz4740-i2s", .owner = THIS_MODULE, -- cgit v1.2.3-70-g09d2 From 0406a40a095ca039e5f5ec63783342253c573d06 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 3 Dec 2013 18:53:03 +0100 Subject: ASoC: jz4740: Use the generic dmaengine PCM driver Now that there is a dmaengine driver for the jz4740 DMA core we can use the generic dmaengine PCM driver. This allows us to remove the custom jz4740-pcm code completely. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/jz4740/Kconfig | 1 + sound/soc/jz4740/jz4740-i2s.c | 54 +++---- sound/soc/jz4740/jz4740-pcm.c | 358 ------------------------------------------ sound/soc/jz4740/jz4740-pcm.h | 20 --- sound/soc/jz4740/qi_lb60.c | 2 +- 5 files changed, 24 insertions(+), 411 deletions(-) delete mode 100644 sound/soc/jz4740/jz4740-pcm.c delete mode 100644 sound/soc/jz4740/jz4740-pcm.h (limited to 'sound/soc') diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig index 5351cba66c9..29f76af5d96 100644 --- a/sound/soc/jz4740/Kconfig +++ b/sound/soc/jz4740/Kconfig @@ -1,6 +1,7 @@ config SND_JZ4740_SOC tristate "SoC Audio for Ingenic JZ4740 SoC" depends on MACH_JZ4740 && SND_SOC + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the JZ4740 I2S interface. You will also need to select the audio diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index a0b6a85d89c..8f220009e0f 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -29,9 +29,11 @@ #include #include #include +#include + +#include #include "jz4740-i2s.h" -#include "jz4740-pcm.h" #define JZ_REG_AIC_CONF 0x00 #define JZ_REG_AIC_CTRL 0x04 @@ -89,8 +91,8 @@ struct jz4740_i2s { struct clk *clk_aic; struct clk *clk_i2s; - struct jz4740_pcm_config pcm_config_playback; - struct jz4740_pcm_config pcm_config_capture; + struct snd_dmaengine_dai_dma_data playback_dma_data; + struct snd_dmaengine_dai_dma_data capture_dma_data; }; static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s, @@ -233,8 +235,6 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); - enum jz4740_dma_width dma_width; - struct jz4740_pcm_config *pcm_config; unsigned int sample_size; uint32_t ctrl; @@ -243,11 +243,9 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: sample_size = 0; - dma_width = JZ4740_DMA_WIDTH_8BIT; break; case SNDRV_PCM_FORMAT_S16: sample_size = 1; - dma_width = JZ4740_DMA_WIDTH_16BIT; break; default: return -EINVAL; @@ -260,22 +258,13 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl |= JZ_AIC_CTRL_MONO_TO_STEREO; else ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO; - - pcm_config = &i2s->pcm_config_playback; - pcm_config->dma_config.dst_width = dma_width; - } else { ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK; ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET; - - pcm_config = &i2s->pcm_config_capture; - pcm_config->dma_config.src_width = dma_width; } jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); - snd_soc_dai_set_dma_data(dai, substream, pcm_config); - return 0; } @@ -342,25 +331,19 @@ static int jz4740_i2s_resume(struct snd_soc_dai *dai) static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) { - struct jz4740_dma_config *dma_config; + struct snd_dmaengine_dai_dma_data *dma_data; /* Playback */ - dma_config = &i2s->pcm_config_playback.dma_config; - dma_config->src_width = JZ4740_DMA_WIDTH_32BIT; - dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; - dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT; - dma_config->flags = JZ4740_DMA_SRC_AUTOINC; - dma_config->mode = JZ4740_DMA_MODE_SINGLE; - i2s->pcm_config_playback.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO; + dma_data = &i2s->playback_dma_data; + dma_data->maxburst = 16; + dma_data->slave_id = JZ4740_DMA_TYPE_AIC_TRANSMIT; + dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; /* Capture */ - dma_config = &i2s->pcm_config_capture.dma_config; - dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT; - dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; - dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE; - dma_config->flags = JZ4740_DMA_DST_AUTOINC; - dma_config->mode = JZ4740_DMA_MODE_SINGLE; - i2s->pcm_config_capture.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO; + dma_data = &i2s->capture_dma_data; + dma_data->maxburst = 16; + dma_data->slave_id = JZ4740_DMA_TYPE_AIC_RECEIVE; + dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; } static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) @@ -371,6 +354,8 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) clk_prepare_enable(i2s->clk_aic); jz4740_i2c_init_pcm_config(i2s); + snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, + &i2s->capture_dma_data); conf = (7 << JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | (8 << JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | @@ -456,8 +441,13 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) platform_set_drvdata(pdev, i2s); - return devm_snd_soc_register_component(&pdev->dev, + ret = devm_snd_soc_register_component(&pdev->dev, &jz4740_i2s_component, &jz4740_i2s_dai, 1); + if (ret) + return ret; + + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_COMPAT); } static struct platform_driver jz4740_i2s_driver = { diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c deleted file mode 100644 index 1d7ef28585e..00000000000 --- a/sound/soc/jz4740/jz4740-pcm.c +++ /dev/null @@ -1,358 +0,0 @@ -/* - * Copyright (C) 2010, Lars-Peter Clausen - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ - -#include -#include -#include -#include -#include -#include - -#include - -#include -#include -#include -#include - -#include -#include "jz4740-pcm.h" - -struct jz4740_runtime_data { - unsigned long dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - - struct jz4740_dma_chan *dma; - - dma_addr_t fifo_addr; -}; - -/* identify hardware playback capabilities */ -static const struct snd_pcm_hardware jz4740_pcm_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, - - .rates = SNDRV_PCM_RATE_8000_48000, - .channels_min = 1, - .channels_max = 2, - .period_bytes_min = 16, - .period_bytes_max = 2 * PAGE_SIZE, - .periods_min = 2, - .periods_max = 128, - .buffer_bytes_max = 128 * 2 * PAGE_SIZE, - .fifo_size = 32, -}; - -static void jz4740_pcm_start_transfer(struct jz4740_runtime_data *prtd, - struct snd_pcm_substream *substream) -{ - unsigned long count; - - if (prtd->dma_pos == prtd->dma_end) - prtd->dma_pos = prtd->dma_start; - - if (prtd->dma_pos + prtd->dma_period > prtd->dma_end) - count = prtd->dma_end - prtd->dma_pos; - else - count = prtd->dma_period; - - jz4740_dma_disable(prtd->dma); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - jz4740_dma_set_src_addr(prtd->dma, prtd->dma_pos); - jz4740_dma_set_dst_addr(prtd->dma, prtd->fifo_addr); - } else { - jz4740_dma_set_src_addr(prtd->dma, prtd->fifo_addr); - jz4740_dma_set_dst_addr(prtd->dma, prtd->dma_pos); - } - - jz4740_dma_set_transfer_count(prtd->dma, count); - - prtd->dma_pos += count; - - jz4740_dma_enable(prtd->dma); -} - -static void jz4740_pcm_dma_transfer_done(struct jz4740_dma_chan *dma, int err, - void *dev_id) -{ - struct snd_pcm_substream *substream = dev_id; - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd = runtime->private_data; - - snd_pcm_period_elapsed(substream); - - jz4740_pcm_start_transfer(prtd, substream); -} - -static int jz4740_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct jz4740_pcm_config *config; - - config = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (!config) - return 0; - - if (!prtd->dma) { - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - prtd->dma = jz4740_dma_request(substream, "PCM Capture"); - else - prtd->dma = jz4740_dma_request(substream, "PCM Playback"); - } - - if (!prtd->dma) - return -EBUSY; - - jz4740_dma_configure(prtd->dma, &config->dma_config); - prtd->fifo_addr = config->fifo_addr; - - jz4740_dma_set_complete_cb(prtd->dma, jz4740_pcm_dma_transfer_done); - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + runtime->dma_bytes; - - return 0; -} - -static int jz4740_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct jz4740_runtime_data *prtd = substream->runtime->private_data; - - snd_pcm_set_runtime_buffer(substream, NULL); - if (prtd->dma) { - jz4740_dma_free(prtd->dma); - prtd->dma = NULL; - } - - return 0; -} - -static int jz4740_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct jz4740_runtime_data *prtd = substream->runtime->private_data; - - if (!prtd->dma) - return -EBUSY; - - prtd->dma_pos = prtd->dma_start; - - return 0; -} - -static int jz4740_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd = runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - jz4740_pcm_start_transfer(prtd, substream); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - jz4740_dma_disable(prtd->dma); - break; - default: - break; - } - - return 0; -} - -static snd_pcm_uframes_t jz4740_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd = runtime->private_data; - unsigned long byte_offset; - snd_pcm_uframes_t offset; - struct jz4740_dma_chan *dma = prtd->dma; - - /* prtd->dma_pos points to the end of the current transfer. So by - * subtracting prdt->dma_start we get the offset to the end of the - * current period in bytes. By subtracting the residue of the transfer - * we get the current offset in bytes. */ - byte_offset = prtd->dma_pos - prtd->dma_start; - byte_offset -= jz4740_dma_get_residue(dma); - - offset = bytes_to_frames(runtime, byte_offset); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -static int jz4740_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd; - - prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - snd_soc_set_runtime_hwparams(substream, &jz4740_pcm_hardware); - - runtime->private_data = prtd; - - return 0; -} - -static int jz4740_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct jz4740_runtime_data *prtd = runtime->private_data; - - kfree(prtd); - - return 0; -} - -static int jz4740_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} - -static struct snd_pcm_ops jz4740_pcm_ops = { - .open = jz4740_pcm_open, - .close = jz4740_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = jz4740_pcm_hw_params, - .hw_free = jz4740_pcm_hw_free, - .prepare = jz4740_pcm_prepare, - .trigger = jz4740_pcm_trigger, - .pointer = jz4740_pcm_pointer, - .mmap = jz4740_pcm_mmap, -}; - -static int jz4740_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = jz4740_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - - buf->area = dma_alloc_noncoherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - - return 0; -} - -static void jz4740_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < SNDRV_PCM_STREAM_LAST; ++stream) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_noncoherent(pcm->card->dev, buf->bytes, buf->area, - buf->addr); - buf->area = NULL; - } -} - -static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = jz4740_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto err; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = jz4740_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto err; - } - -err: - return ret; -} - -static struct snd_soc_platform_driver jz4740_soc_platform = { - .ops = &jz4740_pcm_ops, - .pcm_new = jz4740_pcm_new, - .pcm_free = jz4740_pcm_free, -}; - -static int jz4740_pcm_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &jz4740_soc_platform); -} - -static int jz4740_pcm_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver jz4740_pcm_driver = { - .probe = jz4740_pcm_probe, - .remove = jz4740_pcm_remove, - .driver = { - .name = "jz4740-pcm-audio", - .owner = THIS_MODULE, - }, -}; - -module_platform_driver(jz4740_pcm_driver); - -MODULE_AUTHOR("Lars-Peter Clausen "); -MODULE_DESCRIPTION("Ingenic SoC JZ4740 PCM driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/jz4740/jz4740-pcm.h b/sound/soc/jz4740/jz4740-pcm.h deleted file mode 100644 index 1220cbb4382..00000000000 --- a/sound/soc/jz4740/jz4740-pcm.h +++ /dev/null @@ -1,20 +0,0 @@ -/* - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _JZ4740_PCM_H -#define _JZ4740_PCM_H - -#include -#include - - -struct jz4740_pcm_config { - struct jz4740_dma_config dma_config; - phys_addr_t fifo_addr; -}; - -#endif diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 55fd6b5df55..82b5f37cd2c 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -73,7 +73,7 @@ static struct snd_soc_dai_link qi_lb60_dai = { .name = "jz4740", .stream_name = "jz4740", .cpu_dai_name = "jz4740-i2s", - .platform_name = "jz4740-pcm-audio", + .platform_name = "jz4740-i2s", .codec_dai_name = "jz4740-hifi", .codec_name = "jz4740-codec", .init = qi_lb60_codec_init, -- cgit v1.2.3-70-g09d2 From 0b4bbae85e046042af76a65920db4bb5509c97bd Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 4 Dec 2013 11:18:37 +0800 Subject: ASoC: soc-pcm: Drop the redundant snd_soc_dai_digital_mute() in soc_pcm_close() This patch removed the redundant snd_soc_dai_digital_mute() in close() since it's better to mute in hw_free() which's slightly earlier and symmetrical for the case of reconfiguration: 'aplay 44k1.wav 48k.wav', for example, will be open()->hw_params()->prepare(unmute)->playi1ng->hw_free(mute)->hw_params()-> parepare(unmute)->playing->hw_free(mute)->close() Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 42782c01e41..89d59413877 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -390,11 +390,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (!codec_dai->active) codec_dai->rate = 0; - /* Muting the DAC suppresses artifacts caused during digital - * shutdown, for example from stopping clocks. - */ - snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); - if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); -- cgit v1.2.3-70-g09d2 From 60dbb4f17417777fc56cc909c3e89f4d338e8ca8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 3 Dec 2013 22:09:33 -0800 Subject: ASoC: rcar: use devm_clk_get() instead of clk_get() Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 477465f9507..82b04c6f352 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -652,7 +652,7 @@ int rsnd_ssi_probe(struct platform_device *pdev, snprintf(name, RSND_SSI_NAME_SIZE, "ssi.%d", i); - clk = clk_get(dev, name); + clk = devm_clk_get(dev, name); if (IS_ERR(clk)) return PTR_ERR(clk); @@ -713,7 +713,6 @@ void rsnd_ssi_remove(struct platform_device *pdev, int i; for_each_rsnd_ssi(ssi, priv, i) { - clk_put(ssi->clk); if (rsnd_ssi_dma_available(ssi)) rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod)); } -- cgit v1.2.3-70-g09d2 From 1552c32547ca807f0df7d9abca54468033df8764 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Nov 2013 18:11:38 +0000 Subject: ASoC: adsp: Use async writes where possible This will allow a marginal speed improvement when used with a bus that supports async I/O by reducing the amount of context thrashing between writes, allowing the bus to be more fully utilised. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 46ec0e9744d..6b1c01c66e6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1468,8 +1468,8 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) unsigned int val; int ret, count; - ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA, ADSP2_SYS_ENA); + ret = regmap_update_bits_async(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA, ADSP2_SYS_ENA); if (ret != 0) return ret; @@ -1521,9 +1521,9 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, val = (val & ARIZONA_SYSCLK_FREQ_MASK) >> ARIZONA_SYSCLK_FREQ_SHIFT; - ret = regmap_update_bits(dsp->regmap, - dsp->base + ADSP2_CLOCKING, - ADSP2_CLK_SEL_MASK, val); + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CLOCKING, + ADSP2_CLK_SEL_MASK, val); if (ret != 0) { adsp_err(dsp, "Failed to set clock rate: %d\n", ret); @@ -1586,10 +1586,10 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = regmap_update_bits(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA | ADSP2_START, - ADSP2_CORE_ENA | ADSP2_START); + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_CORE_ENA | ADSP2_START, + ADSP2_CORE_ENA | ADSP2_START); if (ret != 0) goto err; -- cgit v1.2.3-70-g09d2 From 24f4bd57a7b24d10e52a3807f88adec79824e5d8 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 4 Dec 2013 20:27:44 -0200 Subject: ASoC: imx-ssi: Check the return value from clk_prepare_enable() clk_prepare_enable() may fail, so let's check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 6336757e967..df552fa1aa6 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -535,7 +535,9 @@ static int imx_ssi_probe(struct platform_device *pdev) ret); goto failed_clk; } - clk_prepare_enable(ssi->clk); + ret = clk_prepare_enable(ssi->clk); + if (ret) + goto failed_clk; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); ssi->base = devm_ioremap_resource(&pdev->dev, res); -- cgit v1.2.3-70-g09d2 From 7637af2e17f18bfe6264d834c6edee7706a0f15c Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 4 Dec 2013 15:19:27 -0700 Subject: ASoC: tegra: add tegra+MAX98090 machine driver Initially, this binding and driver only describe/support playback to headphones and speakers, and capture from the external microphone, with GPIO-based jack detection for the headphone jack only. This driver is useful for the Venice2 board. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- .../bindings/sound/nvidia,tegra-audio-max98090.txt | 51 ++++ sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_max98090.c | 275 +++++++++++++++++++++ 4 files changed, 338 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt create mode 100644 sound/soc/tegra/tegra_max98090.c (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt new file mode 100644 index 00000000000..9c7c55c7137 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt @@ -0,0 +1,51 @@ +NVIDIA Tegra audio complex, with MAX98090 CODEC + +Required properties: +- compatible : "nvidia,tegra-audio-max98090" +- clocks : Must contain an entry for each entry in clock-names. + See ../clocks/clock-bindings.txt for details. +- clock-names : Must include the following entries: + - pll_a + - pll_a_out0 + - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) +- nvidia,model : The user-visible name of this sound complex. +- nvidia,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the MAX98090's pins (as documented in its binding), and the jacks + on the board: + + * Headphones + * Speakers + * Mic Jack + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's + connected to the CODEC. +- nvidia,audio-codec : The phandle of the MAX98090 audio codec. + +Optional properties: +- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in + +Example: + +sound { + compatible = "nvidia,tegra-audio-max98090-venice2", + "nvidia,tegra-audio-max98090"; + nvidia,model = "NVIDIA Tegra Venice2"; + + nvidia,audio-routing = + "Headphones", "HPR", + "Headphones", "HPL", + "Speakers", "SPKR", + "Speakers", "SPKL", + "Mic Jack", "MICBIAS", + "IN34", "Mic Jack"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&acodec>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; +}; diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 8fc653ca3ab..65a85f54252 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -116,3 +116,13 @@ config SND_SOC_TEGRA_ALC5632 help Say Y or M here if you want to add support for SoC audio on the Toshiba AC100 netbook. + +config SND_SOC_TEGRA_MAX98090 + tristate "SoC Audio support for Tegra boards using a MAX98090 codec" + depends on SND_SOC_TEGRA && I2C && GPIOLIB + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_MAX98090 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the MAX98090 codec, such as Venice2. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 21d2550a08a..5ae588cd96c 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -24,6 +24,7 @@ snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o snd-soc-tegra-trimslice-objs := trimslice.o snd-soc-tegra-alc5632-objs := tegra_alc5632.o +snd-soc-tegra-max98090-objs := tegra_max98090.o obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o @@ -31,3 +32,4 @@ obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o +obj-$(CONFIG_SND_SOC_TEGRA_MAX98090) += snd-soc-tegra-max98090.o diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c new file mode 100644 index 00000000000..0283cfb7c03 --- /dev/null +++ b/sound/soc/tegra/tegra_max98090.c @@ -0,0 +1,275 @@ +/* + * Tegra machine ASoC driver for boards using a MAX90809 CODEC. + * + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-max98090" + +struct tegra_max98090 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; +}; + +static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_card *card = codec->card; + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + switch (srate) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + mclk = 12000000; + break; + } + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_max98090_ops = { + .hw_params = tegra_max98090_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_max98090_hp_jack; + +static struct snd_soc_jack_pin tegra_max98090_hp_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_max98090_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, + .invert = 1, +}; + +static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_kcontrol_new tegra_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Speakers"), +}; + +static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(codec->card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, + &tegra_max98090_hp_jack); + snd_soc_jack_add_pins(&tegra_max98090_hp_jack, + ARRAY_SIZE(tegra_max98090_hp_jack_pins), + tegra_max98090_hp_jack_pins); + + tegra_max98090_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_max98090_hp_jack, + 1, + &tegra_max98090_hp_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_max98090_dai = { + .name = "max98090", + .stream_name = "max98090 PCM", + .codec_dai_name = "HiFi", + .init = tegra_max98090_asoc_init, + .ops = &tegra_max98090_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_max98090 = { + .name = "tegra-max98090", + .owner = THIS_MODULE, + .dai_link = &tegra_max98090_dai, + .num_links = 1, + .controls = tegra_max98090_controls, + .num_controls = ARRAY_SIZE(tegra_max98090_controls), + .dapm_widgets = tegra_max98090_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_max98090_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_max98090_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_max98090; + struct tegra_max98090 *machine; + int ret; + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_max98090), GFP_KERNEL); + if (!machine) { + dev_err(&pdev->dev, "Can't allocate tegra_max98090\n"); + return -ENOMEM; + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_max98090_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!tegra_max98090_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_max98090_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_max98090_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_max98090_dai.platform_of_node = tegra_max98090_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int tegra_max98090_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_jack_free_gpios(&tegra_max98090_hp_jack, 1, + &tegra_max98090_hp_jack_gpio); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_max98090_of_match[] = { + { .compatible = "nvidia,tegra-audio-max98090", }, + {}, +}; + +static struct platform_driver tegra_max98090_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_max98090_of_match, + }, + .probe = tegra_max98090_probe, + .remove = tegra_max98090_remove, +}; +module_platform_driver(tegra_max98090_driver); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra max98090 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_max98090_of_match); -- cgit v1.2.3-70-g09d2 From 8f2fe346822419ee729c081b71c8835b733e0884 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 6 Dec 2013 11:02:50 +0100 Subject: ASoC: Add support for the Analog Devices AXI-I2S core This patch adds support for the AXI-I2S softcore. The core implements a simple bidirectional I2S transceiver and is used by Analog Devices in some of their reference designs for various FPGA platforms. The driver uses the generic PCM dmaengine driver for its PCM. The only restriction is that we need to set the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag as the dmaengine driver for the DMA core (PL330) that is used with this core has no residue reporting capabilities yet. This will be fixed in the future though. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/adi/Kconfig | 13 +++ sound/soc/adi/Makefile | 3 + sound/soc/adi/axi-i2s.c | 277 ++++++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 295 insertions(+) create mode 100644 sound/soc/adi/Kconfig create mode 100644 sound/soc/adi/Makefile create mode 100644 sound/soc/adi/axi-i2s.c (limited to 'sound/soc') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5138b849305..866dfec4b6b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM select SND_DMAENGINE_PCM # All the supported SoCs +source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/blackfin/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8b9e70105dd..c70c7f76d2d 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -8,6 +8,7 @@ endif obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ +obj-$(CONFIG_SND_SOC) += adi/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += blackfin/ diff --git a/sound/soc/adi/Kconfig b/sound/soc/adi/Kconfig new file mode 100644 index 00000000000..46f4b79f7a8 --- /dev/null +++ b/sound/soc/adi/Kconfig @@ -0,0 +1,13 @@ +config SND_SOC_ADI + tristate "Audio support for Analog Devices reference designs" + depends on MICROBLAZE || ARCH_ZYNQ || COMPILE_TEST + help + Audio support for various reference designs by Analog Devices. + +config SND_SOC_ADI_AXI_I2S + tristate "AXI-I2S support" + depends on SND_SOC_ADI + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + ASoC driver for the Analog Devices AXI-I2S softcore peripheral. diff --git a/sound/soc/adi/Makefile b/sound/soc/adi/Makefile new file mode 100644 index 00000000000..d32c21ae14b --- /dev/null +++ b/sound/soc/adi/Makefile @@ -0,0 +1,3 @@ +snd-soc-adi-axi-i2s-objs := axi-i2s.o + +obj-$(CONFIG_SND_SOC_ADI_AXI_I2S) += snd-soc-adi-axi-i2s.o diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c new file mode 100644 index 00000000000..0822c771366 --- /dev/null +++ b/sound/soc/adi/axi-i2s.c @@ -0,0 +1,277 @@ +/* + * Copyright (C) 2012-2013, Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#define AXI_I2S_REG_RESET 0x00 +#define AXI_I2S_REG_CTRL 0x04 +#define AXI_I2S_REG_CLK_CTRL 0x08 +#define AXI_I2S_REG_STATUS 0x10 + +#define AXI_I2S_REG_RX_FIFO 0x28 +#define AXI_I2S_REG_TX_FIFO 0x2C + +#define AXI_I2S_RESET_GLOBAL BIT(0) +#define AXI_I2S_RESET_TX_FIFO BIT(1) +#define AXI_I2S_RESET_RX_FIFO BIT(2) + +#define AXI_I2S_CTRL_TX_EN BIT(0) +#define AXI_I2S_CTRL_RX_EN BIT(1) + +/* The frame size is configurable, but for now we always set it 64 bit */ +#define AXI_I2S_BITS_PER_FRAME 64 + +struct axi_i2s { + struct regmap *regmap; + struct clk *clk; + struct clk *clk_ref; + + struct snd_soc_dai_driver dai_driver; + + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct snd_dmaengine_dai_dma_data playback_dma_data; + + struct snd_ratnum ratnum; + struct snd_pcm_hw_constraint_ratnums rate_constraints; +}; + +static int axi_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = AXI_I2S_CTRL_RX_EN; + else + mask = AXI_I2S_CTRL_TX_EN; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = mask; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, AXI_I2S_REG_CTRL, mask, val); + + return 0; +} + +static int axi_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int bclk_div, word_size; + unsigned int bclk_rate; + + bclk_rate = params_rate(params) * AXI_I2S_BITS_PER_FRAME; + + word_size = AXI_I2S_BITS_PER_FRAME / 2 - 1; + bclk_div = DIV_ROUND_UP(clk_get_rate(i2s->clk_ref), bclk_rate) / 2 - 1; + + regmap_write(i2s->regmap, AXI_I2S_REG_CLK_CTRL, (word_size << 16) | + bclk_div); + + return 0; +} + +static int axi_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); + uint32_t mask; + int ret; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = AXI_I2S_RESET_RX_FIFO; + else + mask = AXI_I2S_RESET_TX_FIFO; + + regmap_write(i2s->regmap, AXI_I2S_REG_RESET, mask); + + ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &i2s->rate_constraints); + if (ret) + return ret; + + return clk_prepare_enable(i2s->clk_ref); +} + +static void axi_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(i2s->clk_ref); +} + +static int axi_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, + &i2s->capture_dma_data); + + return 0; +} + +static const struct snd_soc_dai_ops axi_i2s_dai_ops = { + .startup = axi_i2s_startup, + .shutdown = axi_i2s_shutdown, + .trigger = axi_i2s_trigger, + .hw_params = axi_i2s_hw_params, +}; + +static struct snd_soc_dai_driver axi_i2s_dai = { + .probe = axi_i2s_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE, + }, + .ops = &axi_i2s_dai_ops, + .symmetric_rates = 1, +}; + +static const struct snd_soc_component_driver axi_i2s_component = { + .name = "axi-i2s", +}; + +static const struct regmap_config axi_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = AXI_I2S_REG_STATUS, +}; + +static int axi_i2s_probe(struct platform_device *pdev) +{ + struct resource *res; + struct axi_i2s *i2s; + void __iomem *base; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) + return -ENOMEM; + + platform_set_drvdata(pdev, i2s); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_request_and_ioremap(&pdev->dev, res); + if (!base) + return -EBUSY; + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &axi_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) + return PTR_ERR(i2s->regmap); + + i2s->clk = devm_clk_get(&pdev->dev, "axi"); + if (IS_ERR(i2s->clk)) + return PTR_ERR(i2s->clk); + + i2s->clk_ref = devm_clk_get(&pdev->dev, "ref"); + if (IS_ERR(i2s->clk_ref)) + return PTR_ERR(i2s->clk_ref); + + ret = clk_prepare_enable(i2s->clk); + if (ret) + return ret; + + i2s->playback_dma_data.addr = res->start + AXI_I2S_REG_TX_FIFO; + i2s->playback_dma_data.addr_width = 4; + i2s->playback_dma_data.maxburst = 1; + + i2s->capture_dma_data.addr = res->start + AXI_I2S_REG_RX_FIFO; + i2s->capture_dma_data.addr_width = 4; + i2s->capture_dma_data.maxburst = 1; + + i2s->ratnum.num = clk_get_rate(i2s->clk_ref) / 2 / AXI_I2S_BITS_PER_FRAME; + i2s->ratnum.den_step = 1; + i2s->ratnum.den_min = 1; + i2s->ratnum.den_max = 64; + + i2s->rate_constraints.rats = &i2s->ratnum; + i2s->rate_constraints.nrats = 1; + + regmap_write(i2s->regmap, AXI_I2S_REG_RESET, AXI_I2S_RESET_GLOBAL); + + ret = devm_snd_soc_register_component(&pdev->dev, &axi_i2s_component, + &axi_i2s_dai, 1); + if (ret) + goto err_clk_disable; + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + if (ret) + goto err_clk_disable; + +err_clk_disable: + clk_disable_unprepare(i2s->clk); + return ret; +} + +static int axi_i2s_dev_remove(struct platform_device *pdev) +{ + struct axi_i2s *i2s = platform_get_drvdata(pdev); + + clk_disable_unprepare(i2s->clk); + + return 0; +} + +static const struct of_device_id axi_i2s_of_match[] = { + { .compatible = "adi,axi-i2s-1.00.a", }, + {}, +}; +MODULE_DEVICE_TABLE(of, axi_i2s_of_match); + +static struct platform_driver axi_i2s_driver = { + .driver = { + .name = "axi-i2s", + .owner = THIS_MODULE, + .of_match_table = axi_i2s_of_match, + }, + .probe = axi_i2s_probe, + .remove = axi_i2s_dev_remove, +}; +module_platform_driver(axi_i2s_driver); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("AXI I2S driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From 429e4374cc007f260c7d0e2d0df5247deeaf8fbe Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 6 Dec 2013 11:02:52 +0100 Subject: ASoC: Add support for the Analog Devices AXI-SPDIF driver This patch adds a ASoC driver for the AXI-SPDIF softcore. The core implements a simple SPDIF transmitter and is used on some Analog Devices' reference designs for various FPGA platforms. For now the driver only support the PL330 as the the DMA controller. The driver uses the generic PCM dmaengine driver for its PCM. The only restriction is that we need to set the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag as the dmaengine driver for the DMA core (PL330) that is used with this core has no residue reporting capabilities yet. This will be fixed in the future though. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/adi/Kconfig | 8 ++ sound/soc/adi/Makefile | 2 + sound/soc/adi/axi-spdif.c | 272 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 282 insertions(+) create mode 100644 sound/soc/adi/axi-spdif.c (limited to 'sound/soc') diff --git a/sound/soc/adi/Kconfig b/sound/soc/adi/Kconfig index 46f4b79f7a8..dd763f55eda 100644 --- a/sound/soc/adi/Kconfig +++ b/sound/soc/adi/Kconfig @@ -11,3 +11,11 @@ config SND_SOC_ADI_AXI_I2S select REGMAP_MMIO help ASoC driver for the Analog Devices AXI-I2S softcore peripheral. + +config SND_SOC_ADI_AXI_SPDIF + tristate "AXI-SPDIF support" + depends on SND_SOC_ADI + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + ASoC driver for the Analog Devices AXI-SPDIF softcore peripheral. diff --git a/sound/soc/adi/Makefile b/sound/soc/adi/Makefile index d32c21ae14b..64456c1e534 100644 --- a/sound/soc/adi/Makefile +++ b/sound/soc/adi/Makefile @@ -1,3 +1,5 @@ snd-soc-adi-axi-i2s-objs := axi-i2s.o +snd-soc-adi-axi-spdif-objs := axi-spdif.o obj-$(CONFIG_SND_SOC_ADI_AXI_I2S) += snd-soc-adi-axi-i2s.o +obj-$(CONFIG_SND_SOC_ADI_AXI_SPDIF) += snd-soc-adi-axi-spdif.o diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c new file mode 100644 index 00000000000..d5408d23b43 --- /dev/null +++ b/sound/soc/adi/axi-spdif.c @@ -0,0 +1,272 @@ +/* + * Copyright (C) 2012-2013, Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#define AXI_SPDIF_REG_CTRL 0x0 +#define AXI_SPDIF_REG_STAT 0x4 +#define AXI_SPDIF_REG_TX_FIFO 0xc + +#define AXI_SPDIF_CTRL_TXDATA BIT(1) +#define AXI_SPDIF_CTRL_TXEN BIT(0) +#define AXI_SPDIF_CTRL_CLKDIV_OFFSET 8 +#define AXI_SPDIF_CTRL_CLKDIV_MASK (0xff << 8) + +#define AXI_SPDIF_FREQ_44100 (0x0 << 6) +#define AXI_SPDIF_FREQ_48000 (0x1 << 6) +#define AXI_SPDIF_FREQ_32000 (0x2 << 6) +#define AXI_SPDIF_FREQ_NA (0x3 << 6) + +struct axi_spdif { + struct regmap *regmap; + struct clk *clk; + struct clk *clk_ref; + + struct snd_dmaengine_dai_dma_data dma_data; + + struct snd_ratnum ratnum; + struct snd_pcm_hw_constraint_ratnums rate_constraints; +}; + +static int axi_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); + unsigned int val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = AXI_SPDIF_CTRL_TXDATA; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, + AXI_SPDIF_CTRL_TXDATA, val); + + return 0; +} + +static int axi_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + unsigned int clkdiv, stat; + + switch (params_rate(params)) { + case 32000: + stat = AXI_SPDIF_FREQ_32000; + break; + case 44100: + stat = AXI_SPDIF_FREQ_44100; + break; + case 48000: + stat = AXI_SPDIF_FREQ_48000; + break; + default: + stat = AXI_SPDIF_FREQ_NA; + break; + } + + clkdiv = DIV_ROUND_CLOSEST(clk_get_rate(spdif->clk_ref), + rate * 64 * 2) - 1; + clkdiv <<= AXI_SPDIF_CTRL_CLKDIV_OFFSET; + + regmap_write(spdif->regmap, AXI_SPDIF_REG_STAT, stat); + regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, + AXI_SPDIF_CTRL_CLKDIV_MASK, clkdiv); + + return 0; +} + +static int axi_spdif_dai_probe(struct snd_soc_dai *dai) +{ + struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &spdif->dma_data, NULL); + + return 0; +} + +static int axi_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &spdif->rate_constraints); + if (ret) + return ret; + + ret = clk_prepare_enable(spdif->clk_ref); + if (ret) + return ret; + + regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, + AXI_SPDIF_CTRL_TXEN, AXI_SPDIF_CTRL_TXEN); + + return 0; +} + +static void axi_spdif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, + AXI_SPDIF_CTRL_TXEN, 0); + + clk_disable_unprepare(spdif->clk_ref); +} + +static const struct snd_soc_dai_ops axi_spdif_dai_ops = { + .startup = axi_spdif_startup, + .shutdown = axi_spdif_shutdown, + .trigger = axi_spdif_trigger, + .hw_params = axi_spdif_hw_params, +}; + +static struct snd_soc_dai_driver axi_spdif_dai = { + .probe = axi_spdif_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &axi_spdif_dai_ops, +}; + +static const struct snd_soc_component_driver axi_spdif_component = { + .name = "axi-spdif", +}; + +static const struct regmap_config axi_spdif_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = AXI_SPDIF_REG_STAT, +}; + +static int axi_spdif_probe(struct platform_device *pdev) +{ + struct axi_spdif *spdif; + struct resource *res; + void __iomem *base; + int ret; + + spdif = devm_kzalloc(&pdev->dev, sizeof(*spdif), GFP_KERNEL); + if (!spdif) + return -ENOMEM; + + platform_set_drvdata(pdev, spdif); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_request_and_ioremap(&pdev->dev, res); + if (!base) + return -EBUSY; + + spdif->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &axi_spdif_regmap_config); + if (IS_ERR(spdif->regmap)) + return PTR_ERR(spdif->regmap); + + spdif->clk = devm_clk_get(&pdev->dev, "axi"); + if (IS_ERR(spdif->clk)) + return PTR_ERR(spdif->clk); + + spdif->clk_ref = devm_clk_get(&pdev->dev, "ref"); + if (IS_ERR(spdif->clk_ref)) + return PTR_ERR(spdif->clk_ref); + + ret = clk_prepare_enable(spdif->clk); + if (ret) + return ret; + + spdif->dma_data.addr = res->start + AXI_SPDIF_REG_TX_FIFO; + spdif->dma_data.addr_width = 4; + spdif->dma_data.maxburst = 1; + + spdif->ratnum.num = clk_get_rate(spdif->clk_ref) / 128; + spdif->ratnum.den_step = 1; + spdif->ratnum.den_min = 1; + spdif->ratnum.den_max = 64; + + spdif->rate_constraints.rats = &spdif->ratnum; + spdif->rate_constraints.nrats = 1; + + ret = devm_snd_soc_register_component(&pdev->dev, &axi_spdif_component, + &axi_spdif_dai, 1); + if (ret) + goto err_clk_disable; + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + if (ret) + goto err_clk_disable; + + return 0; + +err_clk_disable: + clk_disable_unprepare(spdif->clk); + return ret; +} + +static int axi_spdif_dev_remove(struct platform_device *pdev) +{ + struct axi_spdif *spdif = platform_get_drvdata(pdev); + + clk_disable_unprepare(spdif->clk); + + return 0; +} + +static const struct of_device_id axi_spdif_of_match[] = { + { .compatible = "adi,axi-spdif-tx-1.00.a", }, + {}, +}; +MODULE_DEVICE_TABLE(of, axi_spdif_of_match); + +static struct platform_driver axi_spdif_driver = { + .driver = { + .name = "axi-spdif", + .owner = THIS_MODULE, + .of_match_table = axi_spdif_of_match, + }, + .probe = axi_spdif_probe, + .remove = axi_spdif_dev_remove, +}; +module_platform_driver(axi_spdif_driver); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("AXI SPDIF driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From 58381da687742a3d8bbb98362152de8326a0c077 Mon Sep 17 00:00:00 2001 From: Jan Weitzel Date: Thu, 5 Dec 2013 09:54:02 +0100 Subject: ASoC: tlv320aic3x: no mono controls 3007 model if codec driver is used for AIC3X_MODEL_3007 the mono iout controls overwrite registers for class-d amplifier. classd amplifier controls are only used for AIC3X_MODEL_3007. Removing all mono snd_kcontrol_new, snd_soc_dapm_widget, snd_soc_dapm_route and aic3x_init stuff from common code and call only for not AIC3X_MODEL_3007 codecs. Testet only with AIC3X_MODEL_3007 Signed-off-by: Jan Weitzel Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 126 +++++++++++++++++++++++++++-------------- 1 file changed, 83 insertions(+), 43 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 546d16b7d38..470fbfb4b38 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -350,16 +350,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", - LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume", - PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", - DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), @@ -383,7 +373,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output pin mute controls */ SOC_DOUBLE_R("Line Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, 0x01, 0), - SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, @@ -412,6 +401,20 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; +static const struct snd_kcontrol_new aic3x_mono_controls[] = { + SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", + LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume", + PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", + DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), +}; + /* * Class-D amplifier gain. From 0 to 18 dB in 6 dB steps */ @@ -565,9 +568,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_PGA("Right HP Out", HPROUT_CTRL, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Right HP Com", HPRCOM_CTRL, 0, 0, NULL, 0), - /* Mono Output */ - SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), - /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, @@ -626,9 +626,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0, &aic3x_right_line_mixer_controls[0], ARRAY_SIZE(aic3x_right_line_mixer_controls)), - SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_mono_mixer_controls[0], - ARRAY_SIZE(aic3x_mono_mixer_controls)), SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_hp_mixer_controls[0], ARRAY_SIZE(aic3x_left_hp_mixer_controls)), @@ -644,7 +641,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LLOUT"), SND_SOC_DAPM_OUTPUT("RLOUT"), - SND_SOC_DAPM_OUTPUT("MONO_LOUT"), SND_SOC_DAPM_OUTPUT("HPLOUT"), SND_SOC_DAPM_OUTPUT("HPROUT"), SND_SOC_DAPM_OUTPUT("HPLCOM"), @@ -666,6 +662,17 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Detection"), }; +static const struct snd_soc_dapm_widget aic3x_dapm_mono_widgets[] = { + /* Mono Output */ + SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_mono_mixer_controls[0], + ARRAY_SIZE(aic3x_mono_mixer_controls)), + + SND_SOC_DAPM_OUTPUT("MONO_LOUT"), +}; + static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = { /* Class-D outputs */ SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0), @@ -754,17 +761,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line Out", NULL, "Right DAC Mux"}, {"RLOUT", NULL, "Right Line Out"}, - /* Mono Output */ - {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, - {"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, - {"Mono Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, - {"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, - {"Mono Mixer", "DACR1 Switch", "Right DAC Mux"}, - - {"Mono Out", NULL, "Mono Mixer"}, - {"MONO_LOUT", NULL, "Mono Out"}, - /* Left HP Output */ {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, @@ -820,6 +816,18 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPRCOM", NULL, "Right HP Com"}, }; +static const struct snd_soc_dapm_route intercon_mono[] = { + /* Mono Output */ + {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Mono Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Mono Mixer", "DACR1 Switch", "Right DAC Mux"}, + {"Mono Out", NULL, "Mono Mixer"}, + {"MONO_LOUT", NULL, "Mono Out"}, +}; + static const struct snd_soc_dapm_route intercon_3007[] = { /* Class-D outputs */ {"Left Class-D Out", NULL, "Left Line Out"}, @@ -833,11 +841,20 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - if (aic3x->model == AIC3X_MODEL_3007) { + switch (aic3x->model) { + case AIC3X_MODEL_3X: + case AIC3X_MODEL_33: + snd_soc_dapm_new_controls(dapm, aic3x_dapm_mono_widgets, + ARRAY_SIZE(aic3x_dapm_mono_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_mono, + ARRAY_SIZE(intercon_mono)); + break; + case AIC3X_MODEL_3007: snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, ARRAY_SIZE(aic3007_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon_3007, ARRAY_SIZE(intercon_3007)); + break; } return 0; @@ -1218,6 +1235,24 @@ static int aic3x_resume(struct snd_soc_codec *codec) return 0; } +static void aic3x_mono_init(struct snd_soc_codec *codec) +{ + /* DAC to Mono Line Out default volume and route to Output mixer */ + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + + /* unmute all outputs */ + snd_soc_update_bits(codec, MONOLOPM_CTRL, UNMUTE, UNMUTE); + + /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); + + /* Line2 to Mono Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); +} + /* * initialise the AIC3X driver * register the mixer and dsp interfaces with the kernel @@ -1241,14 +1276,10 @@ static int aic3x_init(struct snd_soc_codec *codec) /* DAC to Line Out default volume and route to Output mixer */ snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - /* DAC to Mono Line Out default volume and route to Output mixer */ - snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* unmute all outputs */ snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE); snd_soc_update_bits(codec, RLOPM_CTRL, UNMUTE, UNMUTE); - snd_soc_update_bits(codec, MONOLOPM_CTRL, UNMUTE, UNMUTE); snd_soc_update_bits(codec, HPLOUT_CTRL, UNMUTE, UNMUTE); snd_soc_update_bits(codec, HPROUT_CTRL, UNMUTE, UNMUTE); snd_soc_update_bits(codec, HPLCOM_CTRL, UNMUTE, UNMUTE); @@ -1269,9 +1300,6 @@ static int aic3x_init(struct snd_soc_codec *codec) /* PGA to Line Out default volume, disconnect from Output Mixer */ snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); - /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); @@ -1281,12 +1309,15 @@ static int aic3x_init(struct snd_soc_codec *codec) /* Line2 Line Out default volume, disconnect from Output Mixer */ snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); - /* Line2 to Mono Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); - if (aic3x->model == AIC3X_MODEL_3007) { + switch (aic3x->model) { + case AIC3X_MODEL_3X: + case AIC3X_MODEL_33: + aic3x_mono_init(codec); + break; + case AIC3X_MODEL_3007: snd_soc_write(codec, CLASSD_CTRL, 0); + break; } return 0; @@ -1343,8 +1374,17 @@ static int aic3x_probe(struct snd_soc_codec *codec) (aic3x->setup->gpio_func[1] & 0xf) << 4); } - if (aic3x->model == AIC3X_MODEL_3007) - snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); + switch (aic3x->model) { + case AIC3X_MODEL_3X: + case AIC3X_MODEL_33: + snd_soc_add_codec_controls(codec, aic3x_mono_controls, + ARRAY_SIZE(aic3x_mono_controls)); + break; + case AIC3X_MODEL_3007: + snd_soc_add_codec_controls(codec, + &aic3x_classd_amp_gain_ctrl, 1); + break; + } /* set mic bias voltage */ switch (aic3x->micbias_vg) { -- cgit v1.2.3-70-g09d2 From 11b3a7add2841aa698aa0a32396b6db413c22eda Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 3 Dec 2013 14:26:32 -0700 Subject: ASoC: restructure dmaengine_pcm_request_chan_of() Restructure the internals of dmaengine_pcm_request_chan_of() as a loop over all channels to be allocated. This makes it easier to add logic that applies to all allocated channels, without having to duplicate that logic in each of the half-duplex/full-duplex paths. Signed-off-by: Stephen Warren Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 87e86357124..fbc28a799e0 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -291,21 +291,26 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, struct device *dev) { unsigned int i; + const char *name; if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || !dev->of_node) return; - if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { - pcm->chan[0] = dma_request_slave_channel(dev, "rx-tx"); - pcm->chan[1] = pcm->chan[0]; - } else { - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = dma_request_slave_channel(dev, - dmaengine_pcm_dma_channel_names[i]); - } + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; + i++) { + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + name = "rx-tx"; + else + name = dmaengine_pcm_dma_channel_names[i]; + pcm->chan[i] = dma_request_slave_channel(dev, name); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; } + + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + pcm->chan[1] = pcm->chan[0]; } /** -- cgit v1.2.3-70-g09d2 From 194c7dea00c68c1b1f8ff26304fa937a006f66dd Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 3 Dec 2013 14:26:34 -0700 Subject: ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config Add fields to struct snd_dmaengine_pcm_config to allow custom: - DMA channel names. This is useful when the default "tx" and "rx" channel names don't apply, for example if a HW module supports multiple channels, each having different DMA channel names. This is the case with the FIFOs in Tegra's AHUB. This new facility can replace SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME. - DMA device This allows requesting DMA channels for a device other than the device which is registering the "PCM" driver. This is quite unusual, but is currently useful on Tegra. In much HW, and in Tegra20, each DAI HW module contains its own FIFOs which DMA writes to. However, in Tegra30, the DMA FIFOs were split out AHUB HW module, which then routes the data through a cross-bar, and into the DAI HW modules. However, the current ASoC driver structure does not expose this detail, and acts as if the FIFOs are still part of the DAI HW modules. Consequently, the "PCM" driver is registered with the DAI HW module, yet the DMA channels must be looked up in the AHUB HW module's device tree node. This new config field allows that to happen. Eventually, the Tegra drivers will be reworked to fully expose the AHUB, and this config field can be removed. Signed-off-by: Stephen Warren Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/dmaengine_pcm.h | 6 ++++++ sound/soc/soc-generic-dmaengine-pcm.c | 18 ++++++++++++++++-- 2 files changed, 22 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 4ef986cab18..eb73a3a39ec 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -114,6 +114,10 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * @compat_filter_fn: Will be used as the filter function when requesting a * channel for platforms which do not use devicetree. The filter parameter * will be the DAI's DMA data. + * @dma_dev: If set, request DMA channel on this device rather than the DAI + * device. + * @chan_names: If set, these custom DMA channel names will be requested at + * registration time. * @pcm_hardware: snd_pcm_hardware struct to be used for the PCM. * @prealloc_buffer_size: Size of the preallocated audio buffer. * @@ -130,6 +134,8 @@ struct snd_dmaengine_pcm_config { struct snd_soc_pcm_runtime *rtd, struct snd_pcm_substream *substream); dma_filter_fn compat_filter_fn; + struct device *dma_dev; + const char *chan_names[SNDRV_PCM_STREAM_LAST + 1]; const struct snd_pcm_hardware *pcm_hardware; unsigned int prealloc_buffer_size; diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 1cb3494cf27..5b70c556fba 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -288,7 +288,7 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { }; static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, - struct device *dev) + struct device *dev, const struct snd_dmaengine_pcm_config *config) { unsigned int i; const char *name; @@ -298,12 +298,26 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, !dev->of_node) return; + if (config->dma_dev) { + /* + * If this warning is seen, it probably means that your Linux + * device structure does not match your HW device structure. + * It would be best to refactor the Linux device structure to + * correctly match the HW structure. + */ + dev_warn(dev, "DMA channels sourced from device %s", + dev_name(config->dma_dev)); + dev = config->dma_dev; + } + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) name = "rx-tx"; else name = dmaengine_pcm_dma_channel_names[i]; + if (config->chan_names[i]) + name = config->chan_names[i]; pcm->chan[i] = dma_request_slave_channel(dev, name); if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) break; @@ -346,7 +360,7 @@ int snd_dmaengine_pcm_register(struct device *dev, pcm->config = config; pcm->flags = flags; - dmaengine_pcm_request_chan_of(pcm, dev); + dmaengine_pcm_request_chan_of(pcm, dev, config); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) ret = snd_soc_add_platform(dev, &pcm->platform, -- cgit v1.2.3-70-g09d2 From 1a1c75a7982ca1181bb84bb9e83f9f7c752fb104 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Dec 2013 17:26:01 -0800 Subject: ASoC: rsnd: gen: fixup Gen2 channel size Gen2 has 0 - 9, total 10 channels, not 9 channels. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index bf066f73ef0..d0ab203836d 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -217,11 +217,11 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, /* single address mapping */ #define RSND_GEN2_S_REG(gen, reg, id, offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 9) + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 10) /* multi address mapping */ #define RSND_GEN2_M_REG(gen, reg, id, offset, _id_offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 9) + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 10) static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) { -- cgit v1.2.3-70-g09d2 From 860d0c0dd2e7dbf98b47f38d80793137bc6c6ebc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:17 +0200 Subject: ASoC: davinci: Kconfig: Remove help section for SND_DAVINCI_SOC The help text is misleading and the prompt itself explains the purpose of this config section. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 95970f5db3e..fb9182659de 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,11 +1,6 @@ config SND_DAVINCI_SOC tristate "SoC Audio for the TI DAVINCI or AM33XX chip" depends on ARCH_DAVINCI || SOC_AM33XX - help - Platform driver for daVinci or AM33xx - Say Y or M if you want to add support for codecs attached to - the DAVINCI AC97, I2S, or McASP interface. You will also need - to select the audio interfaces to support below. config SND_DAVINCI_SOC_I2S tristate -- cgit v1.2.3-70-g09d2 From c3238a4c058edd1528f0bec9a37fe79479e9e1a8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:18 +0200 Subject: ASoC: davinci: Kconfig/Makefile: Generic EVM machine driver related cleanup We have several boards using the same machine driver for audio support. All of these machines can select a generic machine driver config option to build the needed driver while keeping the config options used within the driver for compile time code path selection. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 20 ++++++++++---------- sound/soc/davinci/Makefile | 8 ++------ 2 files changed, 12 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index fb9182659de..be667719d44 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -11,11 +11,15 @@ config SND_DAVINCI_SOC_MCASP config SND_DAVINCI_SOC_VCIF tristate +config SND_DAVINCI_SOC_GENERIC_EVM + tristate + select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_MCASP + config SND_AM33XX_SOC_EVM tristate "SoC Audio for the AM33XX chip based boards" depends on SND_DAVINCI_SOC && SOC_AM33XX - select SND_SOC_TLV320AIC3X - select SND_DAVINCI_SOC_MCASP + select SND_DAVINCI_SOC_GENERIC_EVM help Say Y or M if you want to add support for SoC audio on AM33XX boards using McASP and TLV320AIC3X codec. For example AM335X-EVM, @@ -26,8 +30,7 @@ config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM - select SND_DAVINCI_SOC_I2S - select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_GENERIC_EVM help Say Y if you want to add support for SoC audio on TI DaVinci DM6446, DM355 or DM365 EVM platforms. @@ -54,8 +57,7 @@ endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM - select SND_DAVINCI_SOC_MCASP - select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_GENERIC_EVM select SND_SOC_SPDIF help @@ -64,8 +66,7 @@ config SND_DM6467_SOC_EVM config SND_DA830_SOC_EVM tristate "SoC Audio support for DA830/OMAP-L137 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM - select SND_DAVINCI_SOC_MCASP - select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_GENERIC_EVM help Say Y if you want to add support for SoC audio on TI @@ -74,8 +75,7 @@ config SND_DA830_SOC_EVM config SND_DA850_SOC_EVM tristate "SoC Audio support for DA850/OMAP-L138 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM - select SND_DAVINCI_SOC_MCASP - select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_GENERIC_EVM help Say Y if you want to add support for SoC audio on TI DA850/OMAP-L138 EVM diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index bc81e79fc30..744d4d9a018 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -9,11 +9,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o -# DAVINCI Machine Support +# Generic DAVINCI/AM33xx Machine Support snd-soc-evm-objs := davinci-evm.o -obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o -obj-$(CONFIG_SND_AM33XX_SOC_EVM) += snd-soc-evm.o -obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o -obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o -obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DAVINCI_SOC_GENERIC_EVM) += snd-soc-evm.o -- cgit v1.2.3-70-g09d2 From a42efd97f7471c78617c6329ed39919e2f31a7cc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:19 +0200 Subject: ASoC: davinci: kconfig: Prepare for AM43xx support AM43xx have the same McASP IP as AM33xx and both platform uses eDMA. Modify the Kconfig so it will be possible to add audio support for AM43xx based boards later. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt | 2 +- sound/soc/davinci/Kconfig | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index ed785b3f67b..1eed972d471 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -4,7 +4,7 @@ Required properties: - compatible : "ti,dm646x-mcasp-audio" : for DM646x platforms "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, TI81xx) + "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - reg : Should contain reg specifiers for the entries in the reg-names property. - reg-names : Should contain: diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index be667719d44..a8ec1fc3e4d 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,6 +1,6 @@ config SND_DAVINCI_SOC - tristate "SoC Audio for the TI DAVINCI or AM33XX chip" - depends on ARCH_DAVINCI || SOC_AM33XX + tristate "SoC Audio for TI DAVINCI or AM33XX/AM43XX chips" + depends on ARCH_DAVINCI || SOC_AM33XX || SOC_AM43XX config SND_DAVINCI_SOC_I2S tristate -- cgit v1.2.3-70-g09d2 From 7b6a772932eca1ed8d22414a1ba985c5232369d0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:20 +0200 Subject: ASoC: davinci-evm: Do not include davinci-mcasp.h There's no need to include this header file here. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 623eb5e7c08..2a00e2dbc9e 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -28,7 +28,6 @@ #include "davinci-pcm.h" #include "davinci-i2s.h" -#include "davinci-mcasp.h" struct snd_soc_card_drvdata_davinci { unsigned sysclk; -- cgit v1.2.3-70-g09d2 From d38970e1363ddf63fa2f681f0c9b47fc91ca3961 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:21 +0200 Subject: ASoC: davinci-evm: Switch to use .dai_fmt of snd_soc_dai_link(s) Specify the dai formats to use within the snd_soc_dai_link structures. In this way we can remove the code dealing with the dai format configuration from the machin driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 47 ++++++++++++++++------------------------- 1 file changed, 18 insertions(+), 29 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 2a00e2dbc9e..70ff3772079 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -33,8 +33,6 @@ struct snd_soc_card_drvdata_davinci { unsigned sysclk; }; -#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ - SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -47,16 +45,6 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *) snd_soc_card_get_drvdata(soc_card))->sysclk; - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); - if (ret < 0) - return ret; - /* set the codec system clock */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); if (ret < 0) @@ -70,24 +58,10 @@ static int evm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int evm_spdif_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - /* set cpu DAI configuration */ - return snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); -} - static struct snd_soc_ops evm_ops = { .hw_params = evm_hw_params, }; -static struct snd_soc_ops evm_spdif_ops = { - .hw_params = evm_spdif_hw_params, -}; - /* davinci-evm machine dapm widgets */ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -164,6 +138,8 @@ static struct snd_soc_dai_link dm6446_evm_dai = { .platform_name = "davinci-mcbsp", .init = evm_aic3x_init, .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; static struct snd_soc_dai_link dm355_evm_dai = { @@ -175,6 +151,8 @@ static struct snd_soc_dai_link dm355_evm_dai = { .platform_name = "davinci-mcbsp.1", .init = evm_aic3x_init, .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; static struct snd_soc_dai_link dm365_evm_dai = { @@ -183,10 +161,12 @@ static struct snd_soc_dai_link dm365_evm_dai = { .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "tlv320aic3x-hifi", - .init = evm_aic3x_init, .codec_name = "tlv320aic3x-codec.1-0018", - .ops = &evm_ops, .platform_name = "davinci-mcbsp", + .init = evm_aic3x_init, + .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, #elif defined(CONFIG_SND_DM365_VOICE_CODEC) .name = "Voice Codec - CQ93VC", .stream_name = "CQ93", @@ -207,6 +187,8 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .codec_name = "tlv320aic3x-codec.0-001a", .init = evm_aic3x_init, .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }, { .name = "McASP", @@ -215,7 +197,8 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .codec_dai_name = "dit-hifi", .codec_name = "spdif_dit", .platform_name = "davinci-mcasp.1", - .ops = &evm_spdif_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }, }; @@ -228,6 +211,8 @@ static struct snd_soc_dai_link da830_evm_dai = { .platform_name = "davinci-mcasp.1", .init = evm_aic3x_init, .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; static struct snd_soc_dai_link da850_evm_dai = { @@ -239,6 +224,8 @@ static struct snd_soc_dai_link da850_evm_dai = { .platform_name = "davinci-mcasp.0", .init = evm_aic3x_init, .ops = &evm_ops, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; /* davinci dm6446 evm audio machine driver */ @@ -335,6 +322,8 @@ static struct snd_soc_dai_link evm_dai_tlv320aic3x = { .codec_dai_name = "tlv320aic3x-hifi", .ops = &evm_ops, .init = evm_aic3x_init, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; static const struct of_device_id davinci_evm_dt_ids[] = { -- cgit v1.2.3-70-g09d2 From ed29cd5e8d9941353f82784a2478ba6babc828da Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:22 +0200 Subject: ASoC: davinci-mcasp: Move DAVINCI_MCASP_RATE from header to source file It is not used outside of the .c file. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 2 ++ sound/soc/davinci/davinci-mcasp.h | 1 - 2 files changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 71e14bb3a8c..9763a5d7555 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -955,6 +955,8 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .set_sysclk = davinci_mcasp_set_sysclk, }; +#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000 + #define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_U8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a2e27e1c32f..a84e79604df 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -23,7 +23,6 @@ #include "davinci-pcm.h" -#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000 #define DAVINCI_MCASP_I2S_DAI 0 #define DAVINCI_MCASP_DIT_DAI 1 -- cgit v1.2.3-70-g09d2 From 18f93506623aacbb269f47cbda9fe90ffc5acda6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:23 +0200 Subject: ASoC: davinci-mcasp: Remove unused DAVINCI_MCASP_I2S/DIT_DAI defines These are not used, probably leftovers from the past. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a84e79604df..70b089b4e0f 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -23,9 +23,6 @@ #include "davinci-pcm.h" -#define DAVINCI_MCASP_I2S_DAI 0 -#define DAVINCI_MCASP_DIT_DAI 1 - struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; -- cgit v1.2.3-70-g09d2 From 58e48d9774d4d8fc5e0785dbd2ccf075b248ad96 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:24 +0200 Subject: ASoC: davinci-mcasp: Correct dai driver struct initialization for 2nd dai Add .name when assigning the dai name. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9763a5d7555..6cde1babc11 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -987,7 +987,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { }, { - "davinci-mcasp.1", + .name = "davinci-mcasp.1", .playback = { .channels_min = 1, .channels_max = 384, -- cgit v1.2.3-70-g09d2 From 02e08d9b6bd67784d4c58e659c21674b31972c34 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:25 +0200 Subject: ASoC: davinci-mcasp: Move register definitions to header file It is better for readability to have the register definitions out from the source file. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 267 -------------------------------------- sound/soc/davinci/davinci-mcasp.h | 267 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 267 insertions(+), 267 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6cde1babc11..1c1585e18ea 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -35,273 +35,6 @@ #include "davinci-pcm.h" #include "davinci-mcasp.h" -/* - * McASP register definitions - */ -#define DAVINCI_MCASP_PID_REG 0x00 -#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 - -#define DAVINCI_MCASP_PFUNC_REG 0x10 -#define DAVINCI_MCASP_PDIR_REG 0x14 -#define DAVINCI_MCASP_PDOUT_REG 0x18 -#define DAVINCI_MCASP_PDSET_REG 0x1c - -#define DAVINCI_MCASP_PDCLR_REG 0x20 - -#define DAVINCI_MCASP_TLGC_REG 0x30 -#define DAVINCI_MCASP_TLMR_REG 0x34 - -#define DAVINCI_MCASP_GBLCTL_REG 0x44 -#define DAVINCI_MCASP_AMUTE_REG 0x48 -#define DAVINCI_MCASP_LBCTL_REG 0x4c - -#define DAVINCI_MCASP_TXDITCTL_REG 0x50 - -#define DAVINCI_MCASP_GBLCTLR_REG 0x60 -#define DAVINCI_MCASP_RXMASK_REG 0x64 -#define DAVINCI_MCASP_RXFMT_REG 0x68 -#define DAVINCI_MCASP_RXFMCTL_REG 0x6c - -#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 -#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 -#define DAVINCI_MCASP_RXTDM_REG 0x78 -#define DAVINCI_MCASP_EVTCTLR_REG 0x7c - -#define DAVINCI_MCASP_RXSTAT_REG 0x80 -#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 -#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 -#define DAVINCI_MCASP_REVTCTL_REG 0x8c - -#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 -#define DAVINCI_MCASP_TXMASK_REG 0xa4 -#define DAVINCI_MCASP_TXFMT_REG 0xa8 -#define DAVINCI_MCASP_TXFMCTL_REG 0xac - -#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 -#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 -#define DAVINCI_MCASP_TXTDM_REG 0xb8 -#define DAVINCI_MCASP_EVTCTLX_REG 0xbc - -#define DAVINCI_MCASP_TXSTAT_REG 0xc0 -#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 -#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 -#define DAVINCI_MCASP_XEVTCTL_REG 0xcc - -/* Left(even TDM Slot) Channel Status Register File */ -#define DAVINCI_MCASP_DITCSRA_REG 0x100 -/* Right(odd TDM slot) Channel Status Register File */ -#define DAVINCI_MCASP_DITCSRB_REG 0x118 -/* Left(even TDM slot) User Data Register File */ -#define DAVINCI_MCASP_DITUDRA_REG 0x130 -/* Right(odd TDM Slot) User Data Register File */ -#define DAVINCI_MCASP_DITUDRB_REG 0x148 - -/* Serializer n Control Register */ -#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 -#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ - (n << 2)) - -/* Transmit Buffer for Serializer n */ -#define DAVINCI_MCASP_TXBUF_REG 0x200 -/* Receive Buffer for Serializer n */ -#define DAVINCI_MCASP_RXBUF_REG 0x280 - -/* McASP FIFO Registers */ -#define DAVINCI_MCASP_WFIFOCTL (0x1010) -#define DAVINCI_MCASP_WFIFOSTS (0x1014) -#define DAVINCI_MCASP_RFIFOCTL (0x1018) -#define DAVINCI_MCASP_RFIFOSTS (0x101C) -#define MCASP_VER3_WFIFOCTL (0x1000) -#define MCASP_VER3_WFIFOSTS (0x1004) -#define MCASP_VER3_RFIFOCTL (0x1008) -#define MCASP_VER3_RFIFOSTS (0x100C) - -/* - * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management - * Register Bits - */ -#define MCASP_FREE BIT(0) -#define MCASP_SOFT BIT(1) - -/* - * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits - */ -#define AXR(n) (1< Date: Thu, 14 Nov 2013 11:35:26 +0200 Subject: ASoC: davinci-mcasp: Move private struct definition to source file Since it is a private struct strictly used by the davinci-mcasp driver it can be moved from header file to the source file. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 30 ++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 35 ----------------------------------- 2 files changed, 30 insertions(+), 35 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1c1585e18ea..70107956dd0 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -37,6 +37,36 @@ #define DAVINCI_MCASP_NUM_SERIALIZER 16 +struct davinci_audio_dev { + struct davinci_pcm_dma_params dma_params[2]; + void __iomem *base; + struct device *dev; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; + u8 version; + u16 bclk_lrclk_ratio; + + /* McASP FIFO related */ + u8 txnumevt; + u8 rxnumevt; + +#ifdef CONFIG_PM_SLEEP + struct { + u32 txfmtctl; + u32 rxfmtctl; + u32 txfmt; + u32 rxfmt; + u32 aclkxctl; + u32 aclkrctl; + u32 pdir; + } context; +#endif +}; + static inline void mcasp_set_bits(void __iomem *reg, u32 val) { __raw_writel(__raw_readl(reg) | val, reg); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 619b98befc8..80e5a184668 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -18,11 +18,6 @@ #ifndef DAVINCI_MCASP_H #define DAVINCI_MCASP_H -#include -#include - -#include "davinci-pcm.h" - /* * McASP register definitions */ @@ -290,34 +285,4 @@ #define NUMEVT_MASK (0xFF << 8) #define NUMDMA_MASK (0xFF) -struct davinci_audio_dev { - struct davinci_pcm_dma_params dma_params[2]; - void __iomem *base; - struct device *dev; - - /* McASP specific data */ - int tdm_slots; - u8 op_mode; - u8 num_serializer; - u8 *serial_dir; - u8 version; - u16 bclk_lrclk_ratio; - - /* McASP FIFO related */ - u8 txnumevt; - u8 rxnumevt; - -#ifdef CONFIG_PM_SLEEP - struct { - u32 txfmtctl; - u32 rxfmtctl; - u32 txfmt; - u32 rxfmt; - u32 aclkxctl; - u32 aclkrctl; - u32 pdir; - } context; -#endif -}; - #endif /* DAVINCI_MCASP_H */ -- cgit v1.2.3-70-g09d2 From 57f439b8676b2dad63a148de4e61d80b4e196c2a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:27 +0200 Subject: ASoC: davinci-mcasp: Remove unused DAVINCI_MCASP_NUM_SERIALIZER define It is not used in the code. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 70107956dd0..e4c0fb4b978 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -35,8 +35,6 @@ #include "davinci-pcm.h" #include "davinci-mcasp.h" -#define DAVINCI_MCASP_NUM_SERIALIZER 16 - struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; -- cgit v1.2.3-70-g09d2 From eba0ecf067913d60768bb3d11f861b949f072a93 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:28 +0200 Subject: ASoC: davinci-mcasp: Do not inline the mcasp_set_ctl_reg function It brings no benefit to inline this function due to it's size and function. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e4c0fb4b978..ce1607b3c6d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -90,7 +90,7 @@ static inline u32 mcasp_get_reg(void __iomem *reg) return (unsigned int)__raw_readl(reg); } -static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +static void mcasp_set_ctl_reg(void __iomem *regs, u32 val) { int i = 0; -- cgit v1.2.3-70-g09d2 From 70091a3e6aa2e7a05eaefcaec1a43c27a5023eb7 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:29 +0200 Subject: ASoC: davinci-mcasp: Rename private struct and it's users (dev -> mcasp) Rename the private struct from davinci_audio_dev to davinci_mcasp. Change the local use of the pointer to this struct from *dev to *mcasp. The aim is to have better readable code for the first look since having dev->xxxx in the code when using the local private struct is a bit surprising. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 418 +++++++++++++++++++------------------- 1 file changed, 209 insertions(+), 209 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ce1607b3c6d..bd85c98bf5a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -35,7 +35,7 @@ #include "davinci-pcm.h" #include "davinci-mcasp.h" -struct davinci_audio_dev { +struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; struct device *dev; @@ -107,36 +107,36 @@ static void mcasp_set_ctl_reg(void __iomem *regs, u32 val) printk(KERN_ERR "GBLCTL write error\n"); } -static void mcasp_start_rx(struct davinci_audio_dev *dev) +static void mcasp_start_rx(struct davinci_mcasp *mcasp) { - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); } -static void mcasp_start_tx(struct davinci_audio_dev *dev) +static void mcasp_start_tx(struct davinci_mcasp *mcasp) { u8 offset = 0, i; u32 cnt; - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); - mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); - for (i = 0; i < dev->num_serializer; i++) { - if (dev->serial_dir[i] == TX_MODE) { + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); + for (i = 0; i < mcasp->num_serializer; i++) { + if (mcasp->serial_dir[i] == TX_MODE) { offset = i; break; } @@ -144,116 +144,116 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) /* wait for TX ready */ cnt = 0; - while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & + while (!(mcasp_get_reg(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & TXSTATE) && (cnt < 100000)) cnt++; - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); } -static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) +static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) { /* enable FIFO */ - switch (dev->version) { + if (mcasp->txnumevt) { /* enable FIFO */ + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); - mcasp_set_bits(dev->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); + mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL, + FIFO_ENABLE); + mcasp_set_bits(mcasp->base + MCASP_VER3_WFIFOCTL, + FIFO_ENABLE); break; default: - mcasp_clr_bits(dev->base + + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); } } - mcasp_start_tx(dev); + mcasp_start_tx(mcasp); } else { - if (dev->rxnumevt) { /* enable FIFO */ - switch (dev->version) { + if (mcasp->rxnumevt) { /* enable FIFO */ + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); - mcasp_set_bits(dev->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); + mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL, + FIFO_ENABLE); + mcasp_set_bits(mcasp->base + MCASP_VER3_RFIFOCTL, + FIFO_ENABLE); break; default: - mcasp_clr_bits(dev->base + + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } } - mcasp_start_rx(dev); + mcasp_start_rx(mcasp); } } -static void mcasp_stop_rx(struct davinci_audio_dev *dev) +static void mcasp_stop_rx(struct davinci_mcasp *mcasp) { - mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); } -static void mcasp_stop_tx(struct davinci_audio_dev *dev) +static void mcasp_stop_tx(struct davinci_mcasp *mcasp) { - mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); } -static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) +static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) { /* disable FIFO */ - switch (dev->version) { + if (mcasp->txnumevt) { /* disable FIFO */ + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); + mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL, + FIFO_ENABLE); break; default: - mcasp_clr_bits(dev->base + + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); } } - mcasp_stop_tx(dev); + mcasp_stop_tx(mcasp); } else { - if (dev->rxnumevt) { /* disable FIFO */ - switch (dev->version) { + if (mcasp->rxnumevt) { /* disable FIFO */ + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); + mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL, + FIFO_ENABLE); break; default: - mcasp_clr_bits(dev->base + + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } } - mcasp_stop_rx(dev); + mcasp_stop_rx(mcasp); } } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - void __iomem *base = dev->base; + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + void __iomem *base = mcasp->base; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_AC97: - mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); break; default: /* configure a full-word SYNC pulse (LRCLK) */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); /* make 1st data bit occur one ACLK cycle after the frame sync */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); - mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); break; } @@ -342,25 +342,25 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); switch (div_id) { case 0: /* MCLK divider */ - mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(div - 1), AHCLKXDIV_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRDIV(div - 1), AHCLKRDIV_MASK); break; case 1: /* BCLK divider */ - mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXDIV(div - 1), ACLKXDIV_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKRCTL_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRDIV(div - 1), ACLKRDIV_MASK); break; case 2: /* BCLK/LRCLK ratio */ - dev->bclk_lrclk_ratio = div; + mcasp->bclk_lrclk_ratio = div; break; default: @@ -373,22 +373,22 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); if (dir == SND_SOC_CLOCK_OUT) { - mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); } else { - mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); } return 0; } -static int davinci_config_channel_size(struct davinci_audio_dev *dev, +static int davinci_config_channel_size(struct davinci_mcasp *mcasp, int word_length) { u32 fmt; @@ -405,70 +405,70 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, * both left and right channels), so it has to be divided by number of * tdm-slots (for I2S - divided by 2). */ - if (dev->bclk_lrclk_ratio) - word_length = dev->bclk_lrclk_ratio / dev->tdm_slots; + if (mcasp->bclk_lrclk_ratio) + word_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots; /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - if (dev->op_mode != DAVINCI_MCASP_DIT_MODE) { - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXROT(tx_rotate), TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rx_rotate), RXROT(7)); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXMASK_REG, mask); } - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXMASK_REG, mask); return 0; } -static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, +static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; u8 ser; - u8 slots = dev->tdm_slots; + u8 slots = mcasp->tdm_slots; u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); /* All PINS as McASP */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); } else { - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_REVTCTL_REG, RXDATADMADIS); } - for (i = 0; i < dev->num_serializer; i++) { - mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), - dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE && + for (i = 0; i < mcasp->num_serializer; i++) { + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i), + mcasp->serial_dir[i]); + if (mcasp->serial_dir[i] == TX_MODE && tx_ser < max_active_serializers) { - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE && + } else if (mcasp->serial_dir[i] == RX_MODE && rx_ser < max_active_serializers) { - mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; } else { - mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i), SRMOD_INACTIVE, SRMOD_MASK); } } @@ -479,127 +479,127 @@ static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, ser = rx_ser; if (ser < max_active_serializers) { - dev_warn(dev->dev, "stream has more channels (%d) than are " + dev_warn(mcasp->dev, "stream has more channels (%d) than are " "enabled in mcasp (%d)\n", channels, ser * slots); return -EINVAL; } - if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt * tx_ser > 64) - dev->txnumevt = 1; + if (mcasp->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (mcasp->txnumevt * tx_ser > 64) + mcasp->txnumevt = 1; - switch (dev->version) { + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, tx_ser, + mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL, tx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, - ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL, + ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); break; default: - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, - ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, + ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); } } - if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { - if (dev->rxnumevt * rx_ser > 64) - dev->rxnumevt = 1; - switch (dev->version) { + if (mcasp->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { + if (mcasp->rxnumevt * rx_ser > 64) + mcasp->rxnumevt = 1; + switch (mcasp->version) { case MCASP_VERSION_3: - mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, rx_ser, + mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL, rx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, - ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL, + ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); break; default: - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, - ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, + ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } return 0; } -static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) +static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) { int i, active_slots; u32 mask = 0; - active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots; for (i = 0; i < active_slots; i++) mask |= (1 << i); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { /* bit stream is MSB first with no delay */ /* DSP_B mode */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); - mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXORD); - if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32)) - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, - FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); else printk(KERN_ERR "playback tdm slot %d not supported\n", - dev->tdm_slots); + mcasp->tdm_slots); } else { /* bit stream is MSB first with no delay */ /* DSP_B mode */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXTDM_REG, mask); - if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32)) - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, - FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) + mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); else printk(KERN_ERR "capture tdm slot %d not supported\n", - dev->tdm_slots); + mcasp->tdm_slots); } } /* S/PDIF */ -static void davinci_hw_dit_param(struct davinci_audio_dev *dev) +static void davinci_hw_dit_param(struct davinci_mcasp *mcasp) { /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE | FSXMOD(0x180)); /* Set the TX tdm : for all the slots */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); /* Set the TX clock controls : div = 1 and internal */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE | TX_ASYNC); - mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); /* Only 44100 and 48000 are valid, both have the same setting */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); /* Enable the DIT */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); } static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); struct davinci_pcm_dma_params *dma_params = - &dev->dma_params[substream->stream]; + &mcasp->dma_params[substream->stream]; int word_length; u8 fifo_level; - u8 slots = dev->tdm_slots; + u8 slots = mcasp->tdm_slots; u8 active_serializers; int channels; struct snd_interval *pcm_channels = hw_param_interval(params, @@ -608,17 +608,17 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, active_serializers = (channels + slots - 1) / slots; - if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL) + if (davinci_hw_common_param(mcasp, substream->stream, channels) == -EINVAL) return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - fifo_level = dev->txnumevt * active_serializers; + fifo_level = mcasp->txnumevt * active_serializers; else - fifo_level = dev->rxnumevt * active_serializers; + fifo_level = mcasp->rxnumevt * active_serializers; - if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) - davinci_hw_dit_param(dev); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(mcasp); else - davinci_hw_param(dev, substream->stream); + davinci_hw_param(mcasp, substream->stream); switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: @@ -652,13 +652,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (dev->version == MCASP_VERSION_2 && !fifo_level) + if (mcasp->version == MCASP_VERSION_2 && !fifo_level) dma_params->acnt = 4; else dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; - davinci_config_channel_size(dev, word_length); + davinci_config_channel_size(mcasp, word_length); return 0; } @@ -666,29 +666,29 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); int ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = pm_runtime_get_sync(dev->dev); + ret = pm_runtime_get_sync(mcasp->dev); if (IS_ERR_VALUE(ret)) - dev_err(dev->dev, "pm_runtime_get_sync() failed\n"); - davinci_mcasp_start(dev, substream->stream); + dev_err(mcasp->dev, "pm_runtime_get_sync() failed\n"); + davinci_mcasp_start(mcasp, substream->stream); break; case SNDRV_PCM_TRIGGER_SUSPEND: - davinci_mcasp_stop(dev, substream->stream); - ret = pm_runtime_put_sync(dev->dev); + davinci_mcasp_stop(mcasp, substream->stream); + ret = pm_runtime_put_sync(mcasp->dev); if (IS_ERR_VALUE(ret)) - dev_err(dev->dev, "pm_runtime_put_sync() failed\n"); + dev_err(mcasp->dev, "pm_runtime_put_sync() failed\n"); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - davinci_mcasp_stop(dev, substream->stream); + davinci_mcasp_stop(mcasp, substream->stream); break; default: @@ -701,9 +701,9 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); + snd_soc_dai_set_dma_data(dai, substream, mcasp->dma_params); return 0; } @@ -915,7 +915,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct davinci_pcm_dma_params *dma_data; struct resource *mem, *ioarea, *res, *dat; struct snd_platform_data *pdata; - struct davinci_audio_dev *dev; + struct davinci_mcasp *mcasp; int ret; if (!pdev->dev.platform_data && !pdev->dev.of_node) { @@ -923,9 +923,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return -EINVAL; } - dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_audio_dev), + mcasp = devm_kzalloc(&pdev->dev, sizeof(struct davinci_mcasp), GFP_KERNEL); - if (!dev) + if (!mcasp) return -ENOMEM; pdata = davinci_mcasp_set_pdata_from_of(pdev); @@ -936,7 +936,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (!mem) { - dev_warn(dev->dev, + dev_warn(mcasp->dev, "\"mpu\" mem resource not found, using index 0\n"); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { @@ -960,27 +960,27 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return ret; } - dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!dev->base) { + mcasp->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!mcasp->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENOMEM; goto err_release_clk; } - dev->op_mode = pdata->op_mode; - dev->tdm_slots = pdata->tdm_slots; - dev->num_serializer = pdata->num_serializer; - dev->serial_dir = pdata->serial_dir; - dev->version = pdata->version; - dev->txnumevt = pdata->txnumevt; - dev->rxnumevt = pdata->rxnumevt; - dev->dev = &pdev->dev; + mcasp->op_mode = pdata->op_mode; + mcasp->tdm_slots = pdata->tdm_slots; + mcasp->num_serializer = pdata->num_serializer; + mcasp->serial_dir = pdata->serial_dir; + mcasp->version = pdata->version; + mcasp->txnumevt = pdata->txnumevt; + mcasp->rxnumevt = pdata->rxnumevt; + mcasp->dev = &pdev->dev; dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); if (!dat) dat = mem; - dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; @@ -993,7 +993,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->channel = pdata->tx_dma_channel; - dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; + dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; @@ -1006,7 +1006,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->channel = pdata->rx_dma_channel; - dev_set_drvdata(&pdev->dev, dev); + dev_set_drvdata(&pdev->dev, mcasp); ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1); @@ -1044,32 +1044,32 @@ static int davinci_mcasp_remove(struct platform_device *pdev) #ifdef CONFIG_PM_SLEEP static int davinci_mcasp_suspend(struct device *dev) { - struct davinci_audio_dev *a = dev_get_drvdata(dev); - void __iomem *base = a->base; + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + void __iomem *base = mcasp->base; - a->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG); - a->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG); - a->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG); - a->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG); - a->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG); - a->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG); - a->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG); + mcasp->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG); + mcasp->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG); + mcasp->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG); + mcasp->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG); + mcasp->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG); + mcasp->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG); + mcasp->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG); return 0; } static int davinci_mcasp_resume(struct device *dev) { - struct davinci_audio_dev *a = dev_get_drvdata(dev); - void __iomem *base = a->base; - - mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, a->context.txfmtctl); - mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, a->context.rxfmtctl); - mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, a->context.txfmt); - mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, a->context.rxfmt); - mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, a->context.aclkxctl); - mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, a->context.aclkrctl); - mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, a->context.pdir); + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + void __iomem *base = mcasp->base; + + mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl); + mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir); return 0; } -- cgit v1.2.3-70-g09d2 From 8f113b77b511c9e26706d4eb077af0ba30893ee4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:30 +0200 Subject: ASoC: davinci-mcasp: Be consistent with the use of base in davinci_mcasp_set_dai_fmt Replace mcasp->base use with plain base in the davinci_mcasp_set_dai_fmt() function since it has been already used by the remaining part of the function. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index bd85c98bf5a..1341f327df8 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -243,17 +243,17 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_AC97: - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); break; default: /* configure a full-word SYNC pulse (LRCLK) */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); /* make 1st data bit occur one ACLK cycle after the frame sync */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); break; } -- cgit v1.2.3-70-g09d2 From 487dce8823cdcb70e645e5312a0d4f7081e1ad13 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:31 +0200 Subject: ASoC: davinci-mcasp: Simplify FIFO configuration code The FIFO registers base address is different in dm646x compared to newer SoCs with McASP IP. Instead of using two paths (switch/case) to handle the difference we can simply pick the correct base address beforehand and use offsets to address the register we need to configure. With this change the indentation depth can be reduced as well. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 101 +++++++++++--------------------------- sound/soc/davinci/davinci-mcasp.h | 16 +++--- 2 files changed, 38 insertions(+), 79 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1341f327df8..72ea45893ab 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -38,6 +38,7 @@ struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; + u32 fifo_base; struct device *dev; /* McASP specific data */ @@ -153,38 +154,20 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) { + u32 reg; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* enable FIFO */ - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); - mcasp_set_bits(mcasp->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); - break; - default: - mcasp_clr_bits(mcasp->base + - DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(mcasp->base + - DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - } + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE); } mcasp_start_tx(mcasp); } else { if (mcasp->rxnumevt) { /* enable FIFO */ - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); - mcasp_set_bits(mcasp->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); - break; - default: - mcasp_clr_bits(mcasp->base + - DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(mcasp->base + - DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); - } + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE); } mcasp_start_rx(mcasp); } @@ -204,31 +187,18 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) { + u32 reg; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* disable FIFO */ - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL, - FIFO_ENABLE); - break; - default: - mcasp_clr_bits(mcasp->base + - DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - } + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); } mcasp_stop_tx(mcasp); } else { if (mcasp->rxnumevt) { /* disable FIFO */ - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL, - FIFO_ENABLE); - break; - - default: - mcasp_clr_bits(mcasp->base + - DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); - } + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); } mcasp_stop_rx(mcasp); } @@ -438,6 +408,7 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, u8 ser; u8 slots = mcasp->tdm_slots; u8 max_active_serializers = (channels + slots - 1) / slots; + u32 reg; /* Default configuration */ mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -488,37 +459,20 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, if (mcasp->txnumevt * tx_ser > 64) mcasp->txnumevt = 1; - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL, tx_ser, - NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL, - ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); - break; - default: - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, - tx_ser, NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL, - ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); - } + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_mod_bits(mcasp->base + reg, tx_ser, NUMDMA_MASK); + mcasp_mod_bits(mcasp->base + reg, + ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); } if (mcasp->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { if (mcasp->rxnumevt * rx_ser > 64) mcasp->rxnumevt = 1; - switch (mcasp->version) { - case MCASP_VERSION_3: - mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL, rx_ser, - NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL, - ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); - break; - default: - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, - rx_ser, NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL, - ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); - } + + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_mod_bits(mcasp->base + reg, rx_ser, NUMDMA_MASK); + mcasp_mod_bits(mcasp->base + reg, + ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } return 0; @@ -974,6 +928,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->version = pdata->version; mcasp->txnumevt = pdata->txnumevt; mcasp->rxnumevt = pdata->rxnumevt; + if (mcasp->version < MCASP_VERSION_3) + mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; + else + mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; + mcasp->dev = &pdev->dev; dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 80e5a184668..8fed757d608 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -90,14 +90,14 @@ #define DAVINCI_MCASP_RXBUF_REG 0x280 /* McASP FIFO Registers */ -#define DAVINCI_MCASP_WFIFOCTL (0x1010) -#define DAVINCI_MCASP_WFIFOSTS (0x1014) -#define DAVINCI_MCASP_RFIFOCTL (0x1018) -#define DAVINCI_MCASP_RFIFOSTS (0x101C) -#define MCASP_VER3_WFIFOCTL (0x1000) -#define MCASP_VER3_WFIFOSTS (0x1004) -#define MCASP_VER3_RFIFOCTL (0x1008) -#define MCASP_VER3_RFIFOSTS (0x100C) +#define DAVINCI_MCASP_V2_AFIFO_BASE (0x1010) +#define DAVINCI_MCASP_V3_AFIFO_BASE (0x1000) + +/* FIFO register offsets from AFIFO base */ +#define MCASP_WFIFOCTL_OFFSET (0x0) +#define MCASP_WFIFOSTS_OFFSET (0x4) +#define MCASP_RFIFOCTL_OFFSET (0x8) +#define MCASP_RFIFOSTS_OFFSET (0xc) /* * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management -- cgit v1.2.3-70-g09d2 From cbc7956c81eea644c0d99aee43f1632897703300 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:32 +0200 Subject: ASoC: davinci-mcasp: Data source (bus) selection support The audio data to/from McASP can be sent/received via two method: Via the data port (preferred) or via the configuration bus. Currently the driver assumes that all data communication will be done via the data port. This patch adds support for selecting the configuration port as data interface. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 38 ++++++++++++++++++++++++++++---------- 1 file changed, 28 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 72ea45893ab..35a6292889a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -53,6 +53,8 @@ struct davinci_mcasp { u8 txnumevt; u8 rxnumevt; + bool dat_port; + #ifdef CONFIG_PM_SLEEP struct { u32 txfmtctl; @@ -482,6 +484,7 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) { int i, active_slots; u32 mask = 0; + u32 busel = 0; active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots; for (i = 0; i < active_slots; i++) @@ -489,11 +492,15 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + if (!mcasp->dat_port) + busel = TXSEL; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { /* bit stream is MSB first with no delay */ /* DSP_B mode */ mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, mask); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, + busel | TXORD); if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, @@ -504,7 +511,8 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) } else { /* bit stream is MSB first with no delay */ /* DSP_B mode */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, + busel | RXORD); mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXTDM_REG, mask); if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) @@ -928,23 +936,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->version = pdata->version; mcasp->txnumevt = pdata->txnumevt; mcasp->rxnumevt = pdata->rxnumevt; - if (mcasp->version < MCASP_VERSION_3) - mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; - else - mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; mcasp->dev = &pdev->dev; dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); - if (!dat) - dat = mem; + if (dat) + mcasp->dat_port = true; dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_playback; - dma_data->dma_addr = dat->start + pdata->tx_dma_offset; + if (dat) + dma_data->dma_addr = dat->start; + else + dma_data->dma_addr = mem->start + pdata->tx_dma_offset; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) @@ -957,7 +964,18 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_capture; - dma_data->dma_addr = dat->start + pdata->rx_dma_offset; + if (dat) + dma_data->dma_addr = dat->start; + else + dma_data->dma_addr = mem->start + pdata->rx_dma_offset; + + if (mcasp->version < MCASP_VERSION_3) { + mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; + /* dma_data->dma_addr is pointing to the data port address */ + mcasp->dat_port = true; + } else { + mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; + } res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (res) -- cgit v1.2.3-70-g09d2 From 4dcb5a0bffaa7dc51e738c4f651a31993b1eb08b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:33 +0200 Subject: ASoC: davinci-mcasp: Fix synchronous master receive mode In synchronous mode both transmit and receive sections are using the TX clocks. In setup like this the TX clocks need to be enabled when capture is running. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 48 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 47 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 35a6292889a..93f2e294d64 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -48,6 +48,7 @@ struct davinci_mcasp { u8 *serial_dir; u8 version; u16 bclk_lrclk_ratio; + int streams; /* McASP FIFO related */ u8 txnumevt; @@ -110,10 +111,31 @@ static void mcasp_set_ctl_reg(void __iomem *regs, u32 val) printk(KERN_ERR "GBLCTL write error\n"); } +static bool mcasp_is_synchronous(struct davinci_mcasp *mcasp) +{ + u32 rxfmctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG); + u32 aclkxctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG); + + return !(aclkxctl & TX_ASYNC) && rxfmctl & AFSRE; +} + static void mcasp_start_rx(struct davinci_mcasp *mcasp) { mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + + /* + * When ASYNC == 0 the transmit and receive sections operate + * synchronously from the transmit clock and frame sync. We need to make + * sure that the TX signlas are enabled when starting reception. + */ + if (mcasp_is_synchronous(mcasp)) { + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, + TXHCLKRST); + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, + TXCLKRST); + } + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0); @@ -123,6 +145,10 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + + if (mcasp_is_synchronous(mcasp)) + mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, + TXFSRST); } static void mcasp_start_tx(struct davinci_mcasp *mcasp) @@ -158,6 +184,8 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) { u32 reg; + mcasp->streams++; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* enable FIFO */ reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; @@ -177,13 +205,29 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) static void mcasp_stop_rx(struct davinci_mcasp *mcasp) { + /* + * In synchronous mode stop the TX clocks if no other stream is + * running + */ + if (mcasp_is_synchronous(mcasp) && !mcasp->streams) + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, 0); mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); } static void mcasp_stop_tx(struct davinci_mcasp *mcasp) { - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + u32 val = 0; + + /* + * In synchronous mode keep TX clocks running if the capture stream is + * still running. + */ + if (mcasp_is_synchronous(mcasp) && mcasp->streams) + val = TXHCLKRST | TXCLKRST | TXFSRST; + + mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, val); mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); } @@ -191,6 +235,8 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) { u32 reg; + mcasp->streams--; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* disable FIFO */ reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; -- cgit v1.2.3-70-g09d2 From 453c499028bf2ecf3b31ccb7c3657fe1b0b28943 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:34 +0200 Subject: ASoC: davinci-mcasp: Support for McASP version found in DRA7xx The IP in DRA7xx is similar to the IP found in TI81xxAM3xxx/AM4xxx type of SoCs but it is is integrated with sDMA instead of eDMA. The suitable pcm driver for DRA7xx is the omap-pcm driver which is using dmaengine. In the driver we can configure both dma related structures used for eDMA and sDMA. The only thing we need to make sure that we set the correct dma_data at startup with snd_soc_dai_set_dma_data() Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 1 + include/linux/platform_data/davinci_asp.h | 1 + sound/soc/davinci/davinci-mcasp.c | 52 +++++++++++++++++++--- 3 files changed, 47 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 1eed972d471..990fa71ce80 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -5,6 +5,7 @@ Required properties: "ti,dm646x-mcasp-audio" : for DM646x platforms "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) + "ti,dra7-mcasp-audio" : for DRA7xx platforms - reg : Should contain reg specifiers for the entries in the reg-names property. - reg-names : Should contain: diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 689a856b86f..5245992b036 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -92,6 +92,7 @@ enum { MCASP_VERSION_1 = 0, /* DM646x */ MCASP_VERSION_2, /* DA8xx/OMAPL1x */ MCASP_VERSION_3, /* TI81xx/AM33xx */ + MCASP_VERSION_4, /* DRA7xxx */ }; enum mcbsp_clk_input_pin { diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 93f2e294d64..fc8c13d2f31 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -31,12 +31,14 @@ #include #include #include +#include #include "davinci-pcm.h" #include "davinci-mcasp.h" struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; u32 fifo_base; struct device *dev; @@ -458,7 +460,9 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, u8 max_active_serializers = (channels + slots - 1) / slots; u32 reg; /* Default configuration */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + if (mcasp->version != MCASP_VERSION_4) + mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, + MCASP_SOFT); /* All PINS as McASP */ mcasp_set_reg(mcasp->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); @@ -605,6 +609,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); struct davinci_pcm_dma_params *dma_params = &mcasp->dma_params[substream->stream]; + struct snd_dmaengine_dai_dma_data *dma_data = + &mcasp->dma_data[substream->stream]; int word_length; u8 fifo_level; u8 slots = mcasp->tdm_slots; @@ -666,6 +672,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_data->maxburst = fifo_level; + davinci_config_channel_size(mcasp, word_length); return 0; @@ -711,7 +719,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - snd_soc_dai_set_dma_data(dai, substream, mcasp->dma_params); + if (mcasp->version == MCASP_VERSION_4) + snd_soc_dai_set_dma_data(dai, substream, + &mcasp->dma_data[substream->stream]); + else + snd_soc_dai_set_dma_data(dai, substream, mcasp->dma_params); + return 0; } @@ -794,6 +807,13 @@ static struct snd_platform_data omap2_mcasp_pdata = { .version = MCASP_VERSION_3, }; +static struct snd_platform_data dra7_mcasp_pdata = { + .tx_dma_offset = 0x200, + .rx_dma_offset = 0x284, + .asp_chan_q = EVENTQ_0, + .version = MCASP_VERSION_4, +}; + static const struct of_device_id mcasp_dt_ids[] = { { .compatible = "ti,dm646x-mcasp-audio", @@ -807,6 +827,10 @@ static const struct of_device_id mcasp_dt_ids[] = { .compatible = "ti,am33xx-mcasp-audio", .data = &omap2_mcasp_pdata, }, + { + .compatible = "ti,dra7-mcasp-audio", + .data = &dra7_mcasp_pdata, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, mcasp_dt_ids); @@ -999,6 +1023,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->dma_addr = mem->start + pdata->tx_dma_offset; + /* Unconditional dmaengine stuff */ + mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_data->dma_addr; + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) dma_data->channel = res->start; @@ -1015,6 +1042,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->dma_addr = mem->start + pdata->rx_dma_offset; + /* Unconditional dmaengine stuff */ + mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_data->dma_addr; + if (mcasp->version < MCASP_VERSION_3) { mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; /* dma_data->dma_addr is pointing to the data port address */ @@ -1029,6 +1059,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_data->channel = pdata->rx_dma_channel; + /* Unconditional dmaengine stuff */ + mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = "tx"; + mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = "rx"; + dev_set_drvdata(&pdev->dev, mcasp); ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1); @@ -1036,10 +1070,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (ret != 0) goto err_release_clk; - ret = davinci_soc_platform_register(&pdev->dev); - if (ret) { - dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - goto err_unregister_component; + if (mcasp->version != MCASP_VERSION_4) { + ret = davinci_soc_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "register PCM failed: %d\n", ret); + goto err_unregister_component; + } } return 0; @@ -1054,9 +1090,11 @@ err_release_clk: static int davinci_mcasp_remove(struct platform_device *pdev) { + struct davinci_mcasp *mcasp = dev_get_drvdata(&pdev->dev); snd_soc_unregister_component(&pdev->dev); - davinci_soc_platform_unregister(&pdev->dev); + if (mcasp->version != MCASP_VERSION_4) + davinci_soc_platform_unregister(&pdev->dev); pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); -- cgit v1.2.3-70-g09d2 From f68205a7f8c0b1fd02cec6116bbb66bb4fd7bc51 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:36 +0200 Subject: ASoC: davinci-mcasp: Change IO functions parameter list Instead of passing __iomem address (mcasp->base + register_offset) pass the main mcasp structure and only access the mcasp->base in the low level IO functions. In most cases this helps with code readability and it will make it easier to switch over to regmap in the future. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 336 ++++++++++++++++++-------------------- 1 file changed, 161 insertions(+), 175 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index fc8c13d2f31..19c66625af4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -71,60 +71,67 @@ struct davinci_mcasp { #endif }; -static inline void mcasp_set_bits(void __iomem *reg, u32 val) +static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, + u32 val) { + void __iomem *reg = mcasp->base + offset; __raw_writel(__raw_readl(reg) | val, reg); } -static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +static inline void mcasp_clr_bits(struct davinci_mcasp *mcasp, u32 offset, + u32 val) { + void __iomem *reg = mcasp->base + offset; __raw_writel((__raw_readl(reg) & ~(val)), reg); } -static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +static inline void mcasp_mod_bits(struct davinci_mcasp *mcasp, u32 offset, + u32 val, u32 mask) { + void __iomem *reg = mcasp->base + offset; __raw_writel((__raw_readl(reg) & ~mask) | val, reg); } -static inline void mcasp_set_reg(void __iomem *reg, u32 val) +static inline void mcasp_set_reg(struct davinci_mcasp *mcasp, u32 offset, + u32 val) { - __raw_writel(val, reg); + __raw_writel(val, mcasp->base + offset); } -static inline u32 mcasp_get_reg(void __iomem *reg) +static inline u32 mcasp_get_reg(struct davinci_mcasp *mcasp, u32 offset) { - return (unsigned int)__raw_readl(reg); + return (u32)__raw_readl(mcasp->base + offset); } -static void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +static void mcasp_set_ctl_reg(struct davinci_mcasp *mcasp, u32 ctl_reg, u32 val) { int i = 0; - mcasp_set_bits(regs, val); + mcasp_set_bits(mcasp, ctl_reg, val); /* programming GBLCTL needs to read back from GBLCTL and verfiy */ /* loop count is to avoid the lock-up */ for (i = 0; i < 1000; i++) { - if ((mcasp_get_reg(regs) & val) == val) + if ((mcasp_get_reg(mcasp, ctl_reg) & val) == val) break; } - if (i == 1000 && ((mcasp_get_reg(regs) & val) != val)) + if (i == 1000 && ((mcasp_get_reg(mcasp, ctl_reg) & val) != val)) printk(KERN_ERR "GBLCTL write error\n"); } static bool mcasp_is_synchronous(struct davinci_mcasp *mcasp) { - u32 rxfmctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG); - u32 aclkxctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG); + u32 rxfmctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG); + u32 aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG); return !(aclkxctl & TX_ASYNC) && rxfmctl & AFSRE; } static void mcasp_start_rx(struct davinci_mcasp *mcasp) { - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); /* * When ASYNC == 0 the transmit and receive sections operate @@ -132,25 +139,22 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) * sure that the TX signlas are enabled when starting reception. */ if (mcasp_is_synchronous(mcasp)) { - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, - TXHCLKRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, - TXCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); } - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXBUF_REG, 0); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXBUF_REG, 0); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); if (mcasp_is_synchronous(mcasp)) - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, - TXFSRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); } static void mcasp_start_tx(struct davinci_mcasp *mcasp) @@ -158,14 +162,14 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) u8 offset = 0, i; u32 cnt; - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); - mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0); for (i = 0; i < mcasp->num_serializer; i++) { if (mcasp->serial_dir[i] == TX_MODE) { offset = i; @@ -175,11 +179,11 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) /* wait for TX ready */ cnt = 0; - while (!(mcasp_get_reg(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & + while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(offset)) & TXSTATE) && (cnt < 100000)) cnt++; - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0); } static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) @@ -191,15 +195,15 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* enable FIFO */ reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); - mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); + mcasp_set_bits(mcasp, reg, FIFO_ENABLE); } mcasp_start_tx(mcasp); } else { if (mcasp->rxnumevt) { /* enable FIFO */ reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); - mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); + mcasp_set_bits(mcasp, reg, FIFO_ENABLE); } mcasp_start_rx(mcasp); } @@ -212,10 +216,10 @@ static void mcasp_stop_rx(struct davinci_mcasp *mcasp) * running */ if (mcasp_is_synchronous(mcasp) && !mcasp->streams) - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, 0); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, 0); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); } static void mcasp_stop_tx(struct davinci_mcasp *mcasp) @@ -229,8 +233,8 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) if (mcasp_is_synchronous(mcasp) && mcasp->streams) val = TXHCLKRST | TXCLKRST | TXFSRST; - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, val); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, val); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); } static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) @@ -242,13 +246,13 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (mcasp->txnumevt) { /* disable FIFO */ reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); } mcasp_stop_tx(mcasp); } else { if (mcasp->rxnumevt) { /* disable FIFO */ reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE); + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); } mcasp_stop_rx(mcasp); } @@ -258,62 +262,57 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); - void __iomem *base = mcasp->base; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_AC97: - mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); break; default: /* configure a full-word SYNC pulse (LRCLK) */ - mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); /* make 1st data bit occur one ACLK cycle after the frame sync */ - mcasp_set_bits(base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); break; } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: /* codec is clock and frame slave */ - mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); - mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); - mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, - ACLKX | ACLKR); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, - AFSX | AFSR); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); - mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, - ACLKX | ACLKR); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, - AFSX | AFSR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); - mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); - mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, - ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, + ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); break; default: @@ -322,35 +321,35 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_IB_NF: - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; case SND_SOC_DAIFMT_NB_IF: - mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); - mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; case SND_SOC_DAIFMT_IB_IF: - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); - mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); - mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; case SND_SOC_DAIFMT_NB_NF: - mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; default: @@ -366,16 +365,16 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div switch (div_id) { case 0: /* MCLK divider */ - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, + mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(div - 1), AHCLKXDIV_MASK); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, + mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRDIV(div - 1), AHCLKRDIV_MASK); break; case 1: /* BCLK divider */ - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, + mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXDIV(div - 1), ACLKXDIV_MASK); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKRCTL_REG, + mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRDIV(div - 1), ACLKRDIV_MASK); break; @@ -396,13 +395,13 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); if (dir == SND_SOC_CLOCK_OUT) { - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); } else { - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); } return 0; @@ -432,19 +431,18 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, fmt = (word_length >> 1) - 1; if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, - TXROT(tx_rotate), TXROT(7)); - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, - RXROT(rx_rotate), RXROT(7)); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXMASK_REG, - mask); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt), + RXSSZ(0x0F)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), + TXSSZ(0x0F)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(tx_rotate), + TXROT(7)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXROT(rx_rotate), + RXROT(7)); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXMASK_REG, mask); } - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, mask); return 0; } @@ -461,38 +459,33 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, u32 reg; /* Default configuration */ if (mcasp->version != MCASP_VERSION_4) - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, - MCASP_SOFT); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); /* All PINS as McASP */ - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + mcasp_set_reg(mcasp, DAVINCI_MCASP_PFUNC_REG, 0x00000000); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG, - TXDATADMADIS); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); } else { - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_REVTCTL_REG, - RXDATADMADIS); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_REVTCTL_REG, RXDATADMADIS); } for (i = 0; i < mcasp->num_serializer; i++) { - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i), - mcasp->serial_dir[i]); + mcasp_set_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + mcasp->serial_dir[i]); if (mcasp->serial_dir[i] == TX_MODE && tx_ser < max_active_serializers) { - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, - AXR(i)); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; } else if (mcasp->serial_dir[i] == RX_MODE && rx_ser < max_active_serializers) { - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, - AXR(i)); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; } else { - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i), - SRMOD_INACTIVE, SRMOD_MASK); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -512,9 +505,9 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, mcasp->txnumevt = 1; reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_mod_bits(mcasp->base + reg, tx_ser, NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + reg, - ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp, reg, tx_ser, NUMDMA_MASK); + mcasp_mod_bits(mcasp, reg, ((mcasp->txnumevt * tx_ser) << 8), + NUMEVT_MASK); } if (mcasp->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { @@ -522,9 +515,9 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, mcasp->rxnumevt = 1; reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_mod_bits(mcasp->base + reg, rx_ser, NUMDMA_MASK); - mcasp_mod_bits(mcasp->base + reg, - ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + mcasp_mod_bits(mcasp, reg, rx_ser, NUMDMA_MASK); + mcasp_mod_bits(mcasp, reg, ((mcasp->rxnumevt * rx_ser) << 8), + NUMEVT_MASK); } return 0; @@ -540,7 +533,7 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) for (i = 0; i < active_slots; i++) mask |= (1 << i); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); if (!mcasp->dat_port) busel = TXSEL; @@ -548,26 +541,24 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { /* bit stream is MSB first with no delay */ /* DSP_B mode */ - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, mask); - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, - busel | TXORD); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, - FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); else printk(KERN_ERR "playback tdm slot %d not supported\n", mcasp->tdm_slots); } else { /* bit stream is MSB first with no delay */ /* DSP_B mode */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, - busel | RXORD); - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, - FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); else printk(KERN_ERR "capture tdm slot %d not supported\n", mcasp->tdm_slots); @@ -579,27 +570,24 @@ static void davinci_hw_dit_param(struct davinci_mcasp *mcasp) { /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, - TXROT(6) | TXSSZ(15)); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, - AFSXE | FSXMOD(0x180)); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE | FSXMOD(0x180)); /* Set the TX tdm : for all the slots */ - mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); /* Set the TX clock controls : div = 1 and internal */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, - ACLKXE | TX_ASYNC); + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE | TX_ASYNC); - mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); /* Only 44100 and 48000 are valid, both have the same setting */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); /* Enable the DIT */ - mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); } static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, @@ -1106,15 +1094,14 @@ static int davinci_mcasp_remove(struct platform_device *pdev) static int davinci_mcasp_suspend(struct device *dev) { struct davinci_mcasp *mcasp = dev_get_drvdata(dev); - void __iomem *base = mcasp->base; - mcasp->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG); - mcasp->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG); - mcasp->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG); - mcasp->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG); - mcasp->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG); - mcasp->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG); - mcasp->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG); + mcasp->context.txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG); + mcasp->context.rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG); + mcasp->context.txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG); + mcasp->context.rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG); + mcasp->context.aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG); + mcasp->context.aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG); + mcasp->context.pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG); return 0; } @@ -1122,15 +1109,14 @@ static int davinci_mcasp_suspend(struct device *dev) static int davinci_mcasp_resume(struct device *dev) { struct davinci_mcasp *mcasp = dev_get_drvdata(dev); - void __iomem *base = mcasp->base; - - mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl); - mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl); - mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt); - mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt); - mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl); - mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl); - mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir); + + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt); + mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir); return 0; } -- cgit v1.2.3-70-g09d2 From b14899da9ddeb8501db13fd08d0d1a8af61529c5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:37 +0200 Subject: ASoC: davinci-mcasp: Correct am33xx snd_platform_data name An earlier patch overlooked this when the compatible has been changed from omap2 -> am33x. Rename omap2_mcasp_pdata to am33xx_mcasp_pdata. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 19c66625af4..8ec87954848 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -788,7 +788,7 @@ static struct snd_platform_data da830_mcasp_pdata = { .version = MCASP_VERSION_2, }; -static struct snd_platform_data omap2_mcasp_pdata = { +static struct snd_platform_data am33xx_mcasp_pdata = { .tx_dma_offset = 0, .rx_dma_offset = 0, .asp_chan_q = EVENTQ_0, @@ -813,7 +813,7 @@ static const struct of_device_id mcasp_dt_ids[] = { }, { .compatible = "ti,am33xx-mcasp-audio", - .data = &omap2_mcasp_pdata, + .data = &am33xx_mcasp_pdata, }, { .compatible = "ti,dra7-mcasp-audio", -- cgit v1.2.3-70-g09d2 From ae726e93946403b14f8cad20d5cbd22d015c9106 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Nov 2013 11:35:35 +0200 Subject: ASoC: davinci-mcasp: Support for fck reparenting Optional DT property to specify the desired parent clock for the McASP fck clock. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 3 +- sound/soc/davinci/davinci-mcasp.c | 44 ++++++++++++++++++++++ 2 files changed, 46 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 990fa71ce80..569b26c4a81 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -37,7 +37,8 @@ Optional properties: - pinctrl-0: Should specify pin control group used for this controller. - pinctrl-names: Should contain only one value - "default", for more details please refer to pinctrl-bindings.txt - +- fck_parent : Should contain a valid clock name which will be used as parent + for the McASP fck Example: diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 8ec87954848..b7858bfa029 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -823,6 +824,46 @@ static const struct of_device_id mcasp_dt_ids[] = { }; MODULE_DEVICE_TABLE(of, mcasp_dt_ids); +static int mcasp_reparent_fck(struct platform_device *pdev) +{ + struct device_node *node = pdev->dev.of_node; + struct clk *gfclk, *parent_clk; + const char *parent_name; + int ret; + + if (!node) + return 0; + + parent_name = of_get_property(node, "fck_parent", NULL); + if (!parent_name) + return 0; + + gfclk = clk_get(&pdev->dev, "fck"); + if (IS_ERR(gfclk)) { + dev_err(&pdev->dev, "failed to get fck\n"); + return PTR_ERR(gfclk); + } + + parent_clk = clk_get(NULL, parent_name); + if (IS_ERR(parent_clk)) { + dev_err(&pdev->dev, "failed to get parent clock\n"); + ret = PTR_ERR(parent_clk); + goto err1; + } + + ret = clk_set_parent(gfclk, parent_clk); + if (ret) { + dev_err(&pdev->dev, "failed to reparent fck\n"); + goto err2; + } + +err2: + clk_put(parent_clk); +err1: + clk_put(gfclk); + return ret; +} + static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( struct platform_device *pdev) { @@ -1052,6 +1093,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = "rx"; dev_set_drvdata(&pdev->dev, mcasp); + + mcasp_reparent_fck(pdev); + ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1); -- cgit v1.2.3-70-g09d2 From a073278228836d7d18fdd6c40b619919c0befb64 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Dec 2013 20:46:59 -0800 Subject: ASoC: fsi: remove original filter from fsi_dma_probe() Remove original filter from fsi_dma_probe(), and use SH-DMA suitable filter. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b33ca7cd085..6101055aae1 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -232,9 +232,9 @@ struct fsi_stream { * these are for DMAEngine */ struct dma_chan *chan; - struct sh_dmae_slave slave; /* see fsi_handler_init() */ struct work_struct work; dma_addr_t dma; + int dma_id; int loop_cnt; int additional_pos; }; @@ -1410,15 +1410,6 @@ static void fsi_dma_do_work(struct work_struct *work) } } -static bool fsi_dma_filter(struct dma_chan *chan, void *param) -{ - struct sh_dmae_slave *slave = param; - - chan->private = slave; - - return true; -} - static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) { schedule_work(&io->work); @@ -1446,15 +1437,34 @@ static int fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) { dma_cap_mask_t mask; + int is_play = fsi_stream_is_play(fsi, io); dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave); + io->chan = dma_request_slave_channel_compat(mask, + shdma_chan_filter, (void *)io->dma_id, + dev, is_play ? "tx" : "rx"); + if (io->chan) { + struct dma_slave_config cfg; + int ret; + + cfg.slave_id = io->dma_id; + cfg.dst_addr = 0; /* use default addr */ + cfg.src_addr = 0; /* use default addr */ + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + + ret = dmaengine_slave_config(io->chan, &cfg); + if (ret < 0) { + dma_release_channel(io->chan); + io->chan = NULL; + } + } + if (!io->chan) { /* switch to PIO handler */ - if (fsi_stream_is_play(fsi, io)) + if (is_play) fsi->playback.handler = &fsi_pio_push_handler; else fsi->capture.handler = &fsi_pio_pop_handler; @@ -1960,7 +1970,7 @@ static void fsi_handler_init(struct fsi_priv *fsi, fsi->capture.priv = fsi; if (info->tx_id) { - fsi->playback.slave.shdma_slave.slave_id = info->tx_id; + fsi->playback.dma_id = info->tx_id; fsi->playback.handler = &fsi_dma_push_handler; } } -- cgit v1.2.3-70-g09d2 From 5eda87b890f867b098e5566b5543642851e8b9c3 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Dec 2013 11:11:02 -0700 Subject: ASoC: dmaengine: support deferred probe for DMA channels Enhance dmaengine_pcm_request_chan_of() to support deferred probe for DMA channels, by using the new dma_request_slave_channel_or_err() API. This prevents snd_dmaengine_pcm_register() from succeeding without acquiring DMA channels due to the relevant DMA controller not yet being registered. Signed-off-by: Stephen Warren Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5b70c556fba..585eaa69e8c 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -287,16 +287,17 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_CAPTURE] = "rx", }; -static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, +static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, struct device *dev, const struct snd_dmaengine_pcm_config *config) { unsigned int i; const char *name; + struct dma_chan *chan; if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || !dev->of_node) - return; + return 0; if (config->dma_dev) { /* @@ -318,13 +319,22 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = dmaengine_pcm_dma_channel_names[i]; if (config->chan_names[i]) name = config->chan_names[i]; - pcm->chan[i] = dma_request_slave_channel(dev, name); + chan = dma_request_slave_channel_reason(dev, name); + if (IS_ERR(chan)) { + if (PTR_ERR(pcm->chan[i]) == -EPROBE_DEFER) + return -EPROBE_DEFER; + pcm->chan[i] = NULL; + } else { + pcm->chan[i] = chan; + } if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) break; } if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) pcm->chan[1] = pcm->chan[0]; + + return 0; } static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) @@ -360,7 +370,9 @@ int snd_dmaengine_pcm_register(struct device *dev, pcm->config = config; pcm->flags = flags; - dmaengine_pcm_request_chan_of(pcm, dev, config); + ret = dmaengine_pcm_request_chan_of(pcm, dev, config); + if (ret) + goto err_free_dma; if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) ret = snd_soc_add_platform(dev, &pcm->platform, -- cgit v1.2.3-70-g09d2 From e9036c2a60f368b8ddc349fbbcde918ed32ab597 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 11 Dec 2013 11:20:50 -0700 Subject: ASoC: dmaengine: fix deferred probe detection Check the return value of dma_request_slave_channel_reason() to see if deferred probe happens, not the variable the return value will be assigned to later. Reported-by: kbuild test robot Fixes: 5eda87b890f8 ("ASoC: dmaengine: support deferred probe for DMA channels") Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 585eaa69e8c..7483922f6ee 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -321,7 +321,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = config->chan_names[i]; chan = dma_request_slave_channel_reason(dev, name); if (IS_ERR(chan)) { - if (PTR_ERR(pcm->chan[i]) == -EPROBE_DEFER) + if (PTR_ERR(chan) == -EPROBE_DEFER) return -EPROBE_DEFER; pcm->chan[i] = NULL; } else { -- cgit v1.2.3-70-g09d2 From 3688569e8173e84cd95d98f158245e17bca4f593 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 19 Oct 2013 15:23:15 +0100 Subject: ASoC: samsung: Provide helper for DMA init In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown --- sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 7 +++++-- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 42 insertions(+), 45 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 350ba23a989..4a88e36c82e 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b494d..ee23194f7ab 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315a3ea..fb09a1c5f75 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4f165..eab0050d457 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { @@ -963,7 +966,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256fc4b2..6a5e4bf6ac9 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- cgit v1.2.3-70-g09d2 From d37bdf736d9b7a198d35aaaf611e96ddc2e00ddf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Dec 2013 14:14:52 +0000 Subject: ASoC: samsung: Use ASoC dmaengine code where possible Since all Exynos platforms have been converted to dmaengine and many of the older platforms are in the process of conversion they do not need to use the legacy s3c-dma APIs for DMA but can instead use the standard ASoC dmaengine helpers. This both allows them to benefit from improvements implemented in the generic code and supports multiplatform. This patch includes some fixes from Padma for Exynos SoCs, her testing was on a slightly earlier version of the patch due to unrelated breakage preventing testing. Signed-off-by: Mark Brown Tested By: Padmavathi Venna --- sound/soc/samsung/Kconfig | 13 +++++-- sound/soc/samsung/Makefile | 6 ++-- sound/soc/samsung/dma.h | 3 ++ sound/soc/samsung/dmaengine.c | 84 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/i2s.c | 2 ++ 5 files changed, 104 insertions(+), 4 deletions(-) create mode 100644 sound/soc/samsung/dmaengine.c (limited to 'sound/soc') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 37459dfd168..27930fc432d 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,13 +1,22 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C64XX_DMA if ARCH_S3C64XX - select S3C24XX_DMA if ARCH_S3C24XX + select S3C2410_DMA if ARCH_S3C24XX + select S3C64XX_PL080 if ARCH_S3C64XX + select SND_S3C_DMA if !ARCH_S3C24XX + select SND_S3C_DMA_LEGACY if ARCH_S3C24XX + select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to select the audio interfaces to support below. +config SND_S3C_DMA + tristate + +config SND_S3C_DMA_LEGACY + tristate + config SND_S3C24XX_I2S tristate select S3C2410_DMA diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 709f6059ad6..86715d8efee 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,5 +1,6 @@ # S3c24XX Platform Support -snd-soc-s3c24xx-objs := dma.o +snd-soc-s3c-dma-objs := dmaengine.o +snd-soc-s3c-dma-legacy-objs := dma.o snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o @@ -9,7 +10,8 @@ snd-soc-samsung-spdif-objs := spdif.o snd-soc-pcm-objs := pcm.o snd-soc-i2s-objs := i2s.o -obj-$(CONFIG_SND_SOC_SAMSUNG) += snd-soc-s3c24xx.o +obj-$(CONFIG_SND_S3C_DMA) += snd-soc-s3c-dma.o +obj-$(CONFIG_SND_S3C_DMA_LEGACY) += snd-soc-s3c-dma-legacy.o obj-$(CONFIG_SND_S3C24XX_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_SAMSUNG_AC97) += snd-soc-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index fb09a1c5f75..225e5378014 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -12,6 +12,8 @@ #ifndef _S3C_AUDIO_H #define _S3C_AUDIO_H +#include + struct s3c_dma_params { struct s3c2410_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ @@ -20,6 +22,7 @@ struct s3c_dma_params { unsigned ch; struct samsung_dma_ops *ops; char *ch_name; + struct snd_dmaengine_dai_dma_data dma_data; }; void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c new file mode 100644 index 00000000000..3be479d51b9 --- /dev/null +++ b/sound/soc/samsung/dmaengine.c @@ -0,0 +1,84 @@ +/* + * dmaengine.c - Samsung dmaengine wrapper + * + * Author: Mark Brown + * Copyright 2013 Linaro + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#include +#include + +#include +#include +#include +#include +#include +#include + +#include "dma.h" + +#ifdef CONFIG_ARCH_S3C64XX +#define filter_fn pl08x_filter_id +#else +#define filter_fn NULL +#endif + +static const struct snd_dmaengine_pcm_config samsung_dmaengine_pcm_config = { + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .compat_filter_fn = filter_fn, +}; + +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + struct snd_dmaengine_dai_dma_data *playback_data = NULL; + struct snd_dmaengine_dai_dma_data *capture_data = NULL; + + if (playback) { + playback_data = &playback->dma_data; + playback_data->filter_data = (void *)playback->channel; + playback_data->chan_name = playback->ch_name; + playback_data->addr = playback->dma_addr; + playback_data->addr_width = playback->dma_size; + } + if (capture) { + capture_data = &capture->dma_data; + capture_data->filter_data = (void *)capture->channel; + capture_data->chan_name = capture->ch_name; + capture_data->addr = capture->dma_addr; + capture_data->addr_width = capture->dma_size; + } + + snd_soc_dai_init_dma_data(dai, playback_data, capture_data); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + +int samsung_asoc_dma_platform_register(struct device *dev) +{ + return snd_dmaengine_pcm_register(dev, &samsung_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME | + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_COMPAT); +} +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); + +void samsung_asoc_dma_platform_unregister(struct device *dev) +{ + return snd_dmaengine_pcm_unregister(dev); +} +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_unregister); + +MODULE_AUTHOR("Mark Brown "); +MODULE_DESCRIPTION("Samsung dmaengine ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index eab0050d457..92f64363427 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -702,6 +702,8 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, } writel(mod, i2s->addr + I2SMOD); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + i2s->frmclk = params_rate(params); return 0; -- cgit v1.2.3-70-g09d2 From 753834cb593da03b4efc468bb8cb76dbc0743b31 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Sat, 14 Dec 2013 13:29:11 +0800 Subject: ASoC: tegra20-ac97: add missing clk_disable_unprepare() on error path Add the missing clk_disable_unprepare() before return from tegra20_ac97_platform_probe() in the error handling case. Signed-off-by: Wei Yongjun Reviewed-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index ae27bcd586d..088518d7694 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -404,7 +404,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); if (ret) { dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); - goto err_asoc_utils_fini; + goto err_clk_disable_unprepare; } ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, @@ -412,7 +412,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; - goto err_asoc_utils_fini; + goto err_clk_disable_unprepare; } ret = tegra_pcm_platform_register(&pdev->dev); @@ -428,6 +428,8 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) err_unregister_component: snd_soc_unregister_component(&pdev->dev); +err_clk_disable_unprepare: + clk_disable_unprepare(ac97->clk_ac97); err_asoc_utils_fini: tegra_asoc_utils_fini(&ac97->util_data); err_clk_put: -- cgit v1.2.3-70-g09d2 From 527659766778bcb0a2b739867ce4d95b4c57c7d7 Mon Sep 17 00:00:00 2001 From: Nenghua Cao Date: Fri, 13 Dec 2013 20:13:49 +0800 Subject: ASoC: dapm: update DPCM runtime when mixer/mux changes DPCM can dynamically alter the FE to BE PCM links at runtime based on mixer/mux setting updates. Add soc_dpcm_runtime_update() calling in get/put function for mixer/mux to support this feature. Signed-off-by: Nenghua Cao Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++++---- 1 file changed, 24 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcade130157..67e63ab1f11 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2868,6 +2868,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, unsigned int val; int connect, change; struct snd_soc_dapm_update update; + int ret = 0; if (snd_soc_volsw_is_stereo(mc)) dev_warn(codec->dapm.dev, @@ -2901,12 +2902,16 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, card->update = &update; } - soc_dapm_mixer_update_power(card, kcontrol, connect); + ret = soc_dapm_mixer_update_power(card, kcontrol, connect); card->update = NULL; } mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2955,6 +2960,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, mux, change; unsigned int mask; struct snd_soc_dapm_update update; + int ret = 0; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -2978,12 +2984,16 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; - soc_dapm_mux_update_power(card, kcontrol, mux, e); + ret = soc_dapm_mux_update_power(card, kcontrol, mux, e); card->update = NULL; } mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); @@ -3019,6 +3029,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; + int ret = 0; if (ucontrol->value.enumerated.item[0] >= e->max) return -EINVAL; @@ -3028,9 +3039,13 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, value = ucontrol->value.enumerated.item[0]; change = dapm_kcontrol_set_value(kcontrol, value); if (change) - soc_dapm_mux_update_power(card, kcontrol, value, e); + ret = soc_dapm_mux_update_power(card, kcontrol, value, e); mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); @@ -3097,6 +3112,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, mux, change; unsigned int mask; struct snd_soc_dapm_update update; + int ret = 0; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -3120,12 +3136,16 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; - soc_dapm_mux_update_power(card, kcontrol, mux, e); + ret = soc_dapm_mux_update_power(card, kcontrol, mux, e); card->update = NULL; } mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double); -- cgit v1.2.3-70-g09d2 From 252e91ff1094eefacd25b401c3b77e549803cae6 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 13 Dec 2013 14:43:02 +0800 Subject: ASoC: sgtl5000: read chip revision for once Store chip revision in struct sgtl5000_priv when sgtl5000_i2c_probe() reads it out from register, so that we can use it in sgtl5000_enable_regulators() with no need to read register again. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f4093f3f3a..bd291d2b327 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -115,6 +115,7 @@ struct sgtl5000_priv { struct ldo_regulator *ldo; struct regmap *regmap; struct clk *mclk; + int revision; }; /* @@ -1300,9 +1301,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { - int reg; int ret; - int rev; int i; int external_vddd = 0; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); @@ -1332,14 +1331,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) * workaround for revision 0x11 and later, * roll back to use internal LDO */ - - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, ®); - if (ret) - goto err_regulator_disable; - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - - if (external_vddd && rev >= 0x11) { + if (external_vddd && sgtl5000->revision >= 0x11) { /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); @@ -1362,9 +1354,6 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) return 0; -err_regulator_disable: - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); err_regulator_free: regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); @@ -1566,6 +1555,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); + sgtl5000->revision = rev; i2c_set_clientdata(client, sgtl5000); -- cgit v1.2.3-70-g09d2 From 7fd7a48bd584a66cc4b0e3b92bb75b061578e19e Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Mon, 16 Dec 2013 13:07:23 +0100 Subject: ASoC: axi-spdif: Use devm_ioremap_resource() instead of devm_request_and_ioremap() devm_request_and_ioremap() has been deprecated in favour of devm_ioremap_resource(). Fixes the following coccinelle warning: sound/soc/adi/axi-i2s.c:195:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 195 Generated by: coccinelle/api/devm_ioremap_resource.cocci Signed-off-by: Fengguang Wu Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/adi/axi-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 0822c771366..7f91a86dd73 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -192,9 +192,9 @@ static int axi_i2s_probe(struct platform_device *pdev) platform_set_drvdata(pdev, i2s); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_request_and_ioremap(&pdev->dev, res); - if (!base) - return -EBUSY; + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); i2s->regmap = devm_regmap_init_mmio(&pdev->dev, base, &axi_i2s_regmap_config); -- cgit v1.2.3-70-g09d2 From bbe580302d33cff282129e26c44f9c3450d6a086 Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Mon, 16 Dec 2013 13:07:24 +0100 Subject: ASoC: axi-spdif: Use devm_ioremap_resource() instead of devm_request_and_ioremap() devm_request_and_ioremap() has been deprecated in favour of devm_ioremap_resource(). Fixes the following coccinelle warning: sound/soc/adi/axi-spdif.c:194:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 194 Generated by: coccinelle/api/devm_ioremap_resource.cocci Signed-off-by: Fengguang Wu Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/adi/axi-spdif.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index d5408d23b43..8db7a992069 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -191,9 +191,9 @@ static int axi_spdif_probe(struct platform_device *pdev) platform_set_drvdata(pdev, spdif); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_request_and_ioremap(&pdev->dev, res); - if (!base) - return -EBUSY; + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); spdif->regmap = devm_regmap_init_mmio(&pdev->dev, base, &axi_spdif_regmap_config); -- cgit v1.2.3-70-g09d2 From 2b67f8ba41ac7acf01c8d5c742c713ead8b589cd Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 17 Dec 2013 15:16:40 +0800 Subject: ASoC: dmaengine: Add config parameter NULL check. Because the "ASoC: dmaengine-pcm: Provide default config" has provided us one defualt config of DMA. When using this, the config parameter of devm_snd_dmaengine_pcm_register() will be NULL, so here we need to have a check before using it. Signed-off-by: Xiubo Li Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 7483922f6ee..2a6c569d991 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -299,7 +299,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, !dev->of_node) return 0; - if (config->dma_dev) { + if (config && config->dma_dev) { /* * If this warning is seen, it probably means that your Linux * device structure does not match your HW device structure. @@ -317,7 +317,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = "rx-tx"; else name = dmaengine_pcm_dma_channel_names[i]; - if (config->chan_names[i]) + if (config && config->chan_names[i]) name = config->chan_names[i]; chan = dma_request_slave_channel_reason(dev, name); if (IS_ERR(chan)) { -- cgit v1.2.3-70-g09d2 From 3c43c69537daa044c61965fad24e24ad392c4166 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Dec 2013 00:49:22 +0000 Subject: ASoC: arizona: Use async writes Where possible write to the device asynchronously, allowing better performance when used with a bus like SPI which supports this by minimising the need to context switch back to the driver to get the next bit of data. Signed-off-by: Mark Brown Tested-by: Charles Keepax Reviewed-by: Charles Keepax --- sound/soc/codecs/arizona.c | 163 +++++++++++++++++++++++++-------------------- 1 file changed, 89 insertions(+), 74 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index eb9f5d4d892..6bfd8031c0c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -93,7 +93,7 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (!priv->spk_ena && manual_ena) { - snd_soc_write(codec, 0x4f5, 0x25a); + regmap_write_async(arizona->regmap, 0x4f5, 0x25a); priv->spk_ena_pending = true; } break; @@ -105,12 +105,13 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, return -EBUSY; } - snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1, - 1 << w->shift, 1 << w->shift); + regmap_update_bits_async(arizona->regmap, + ARIZONA_OUTPUT_ENABLES_1, + 1 << w->shift, 1 << w->shift); if (priv->spk_ena_pending) { msleep(75); - snd_soc_write(codec, 0x4f5, 0xda); + regmap_write_async(arizona->regmap, 0x4f5, 0xda); priv->spk_ena_pending = false; priv->spk_ena++; } @@ -119,16 +120,19 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, if (manual_ena) { priv->spk_ena--; if (!priv->spk_ena) - snd_soc_write(codec, 0x4f5, 0x25a); + regmap_write_async(arizona->regmap, + 0x4f5, 0x25a); } - snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1, - 1 << w->shift, 0); + regmap_update_bits_async(arizona->regmap, + ARIZONA_OUTPUT_ENABLES_1, + 1 << w->shift, 0); break; case SND_SOC_DAPM_POST_PMD: if (manual_ena) { if (!priv->spk_ena) - snd_soc_write(codec, 0x4f5, 0x0da); + regmap_write_async(arizona->regmap, + 0x4f5, 0x0da); } break; } @@ -687,6 +691,7 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, int event) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); + struct arizona *arizona = priv->arizona; unsigned int mask = 1 << w->shift; unsigned int val; @@ -709,7 +714,8 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, if (priv->arizona->hpdet_magic) val = 0; - snd_soc_update_bits(w->codec, ARIZONA_OUTPUT_ENABLES_1, mask, val); + regmap_update_bits_async(arizona->regmap, ARIZONA_OUTPUT_ENABLES_1, + mask, val); return arizona_out_ev(w, kcontrol, event); } @@ -864,6 +870,8 @@ EXPORT_SYMBOL_GPL(arizona_set_sysclk); static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; int lrclk, bclk, mode, base; base = dai->driver->base; @@ -920,17 +928,19 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, - ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR, - bclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL, - ARIZONA_AIF1TX_LRCLK_INV | - ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL, - ARIZONA_AIF1RX_LRCLK_INV | - ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT, - ARIZONA_AIF1_FMT_MASK, mode); + regmap_update_bits_async(arizona->regmap, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_INV | + ARIZONA_AIF1_BCLK_MSTR, + bclk); + regmap_update_bits_async(arizona->regmap, base + ARIZONA_AIF_TX_PIN_CTRL, + ARIZONA_AIF1TX_LRCLK_INV | + ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_RX_PIN_CTRL, + ARIZONA_AIF1RX_LRCLK_INV | + ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); + regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FORMAT, + ARIZONA_AIF1_FMT_MASK, mode); return 0; } @@ -1182,18 +1192,22 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, if (ret != 0) return ret; - snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, - ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, - ARIZONA_AIF1TX_BCPF_MASK, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE, - ARIZONA_AIF1RX_BCPF_MASK, lrclk); - snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1, - ARIZONA_AIF1TX_WL_MASK | - ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); - snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2, - ARIZONA_AIF1RX_WL_MASK | - ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); return 0; } @@ -1446,31 +1460,31 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, struct arizona_fll_cfg *cfg, int source, bool sync) { - regmap_update_bits(arizona->regmap, base + 3, - ARIZONA_FLL1_THETA_MASK, cfg->theta); - regmap_update_bits(arizona->regmap, base + 4, - ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); - regmap_update_bits(arizona->regmap, base + 5, - ARIZONA_FLL1_FRATIO_MASK, - cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); - regmap_update_bits(arizona->regmap, base + 6, - ARIZONA_FLL1_CLK_REF_DIV_MASK | - ARIZONA_FLL1_CLK_REF_SRC_MASK, - cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | - source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 3, + ARIZONA_FLL1_THETA_MASK, cfg->theta); + regmap_update_bits_async(arizona->regmap, base + 4, + ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); + regmap_update_bits_async(arizona->regmap, base + 5, + ARIZONA_FLL1_FRATIO_MASK, + cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 6, + ARIZONA_FLL1_CLK_REF_DIV_MASK | + ARIZONA_FLL1_CLK_REF_SRC_MASK, + cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | + source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); if (sync) - regmap_update_bits(arizona->regmap, base + 0x7, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 0x7, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); else - regmap_update_bits(arizona->regmap, base + 0x9, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + regmap_update_bits_async(arizona->regmap, base + 0x9, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); - regmap_update_bits(arizona->regmap, base + 2, - ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, - ARIZONA_FLL1_CTRL_UPD | cfg->n); + regmap_update_bits_async(arizona->regmap, base + 2, + ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, + ARIZONA_FLL1_CTRL_UPD | cfg->n); } static bool arizona_is_enabled_fll(struct arizona_fll *fll) @@ -1503,9 +1517,9 @@ static void arizona_enable_fll(struct arizona_fll *fll, */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - regmap_update_bits(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + regmap_update_bits_async(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, false); @@ -1515,15 +1529,15 @@ static void arizona_enable_fll(struct arizona_fll *fll, use_sync = true; } } else if (fll->sync_src >= 0) { - regmap_update_bits(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + regmap_update_bits_async(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, sync, fll->sync_src, false); - regmap_update_bits(arizona->regmap, fll->base + 0x11, - ARIZONA_FLL1_SYNC_ENA, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); } else { arizona_fll_err(fll, "No clocks provided\n"); return; @@ -1534,11 +1548,12 @@ static void arizona_enable_fll(struct arizona_fll *fll, * sync source. */ if (use_sync && fll->sync_freq > 100000) - regmap_update_bits(arizona->regmap, fll->base + 0x17, - ARIZONA_FLL1_SYNC_BW, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, 0); else - regmap_update_bits(arizona->regmap, fll->base + 0x17, - ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW); + regmap_update_bits_async(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, + ARIZONA_FLL1_SYNC_BW); if (!arizona_is_enabled_fll(fll)) pm_runtime_get(arizona->dev); @@ -1546,14 +1561,14 @@ static void arizona_enable_fll(struct arizona_fll *fll, /* Clear any pending completions */ try_wait_for_completion(&fll->ok); - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, 0); - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) - regmap_update_bits(arizona->regmap, fll->base + 0x11, - ARIZONA_FLL1_SYNC_ENA, - ARIZONA_FLL1_SYNC_ENA); + regmap_update_bits_async(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, + ARIZONA_FLL1_SYNC_ENA); ret = wait_for_completion_timeout(&fll->ok, msecs_to_jiffies(250)); @@ -1566,8 +1581,8 @@ static void arizona_disable_fll(struct arizona_fll *fll) struct arizona *arizona = fll->arizona; bool change; - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); regmap_update_bits_check(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, 0, &change); regmap_update_bits(arizona->regmap, fll->base + 0x11, -- cgit v1.2.3-70-g09d2 From bd4893492bfc6081da55e445d200c1a832770a06 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Dec 2013 00:49:50 +0000 Subject: ASoC: wm5102: Use async writes When writing the patch write to the device asynchronously, allowing better performance when used with a bus like SPI which supports this by minimising the need to context switch back to the driver to get the next bit of data. Signed-off-by: Mark Brown Reviewed-by: Charles Keepax --- sound/soc/codecs/wm5102.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a08e8bf6d07..ce9c8e14d4b 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -601,8 +601,8 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (patch) for (i = 0; i < patch_size; i++) - regmap_write(regmap, patch[i].reg, - patch[i].def); + regmap_write_async(regmap, patch[i].reg, + patch[i].def); break; default: -- cgit v1.2.3-70-g09d2 From 959e4083cd9c43fb3e818984926f9c590ee0aa2b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Dec 2013 00:49:56 +0000 Subject: ASoC: wm5110: Use async writes When writing the patch write to the device asynchronously, allowing better performance when used with a bus like SPI which supports this by minimising the need to context switch back to the driver to get the next bit of data. Signed-off-by: Mark Brown Tested-by: Charles Keepax Reviewed-by: Charles Keepax --- sound/soc/codecs/wm5110.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 0e63d8ce533..ebcbe7831e5 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -105,8 +105,8 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (patch) for (i = 0; i < patch_size; i++) - regmap_write(regmap, patch[i].reg, - patch[i].def); + regmap_write_async(regmap, patch[i].reg, + patch[i].def); break; default: -- cgit v1.2.3-70-g09d2 From 1f4fe272f068813377845a959cab5ce786a155bf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Dec 2013 00:50:04 +0000 Subject: ASoC: wm8997: Use async writes When writing the patch write to the device asynchronously, allowing better performance when used with a bus like SPI which supports this by minimising the need to context switch back to the driver to get the next bit of data. Signed-off-by: Mark Brown Reviewed-by: Charles Keepax --- sound/soc/codecs/wm8997.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 1392bb3c925..555115ee215 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -103,8 +103,8 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (patch) for (i = 0; i < patch_size; i++) - regmap_write(regmap, patch[i].reg, - patch[i].def); + regmap_write_async(regmap, patch[i].reg, + patch[i].def); break; default: break; -- cgit v1.2.3-70-g09d2 From 48b752ac2f80f483a3059ae109f9de02dcc054dd Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Tue, 17 Dec 2013 16:22:24 +0800 Subject: ASoC: mmp-pcm: config pcm slave via generic dmaengine use snd_dmaengine_pcm_prepare_slave_config to set slave config, and remove the max_burst_size = 4 hard code. select SND_SOC_GENERIC_DMAENGINE_PCM for mmp-pcm. Signed-off-by: Qiao Zhou Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/mmp-pcm.c | 18 +++--------------- 2 files changed, 4 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 4db74a083db..6473052b689 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -11,7 +11,7 @@ config SND_PXA2XX_SOC config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" depends on ARCH_MMP - select SND_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM select SND_ARM help Say Y if you want to add support for codecs attached to diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 7929e19b0ef..682ee52942b 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -67,27 +67,15 @@ static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_dmaengine_dai_dma_data *dma_params; struct dma_slave_config slave_config; int ret; - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - if (!dma_params) - return 0; - - ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); + ret = + snd_dmaengine_pcm_prepare_slave_config(substream, params, + &slave_config); if (ret) return ret; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.dst_addr = dma_params->addr; - slave_config.dst_maxburst = 4; - } else { - slave_config.src_addr = dma_params->addr; - slave_config.src_maxburst = 4; - } - ret = dmaengine_slave_config(chan, &slave_config); if (ret) return ret; -- cgit v1.2.3-70-g09d2 From 58e354be337dff43dbb66c4564bb9479354cc5dd Mon Sep 17 00:00:00 2001 From: Lucas Stach Date: Tue, 17 Dec 2013 11:52:43 +0100 Subject: ASoC: tegra: Tweak matching of AC97 components Matching works completely based on the cpu of_node. Signed-off-by: Lucas Stach Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm9712.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 5e119630b0e..45b57892b6a 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -55,7 +55,6 @@ static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", - .cpu_dai_name = "tegra20-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", .init = tegra_wm9712_init, -- cgit v1.2.3-70-g09d2 From 5095f55d7cc327026daaa3fa583aa4c1388ca556 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 18 Dec 2013 09:25:48 +0000 Subject: ASoC: wm_adsp: Remove duplicate info message for DSP RAM ready Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6b1c01c66e6..8f720ded27c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1488,7 +1488,6 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) } adsp_dbg(dsp, "RAM ready after %d polls\n", count); - adsp_info(dsp, "RAM ready after %d polls\n", count); return 0; } -- cgit v1.2.3-70-g09d2 From dd407a3243234c6a17ba624d698e6824067003c9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Dec 2013 13:50:10 +0000 Subject: ASoC: fsl/mxs: Remove unnecessarily gendered language The kernel as a number of cases of gendered language. The majority of these refer to objects that don't have gender in English, and so I've replaced them with "it" and "its". Some refer to people (developers or users), and I've replaced these with the singular "they" variant. Some are simply typos that I've fixed up. I've left cases where gendered language was used to refer to specific individuals, was a quote or is part of license text. Signed-off-by: Matthew Garrett Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/mxs/mxs-saif.c | 10 +++++----- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index fb9bb9eb5ca..d570f8c81dc 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -852,7 +852,7 @@ static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm) } /** - * find_ssi_node -- returns the SSI node that points to his DMA channel node + * find_ssi_node -- returns the SSI node that points to its DMA channel node * * Although this DMA driver attempts to operate independently of the other * devices, it still needs to determine some information about the SSI device diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 54e622acac3..d6cb9a51dd5 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -50,9 +50,9 @@ static struct mxs_saif *mxs_saif[2]; * This also means that both SAIFs must operate at the same sample rate. * * We abstract this as each saif has a master, the master could be - * himself or other saifs. In the generic saif driver, saif does not need - * to know the different clkmux. Saif only needs to know who is his master - * and operating his master to generate the proper clock rate for him. + * itself or other saifs. In the generic saif driver, saif does not need + * to know the different clkmux. Saif only needs to know who is its master + * and operating its master to generate the proper clock rate for it. * The master id is provided in mach-specific layer according to different * clkmux setting. */ @@ -76,7 +76,7 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK * is provided by other SAIF, we provide a interface here to get its master * from its master_id. - * Note that the master could be himself. + * Note that the master could be itself. */ static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) { @@ -516,7 +516,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, } /* - * If the saif's master is not himself, we also need to enable + * If the saif's master is not itself, we also need to enable * itself clk for its internal basic logic to work. */ if (saif != master_saif) { -- cgit v1.2.3-70-g09d2 From 4355082149429d1f87b6fbfc3ebc6305a5372ce2 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 17 Dec 2013 11:24:38 +0800 Subject: ASoC: Add SAI SoC Digital Audio Interface driver. This adds Freescale SAI ASoC Audio support. This implementation is only compatible with device tree definition. Features: o Supports playback/capture o Supports 16/20/24 bit PCM o Supports 8k - 96k sample rates o Supports master and slave mode. Signed-off-by: Alison Wang Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 4 + sound/soc/fsl/Makefile | 4 +- sound/soc/fsl/fsl_sai.c | 492 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_sai.h | 114 +++++++++++ 4 files changed, 613 insertions(+), 1 deletion(-) create mode 100644 sound/soc/fsl/fsl_sai.c create mode 100644 sound/soc/fsl/fsl_sai.h (limited to 'sound/soc') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index b7ab71f2ccc..ac4fe4ea15a 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,3 +1,7 @@ +config SND_SOC_FSL_SAI + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + config SND_SOC_FSL_SSI tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 8db705b0fdf..aaccbee1700 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -10,11 +10,13 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o -# Freescale PowerPC SSI/DMA Platform Support +# Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c new file mode 100644 index 00000000000..50a797e6578 --- /dev/null +++ b/sound/soc/fsl/fsl_sai.c @@ -0,0 +1,492 @@ +/* + * Freescale ALSA SoC Digital Audio Interface (SAI) driver. + * + * Copyright 2012-2013 Freescale Semiconductor, Inc. + * + * This program is free software, you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation, either version 2 of the License, or(at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "fsl_sai.h" + +static inline u32 sai_readl(struct fsl_sai *sai, + const void __iomem *addr) +{ + u32 val; + + val = __raw_readl(addr); + + if (likely(sai->big_endian_regs)) + val = be32_to_cpu(val); + else + val = le32_to_cpu(val); + rmb(); + + return val; +} + +static inline void sai_writel(struct fsl_sai *sai, + u32 val, void __iomem *addr) +{ + wmb(); + if (likely(sai->big_endian_regs)) + val = cpu_to_be32(val); + else + val = cpu_to_le32(val); + + __raw_writel(val, addr); +} + +static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int fsl_dir) +{ + u32 val_cr2, reg_cr2; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (fsl_dir == FSL_FMT_TRANSMITTER) + reg_cr2 = FSL_SAI_TCR2; + else + reg_cr2 = FSL_SAI_RCR2; + + val_cr2 = sai_readl(sai, sai->base + reg_cr2); + switch (clk_id) { + case FSL_SAI_CLK_BUS: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_BUS; + break; + case FSL_SAI_CLK_MAST1: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_MCLK1; + break; + case FSL_SAI_CLK_MAST2: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_MCLK2; + break; + case FSL_SAI_CLK_MAST3: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_MCLK3; + break; + default: + return -EINVAL; + } + sai_writel(sai, val_cr2, sai->base + reg_cr2); + + return 0; +} + +static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + int ret; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (dir == SND_SOC_CLOCK_IN) + return 0; + + ret = clk_prepare_enable(sai->clk); + if (ret) + return ret; + + sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); + sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); + sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); + sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); + + ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, + FSL_FMT_TRANSMITTER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's transmitter sysclk: %d\n", + ret); + return ret; + } + + ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, + FSL_FMT_RECEIVER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's receiver sysclk: %d\n", + ret); + return ret; + } + + clk_disable_unprepare(sai->clk); + + return 0; +} + +static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, + unsigned int fmt, int fsl_dir) +{ + u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (fsl_dir == FSL_FMT_TRANSMITTER) { + reg_cr2 = FSL_SAI_TCR2; + reg_cr3 = FSL_SAI_TCR3; + reg_cr4 = FSL_SAI_TCR4; + } else { + reg_cr2 = FSL_SAI_RCR2; + reg_cr3 = FSL_SAI_RCR3; + reg_cr4 = FSL_SAI_RCR4; + } + + val_cr2 = sai_readl(sai, sai->base + reg_cr2); + val_cr3 = sai_readl(sai, sai->base + reg_cr3); + val_cr4 = sai_readl(sai, sai->base + reg_cr4); + + if (sai->big_endian_data) + val_cr4 |= FSL_SAI_CR4_MF; + else + val_cr4 &= ~FSL_SAI_CR4_MF; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val_cr4 |= FSL_SAI_CR4_FSE; + val_cr4 |= FSL_SAI_CR4_FSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + val_cr4 |= FSL_SAI_CR4_FSP; + val_cr2 &= ~FSL_SAI_CR2_BCP; + break; + case SND_SOC_DAIFMT_IB_NF: + val_cr4 &= ~FSL_SAI_CR4_FSP; + val_cr2 &= ~FSL_SAI_CR2_BCP; + break; + case SND_SOC_DAIFMT_NB_IF: + val_cr4 |= FSL_SAI_CR4_FSP; + val_cr2 |= FSL_SAI_CR2_BCP; + break; + case SND_SOC_DAIFMT_NB_NF: + val_cr4 &= ~FSL_SAI_CR4_FSP; + val_cr2 |= FSL_SAI_CR2_BCP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val_cr2 |= FSL_SAI_CR2_BCD_MSTR; + val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; + val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; + break; + default: + return -EINVAL; + } + + val_cr3 |= FSL_SAI_CR3_TRCE; + + if (fsl_dir == FSL_FMT_RECEIVER) + val_cr2 |= FSL_SAI_CR2_SYNC; + + sai_writel(sai, val_cr2, sai->base + reg_cr2); + sai_writel(sai, val_cr3, sai->base + reg_cr3); + sai_writel(sai, val_cr4, sai->base + reg_cr4); + + return 0; +} + +static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + int ret; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + ret = clk_prepare_enable(sai->clk); + if (ret) + return ret; + + ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's transmitter format: %d\n", + ret); + return ret; + } + + ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's receiver format: %d\n", + ret); + return ret; + } + + clk_disable_unprepare(sai->clk); + + return 0; +} + +static int fsl_sai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr, word_width; + unsigned int channels = params_channels(params); + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg_cr4 = FSL_SAI_TCR4; + reg_cr5 = FSL_SAI_TCR5; + reg_mr = FSL_SAI_TMR; + } else { + reg_cr4 = FSL_SAI_RCR4; + reg_cr5 = FSL_SAI_RCR5; + reg_mr = FSL_SAI_RMR; + } + + val_cr4 = sai_readl(sai, sai->base + reg_cr4); + val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK; + val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK; + + val_cr5 = sai_readl(sai, sai->base + reg_cr5); + val_cr5 &= ~FSL_SAI_CR5_WNW_MASK; + val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; + val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_width = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + word_width = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + word_width = 24; + break; + default: + return -EINVAL; + } + + val_cr4 |= FSL_SAI_CR4_SYWD(word_width); + val_cr5 |= FSL_SAI_CR5_WNW(word_width); + val_cr5 |= FSL_SAI_CR5_W0W(word_width); + + if (sai->big_endian_data) + val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); + else + val_cr5 |= FSL_SAI_CR5_FBT(0); + + val_cr4 |= FSL_SAI_CR4_FRSZ(channels); + if (channels == 2 || channels == 1) + val_mr = ~0UL - ((1 << channels) - 1); + else + return -EINVAL; + + sai_writel(sai, val_cr4, sai->base + reg_cr4); + sai_writel(sai, val_cr5, sai->base + reg_cr5); + sai_writel(sai, val_mr, sai->base + reg_mr); + + return 0; +} + +static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + unsigned int tcsr, rcsr; + + tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); + rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + tcsr |= FSL_SAI_CSR_FRDE; + rcsr &= ~FSL_SAI_CSR_FRDE; + } else { + rcsr |= FSL_SAI_CSR_FRDE; + tcsr &= ~FSL_SAI_CSR_FRDE; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + tcsr |= FSL_SAI_CSR_TERE; + rcsr |= FSL_SAI_CSR_TERE; + sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); + sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!(cpu_dai->playback_active || cpu_dai->capture_active)) { + tcsr &= ~FSL_SAI_CSR_TERE; + rcsr &= ~FSL_SAI_CSR_TERE; + } + sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); + sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int fsl_sai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + int ret; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + ret = clk_prepare_enable(sai->clk); + + return ret; +} + +static void fsl_sai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + clk_disable_unprepare(sai->clk); +} + +static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { + .set_sysclk = fsl_sai_set_dai_sysclk, + .set_fmt = fsl_sai_set_dai_fmt, + .hw_params = fsl_sai_hw_params, + .trigger = fsl_sai_trigger, + .startup = fsl_sai_startup, + .shutdown = fsl_sai_shutdown, +}; + +static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); + + cpu_dai->playback_dma_data = &sai->dma_params_tx; + cpu_dai->capture_dma_data = &sai->dma_params_rx; + + snd_soc_dai_set_drvdata(cpu_dai, sai); + + return 0; +} + +static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai) +{ + cpu_dai->playback_dma_data = NULL; + cpu_dai->capture_dma_data = NULL; + + snd_soc_dai_set_drvdata(cpu_dai, NULL); + + return 0; +} + +static struct snd_soc_dai_driver fsl_sai_dai = { + .probe = fsl_sai_dai_probe, + .remove = fsl_sai_dai_remove, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_SAI_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_SAI_FORMATS, + }, + .ops = &fsl_sai_pcm_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_component = { + .name = "fsl-sai", +}; + +static int fsl_sai_probe(struct platform_device *pdev) +{ + int ret; + struct fsl_sai *sai; + struct resource *res; + struct device_node *np = pdev->dev.of_node; + + sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); + if (!sai) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + sai->base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(sai->base)) + return PTR_ERR(sai->base); + + sai->clk = devm_clk_get(&pdev->dev, "sai"); + if (IS_ERR(sai->clk)) { + dev_err(&pdev->dev, "Cannot get SAI's clock\n"); + return PTR_ERR(sai->clk); + } + + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; + sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; + sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; + sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX; + + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); + sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + + platform_set_drvdata(pdev, sai); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, + &fsl_sai_dai, 1); + if (ret) + return ret; + + ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + if (ret) + return ret; + + return 0; +} + +static int fsl_sai_remove(struct platform_device *pdev) +{ + snd_dmaengine_pcm_unregister(&pdev->dev); + + return 0; +} + +static const struct of_device_id fsl_sai_ids[] = { + { .compatible = "fsl,vf610-sai", }, + { /* sentinel */ } +}; + +static struct platform_driver fsl_sai_driver = { + .probe = fsl_sai_probe, + .remove = fsl_sai_remove, + + .driver = { + .name = "fsl-sai", + .owner = THIS_MODULE, + .of_match_table = fsl_sai_ids, + }, +}; +module_platform_driver(fsl_sai_driver); + +MODULE_DESCRIPTION("Freescale Soc SAI Interface"); +MODULE_AUTHOR("Xiubo Li, "); +MODULE_ALIAS("platform:fsl-sai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h new file mode 100644 index 00000000000..41bb62e6936 --- /dev/null +++ b/sound/soc/fsl/fsl_sai.h @@ -0,0 +1,114 @@ +/* + * Copyright 2012-2013 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __FSL_SAI_H +#define __FSL_SAI_H + +#include + +#define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +/* SAI Transmit/Recieve Control Register */ +#define FSL_SAI_TCSR 0x00 +#define FSL_SAI_RCSR 0x80 +#define FSL_SAI_CSR_TERE BIT(31) +#define FSL_SAI_CSR_FWF BIT(17) +#define FSL_SAI_CSR_FRIE BIT(8) +#define FSL_SAI_CSR_FRDE BIT(0) + +/* SAI Transmit Data/FIFO/MASK Register */ +#define FSL_SAI_TDR 0x20 +#define FSL_SAI_TFR 0x40 +#define FSL_SAI_TMR 0x60 + +/* SAI Recieve Data/FIFO/MASK Register */ +#define FSL_SAI_RDR 0xa0 +#define FSL_SAI_RFR 0xc0 +#define FSL_SAI_RMR 0xe0 + +/* SAI Transmit and Recieve Configuration 1 Register */ +#define FSL_SAI_TCR1 0x04 +#define FSL_SAI_RCR1 0x84 + +/* SAI Transmit and Recieve Configuration 2 Register */ +#define FSL_SAI_TCR2 0x08 +#define FSL_SAI_RCR2 0x88 +#define FSL_SAI_CR2_SYNC BIT(30) +#define FSL_SAI_CR2_MSEL_MASK (0xff << 26) +#define FSL_SAI_CR2_MSEL_BUS 0 +#define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) +#define FSL_SAI_CR2_MSEL_MCLK2 BIT(27) +#define FSL_SAI_CR2_MSEL_MCLK3 (BIT(26) | BIT(27)) +#define FSL_SAI_CR2_BCP BIT(25) +#define FSL_SAI_CR2_BCD_MSTR BIT(24) + +/* SAI Transmit and Recieve Configuration 3 Register */ +#define FSL_SAI_TCR3 0x0c +#define FSL_SAI_RCR3 0x8c +#define FSL_SAI_CR3_TRCE BIT(16) +#define FSL_SAI_CR3_WDFL(x) (x) +#define FSL_SAI_CR3_WDFL_MASK 0x1f + +/* SAI Transmit and Recieve Configuration 4 Register */ +#define FSL_SAI_TCR4 0x10 +#define FSL_SAI_RCR4 0x90 +#define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) +#define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) +#define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) +#define FSL_SAI_CR4_SYWD_MASK (0x1f << 8) +#define FSL_SAI_CR4_MF BIT(4) +#define FSL_SAI_CR4_FSE BIT(3) +#define FSL_SAI_CR4_FSP BIT(1) +#define FSL_SAI_CR4_FSD_MSTR BIT(0) + +/* SAI Transmit and Recieve Configuration 5 Register */ +#define FSL_SAI_TCR5 0x14 +#define FSL_SAI_RCR5 0x94 +#define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24) +#define FSL_SAI_CR5_WNW_MASK (0x1f << 24) +#define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16) +#define FSL_SAI_CR5_W0W_MASK (0x1f << 16) +#define FSL_SAI_CR5_FBT(x) ((x) << 8) +#define FSL_SAI_CR5_FBT_MASK (0x1f << 8) + +/* SAI type */ +#define FSL_SAI_DMA BIT(0) +#define FSL_SAI_USE_AC97 BIT(1) +#define FSL_SAI_NET BIT(2) +#define FSL_SAI_TRA_SYN BIT(3) +#define FSL_SAI_REC_SYN BIT(4) +#define FSL_SAI_USE_I2S_SLAVE BIT(5) + +#define FSL_FMT_TRANSMITTER 0 +#define FSL_FMT_RECEIVER 1 + +/* SAI clock sources */ +#define FSL_SAI_CLK_BUS 0 +#define FSL_SAI_CLK_MAST1 1 +#define FSL_SAI_CLK_MAST2 2 +#define FSL_SAI_CLK_MAST3 3 + +/* SAI data transfer numbers per DMA request */ +#define FSL_SAI_MAXBURST_TX 6 +#define FSL_SAI_MAXBURST_RX 6 + +struct fsl_sai { + struct clk *clk; + + void __iomem *base; + + bool big_endian_regs; + bool big_endian_data; + + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; +}; + +#endif /* __FSL_SAI_H */ -- cgit v1.2.3-70-g09d2 From 11db0da831b1e6ae3c1f8743599434281db294db Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 13 Dec 2013 14:43:03 +0800 Subject: ASoC: sgtl5000: clean up sgtl5000_enable_regulators() Function sgtl5000_enable_regulators() is somehow odd in handling the optional external VDDD supply. The driver can only enable this supply on SGTL5000 chip before revision 0x11, and of course when this external VDDD is present. It currently does something like below. 1. Check if regulator_bulk_get() on VDDA, VDDIO and VDDD will fail. If it fails, VDDD must be absent and it falls on internal LDO by calling sgtl5000_replace_vddd_with_ldo(). Otherwise, VDDD is used. And in either case, regulator_bulk_enable() will be called to enable 3 supplies. 2. In case that SGTL5000 revision is later than 0x11, even if external VDDD is present, it has to roll back the 'enable' and 'get' calls with regulator_bulk_disable() and regulator_bulk_free(), and starts over again by calling sgtl5000_replace_vddd_with_ldo() and regulator_bulk_enable(). Such back and forth calls sequence is complicated and unnecessary. Also, since commit 4ddfebd (regulator: core: Provide a dummy regulator with full constraints), regulator_bulk_get() will always succeeds because of the dummy regulator. Thus the VDDD detection is broken. The patch changes the flow to something like the following, which should be more reasonable and clear, and also fix the VDDD detection breakage. 1. Check if we're running a chip before revision 0x11, on which an external VDDD can possibly be an option. 2. If it is an early revision, call regulator_get_optional() to detect whether an external VDDD supply is available. 3. If external VDDD is present, call sgtl5000_replace_vddd_with_ldo() to update sgtl5000->supplies info. 4. Drop regulator_bulk_get() call in sgtl5000_replace_vddd_with_ldo(), and call it in sgtl5000_enable_regulators() no matter it's an external VDDD or internal LDO. 5. Call regulator_bulk_enable() to enable these 3 regulators. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 62 ++++++++++++++++----------------------------- 1 file changed, 22 insertions(+), 40 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index bd291d2b327..0fcbe90f3ef 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1286,15 +1286,6 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - if (ret) { - ldo_regulator_remove(codec); - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - dev_info(codec->dev, "Using internal LDO instead of VDDD\n"); return 0; } @@ -1305,20 +1296,35 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) int i; int external_vddd = 0; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + struct regulator *vddd; for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) sgtl5000->supplies[i].supply = supply_names[i]; - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (!ret) - external_vddd = 1; - else { + /* External VDDD only works before revision 0x11 */ + if (sgtl5000->revision < 0x11) { + vddd = regulator_get_optional(codec->dev, "VDDD"); + if (IS_ERR(vddd)) { + /* See if it's just not registered yet */ + if (PTR_ERR(vddd) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { + external_vddd = 1; + regulator_put(vddd); + } + } + + if (!external_vddd) { ret = sgtl5000_replace_vddd_with_ldo(codec); if (ret) return ret; } + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + goto err_ldo_remove; + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) @@ -1327,37 +1333,13 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) /* wait for all power rails bring up */ udelay(10); - /* - * workaround for revision 0x11 and later, - * roll back to use internal LDO - */ - if (external_vddd && sgtl5000->revision >= 0x11) { - /* disable all regulator first */ - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - /* free VDDD regulator */ - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - ret = sgtl5000_replace_vddd_with_ldo(codec); - if (ret) - return ret; - - ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (ret) - goto err_regulator_free; - - /* wait for all power rails bring up */ - udelay(10); - } - return 0; err_regulator_free: regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - if (external_vddd) +err_ldo_remove: + if (!external_vddd) ldo_regulator_remove(codec); return ret; -- cgit v1.2.3-70-g09d2 From 6f2032a18969d22740a865e0b4f2e48cf5338f36 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Dec 2013 12:34:45 -0700 Subject: ASoC: ep93xx: get rid of ep93xx-pcm-audio struct device Modify the ep93xx PCM driver so that it's a utility library that can be registered on each DAI, rather than a separate struct device. This is more in line with how many recent DT-converted platforms operate, and avoids the need for yet another struct device. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 8 ++++++++ sound/soc/cirrus/ep93xx-i2s.c | 8 ++++++++ sound/soc/cirrus/ep93xx-pcm.c | 19 +++++-------------- sound/soc/cirrus/ep93xx-pcm.h | 22 ++++++++++++++++++++++ sound/soc/cirrus/simone.c | 2 +- sound/soc/cirrus/snappercl15.c | 2 +- 7 files changed, 46 insertions(+), 17 deletions(-) create mode 100644 sound/soc/cirrus/ep93xx-pcm.h (limited to 'sound/soc') diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index c43fb214558..4f900efc437 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -63,7 +63,7 @@ static struct snd_soc_ops edb93xx_ops = { static struct snd_soc_dai_link edb93xx_dai = { .name = "CS4271", .stream_name = "CS4271 HiFi", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-i2s", .cpu_dai_name = "ep93xx-i2s", .codec_name = "spi0.0", .codec_dai_name = "cs4271-hifi", diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index efa75b5086a..cc5583da538 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -24,6 +24,8 @@ #include +#include "ep93xx-pcm.h" + /* * Per channel (1-4) registers. */ @@ -394,8 +396,14 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) if (ret) goto fail; + ret = devm_ep93xx_pcm_platform_register(&pdev->dev); + if (ret) + goto fail_unregister; + return 0; +fail_unregister: + snd_soc_unregister_component(&pdev->dev); fail: ep93xx_ac97_info = NULL; snd_soc_set_ac97_ops(NULL); diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index a57643d6402..167728a73da 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -30,6 +30,8 @@ #include #include +#include "ep93xx-pcm.h" + #define EP93XX_I2S_TXCLKCFG 0x00 #define EP93XX_I2S_RXCLKCFG 0x04 #define EP93XX_I2S_GLCTRL 0x0C @@ -405,8 +407,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) if (err) goto fail_put_lrclk; + err = devm_ep93xx_pcm_platform_register(&pdev->dev); + if (err) + goto fail_unregister; + return 0; +fail_unregister: + snd_soc_unregister_component(&pdev->dev); fail_put_lrclk: clk_put(info->lrclk); fail_put_sclk: diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index fdb8b8feb4e..198c5405f2c 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -23,6 +23,8 @@ #include +#include "ep93xx-pcm.h" + static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -76,27 +78,16 @@ static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { .prealloc_buffer_size = 131072, }; -static int ep93xx_soc_platform_probe(struct platform_device *pdev) +int devm_ep93xx_pcm_platform_register(struct device *dev) { - return devm_snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(dev, &ep93xx_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } - -static struct platform_driver ep93xx_pcm_driver = { - .driver = { - .name = "ep93xx-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = ep93xx_soc_platform_probe, -}; - -module_platform_driver(ep93xx_pcm_driver); +EXPORT_SYMBOL_GPL(devm_ep93xx_pcm_platform_register); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:ep93xx-pcm-audio"); diff --git a/sound/soc/cirrus/ep93xx-pcm.h b/sound/soc/cirrus/ep93xx-pcm.h new file mode 100644 index 00000000000..b7a12a2fae9 --- /dev/null +++ b/sound/soc/cirrus/ep93xx-pcm.h @@ -0,0 +1,22 @@ +/* + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + */ + +#ifndef __EP93XX_PCM_H__ +#define __EP93XX_PCM_H__ + +int devm_ep93xx_pcm_platform_register(struct device *dev); + +#endif diff --git a/sound/soc/cirrus/simone.c b/sound/soc/cirrus/simone.c index 4d094d00c34..822a19a89e7 100644 --- a/sound/soc/cirrus/simone.c +++ b/sound/soc/cirrus/simone.c @@ -27,7 +27,7 @@ static struct snd_soc_dai_link simone_dai = { .cpu_dai_name = "ep93xx-ac97", .codec_dai_name = "ac97-hifi", .codec_name = "ac97-codec", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-ac97", }; static struct snd_soc_card snd_soc_simone = { diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 69041074f2c..29238a7476d 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link snappercl15_dai = { .cpu_dai_name = "ep93xx-i2s", .codec_dai_name = "tlv320aic23-hifi", .codec_name = "tlv320aic23-codec.0-001a", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-i2s", .init = snappercl15_tlv320aic23_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS, -- cgit v1.2.3-70-g09d2 From a8983d4b0a8c439bddae0c9fd1e8a4cf7c402262 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Dec 2013 12:34:46 -0700 Subject: ASoC: ep93xx: remove custom DMA alloc compat function ep93xx_compat_request_channel() is almost identical to dmaengine_pcm_compat_request_channel(), with the exception that the latter: a) Assumes that the DAI DMA data is a struct snd_dmaengine_dai_dma_data pointer rather than some custom type. b) dma_data->filter_data rather than dma_data should be passed to snd_dmaengine_pcm_request_channel() as the filter data. Make minor changes to the ep93xx DAI drivers so that those two conditions are met. This allows removal of the custom .compat_request_channel(). Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-ac97.c | 12 ++++++++++-- sound/soc/cirrus/ep93xx-i2s.c | 14 ++++++++++++-- sound/soc/cirrus/ep93xx-pcm.c | 13 ------------- 3 files changed, 22 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index cc5583da538..f30dadf85b9 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -19,6 +19,7 @@ #include #include +#include #include #include @@ -97,6 +98,8 @@ struct ep93xx_ac97_info { struct device *dev; void __iomem *regs; struct completion done; + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; }; /* currently ALSA only supports a single AC97 device */ @@ -317,8 +320,13 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, static int ep93xx_ac97_dai_probe(struct snd_soc_dai *dai) { - dai->playback_dma_data = &ep93xx_ac97_pcm_out; - dai->capture_dma_data = &ep93xx_ac97_pcm_in; + struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai); + + info->dma_params_tx.filter_data = &ep93xx_ac97_pcm_out; + info->dma_params_rx.filter_data = &ep93xx_ac97_pcm_in; + + dai->playback_dma_data = &info->dma_params_tx; + dai->capture_dma_data = &info->dma_params_rx; return 0; } diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 167728a73da..943145f9d1b 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -21,6 +21,7 @@ #include #include +#include #include #include #include @@ -63,6 +64,8 @@ struct ep93xx_i2s_info { struct clk *sclk; struct clk *lrclk; void __iomem *regs; + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; }; static struct ep93xx_dma_data ep93xx_i2s_dma_data[] = { @@ -142,8 +145,15 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) static int ep93xx_i2s_dai_probe(struct snd_soc_dai *dai) { - dai->playback_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dai->capture_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE]; + struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); + + info->dma_params_tx.filter_data = + &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + info->dma_params_rx.filter_data = + &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + dai->playback_dma_data = &info->dma_params_tx; + dai->capture_dma_data = &info->dma_params_rx; return 0; } diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index 198c5405f2c..ca6698d7573 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -59,22 +59,9 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) return false; } -static struct dma_chan *ep93xx_compat_request_channel( - struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_substream *substream) -{ - struct snd_dmaengine_dai_dma_data *dma_data; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - return snd_dmaengine_pcm_request_channel(ep93xx_pcm_dma_filter, - dma_data); -} - static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { .pcm_hardware = &ep93xx_pcm_hardware, .compat_filter_fn = ep93xx_pcm_dma_filter, - .compat_request_channel = ep93xx_compat_request_channel, .prealloc_buffer_size = 131072, }; -- cgit v1.2.3-70-g09d2 From ede38884ac25ed78e43f3480056670963a9980f0 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Dec 2013 12:35:24 -0700 Subject: ASoC: SPEAr: get rid of spear-pcm-audio struct device Modify the SPEAr PCM driver so that it's a utility library that can be registered on each DAI, rather than a separate struct device. This is more in line with how many recent DT-converted platforms operate, and avoids the need for yet another struct device. This is also required as a pre-cursor to removing spear_pcm_request_chan(). Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/spear/spdif_in.c | 9 +++++++-- sound/soc/spear/spdif_out.c | 10 ++++++++-- sound/soc/spear/spear_pcm.c | 18 ++++-------------- sound/soc/spear/spear_pcm.h | 22 ++++++++++++++++++++++ 4 files changed, 41 insertions(+), 18 deletions(-) create mode 100644 sound/soc/spear/spear_pcm.h (limited to 'sound/soc') diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 21a8c954af1..4627110f344 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -24,6 +24,7 @@ #include #include #include "spdif_in_regs.h" +#include "spear_pcm.h" struct spdif_in_params { u32 format; @@ -257,8 +258,12 @@ static int spdif_in_probe(struct platform_device *pdev) return ret; } - return devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, - &spdif_in_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, + &spdif_in_dai, 1); + if (ret) + return ret; + + return devm_spear_pcm_platform_register(&pdev->dev); } static struct platform_driver spdif_in_driver = { diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index b6ef6f78dc7..731a1e0cfea 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -22,6 +22,7 @@ #include #include #include "spdif_out_regs.h" +#include "spear_pcm.h" struct spdif_out_params { u32 rate; @@ -280,6 +281,7 @@ static int spdif_out_probe(struct platform_device *pdev) struct spdif_out_dev *host; struct spear_spdif_platform_data *pdata; struct resource *res; + int ret; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); if (!host) { @@ -306,8 +308,12 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - return devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, - &spdif_out_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, + &spdif_out_dai, 1); + if (ret) + return ret; + + return devm_spear_pcm_platform_register(&pdev->dev); } #ifdef CONFIG_PM diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9a02141666e..f288724961d 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -18,6 +18,7 @@ #include #include #include +#include "spear_pcm.h" static const struct snd_pcm_hardware spear_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -47,26 +48,15 @@ static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = { .prealloc_buffer_size = 16 * 1024, }; -static int spear_soc_platform_probe(struct platform_device *pdev) +int devm_spear_pcm_platform_register(struct device *dev) { - return devm_snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(dev, &spear_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } - -static struct platform_driver spear_pcm_driver = { - .driver = { - .name = "spear-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = spear_soc_platform_probe, -}; - -module_platform_driver(spear_pcm_driver); +EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register); MODULE_AUTHOR("Rajeev Kumar "); MODULE_DESCRIPTION("SPEAr PCM DMA module"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:spear-pcm-audio"); diff --git a/sound/soc/spear/spear_pcm.h b/sound/soc/spear/spear_pcm.h new file mode 100644 index 00000000000..631e2aa1fb3 --- /dev/null +++ b/sound/soc/spear/spear_pcm.h @@ -0,0 +1,22 @@ +/* + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + */ + +#ifndef __SPEAR_PCM_H__ +#define __SPEAR_PCM_H__ + +int devm_spear_pcm_platform_register(struct device *dev); + +#endif -- cgit v1.2.3-70-g09d2 From e1771bcf99b0dc91f4ba645c1740fd5031702f49 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Dec 2013 12:35:25 -0700 Subject: ASoC: SPEAr: remove custom DMA alloc compat function spear_pcm_request_chan() is almost identical to dmaengine_pcm_compat_request_channel(), with the exception that the latter: a) Assumes that the DAI DMA data is a struct snd_dmaengine_dai_dma_data pointer rather than some custom type. b) dma_data->filter_data rather than dma_data should be passed to snd_dmaengine_pcm_request_channel() as the filter data. Make minor changes to the SPEAr DAI drivers so that those two conditions are met. This allows removal of the custom .compat_request_channel(). Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- include/sound/spear_dma.h | 1 - sound/soc/spear/spdif_in.c | 10 +++++++--- sound/soc/spear/spdif_out.c | 10 +++++++--- sound/soc/spear/spear_pcm.c | 21 +++++++-------------- sound/soc/spear/spear_pcm.h | 4 +++- 5 files changed, 24 insertions(+), 22 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/spear_dma.h b/include/sound/spear_dma.h index 1b365bfdfb3..65aca51fe25 100644 --- a/include/sound/spear_dma.h +++ b/include/sound/spear_dma.h @@ -29,7 +29,6 @@ struct spear_dma_data { dma_addr_t addr; u32 max_burst; enum dma_slave_buswidth addr_width; - bool (*filter)(struct dma_chan *chan, void *slave); }; #endif /* SPEAR_DMA_H */ diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 4627110f344..4ab442a63d7 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -38,6 +39,8 @@ struct spdif_in_dev { struct device *dev; void (*reset_perip)(void); int irq; + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_pcm_config config; }; static void spdif_in_configure(struct spdif_in_dev *host) @@ -54,7 +57,8 @@ static int spdif_in_dai_probe(struct snd_soc_dai *dai) { struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); - dai->capture_dma_data = &host->dma_params; + host->dma_params_rx.filter_data = &host->dma_params; + dai->capture_dma_data = &host->dma_params_rx; return 0; } @@ -245,7 +249,6 @@ static int spdif_in_probe(struct platform_device *pdev) host->dma_params.addr = res_fifo->start; host->dma_params.max_burst = 16; host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - host->dma_params.filter = pdata->filter; host->reset_perip = pdata->reset_perip; host->dev = &pdev->dev; @@ -263,7 +266,8 @@ static int spdif_in_probe(struct platform_device *pdev) if (ret) return ret; - return devm_spear_pcm_platform_register(&pdev->dev); + return devm_spear_pcm_platform_register(&pdev->dev, &host->config, + pdata->filter); } static struct platform_driver spdif_in_driver = { diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 731a1e0cfea..fe99f461aff 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -36,6 +37,8 @@ struct spdif_out_dev { struct spdif_out_params saved_params; u32 running; void __iomem *io_base; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct snd_dmaengine_pcm_config config; }; static void spdif_out_configure(struct spdif_out_dev *host) @@ -245,7 +248,8 @@ static int spdif_soc_dai_probe(struct snd_soc_dai *dai) { struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); - dai->playback_dma_data = &host->dma_params; + host->dma_params_tx.filter_data = &host->dma_params; + dai->playback_dma_data = &host->dma_params_tx; return snd_soc_add_dai_controls(dai, spdif_out_controls, ARRAY_SIZE(spdif_out_controls)); @@ -304,7 +308,6 @@ static int spdif_out_probe(struct platform_device *pdev) host->dma_params.addr = res->start + SPDIF_OUT_FIFO_DATA; host->dma_params.max_burst = 16; host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - host->dma_params.filter = pdata->filter; dev_set_drvdata(&pdev->dev, host); @@ -313,7 +316,8 @@ static int spdif_out_probe(struct platform_device *pdev) if (ret) return ret; - return devm_spear_pcm_platform_register(&pdev->dev); + return devm_spear_pcm_platform_register(&pdev->dev, &host->config, + pdata->filter); } #ifdef CONFIG_PM diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index f288724961d..0e5a8f35d0a 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -32,26 +32,19 @@ static const struct snd_pcm_hardware spear_pcm_hardware = { .fifo_size = 0, /* fifo size in bytes */ }; -static struct dma_chan *spear_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_substream *substream) -{ - struct spear_dma_data *dma_data; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - return snd_dmaengine_pcm_request_channel(dma_data->filter, dma_data); -} - static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = { .pcm_hardware = &spear_pcm_hardware, - .compat_request_channel = spear_pcm_request_chan, .prealloc_buffer_size = 16 * 1024, }; -int devm_spear_pcm_platform_register(struct device *dev) +int devm_spear_pcm_platform_register(struct device *dev, + struct snd_dmaengine_pcm_config *config, + bool (*filter)(struct dma_chan *chan, void *slave)) { - return devm_snd_dmaengine_pcm_register(dev, - &spear_dmaengine_pcm_config, + *config = spear_dmaengine_pcm_config; + config->compat_filter_fn = filter; + + return snd_dmaengine_pcm_register(dev, config, SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } diff --git a/sound/soc/spear/spear_pcm.h b/sound/soc/spear/spear_pcm.h index 631e2aa1fb3..9b0ca62d6f0 100644 --- a/sound/soc/spear/spear_pcm.h +++ b/sound/soc/spear/spear_pcm.h @@ -17,6 +17,8 @@ #ifndef __SPEAR_PCM_H__ #define __SPEAR_PCM_H__ -int devm_spear_pcm_platform_register(struct device *dev); +int devm_spear_pcm_platform_register(struct device *dev, + struct snd_dmaengine_pcm_config *config, + bool (*filter)(struct dma_chan *chan, void *slave)); #endif -- cgit v1.2.3-70-g09d2 From a4c0d2735c66c2363100575324a2f7659fb1f684 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 Dec 2013 09:11:02 +0000 Subject: ASoC: wm5110: Expose switch controls for DRE Certain use-cases require the DRE to be disabled so expose controls for the enables. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index ebcbe7831e5..3487ffa2e95 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -371,6 +371,13 @@ SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, ARIZONA_SPK2R_MUTE_SHIFT, 1, 1), +SOC_DOUBLE("HPOUT1 DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE1L_ENA_SHIFT, ARIZONA_DRE1R_ENA_SHIFT, 1, 0), +SOC_DOUBLE("HPOUT2 DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE2L_ENA_SHIFT, ARIZONA_DRE2R_ENA_SHIFT, 1, 0), +SOC_DOUBLE("HPOUT3 DRE Switch", ARIZONA_DRE_ENABLE, + ARIZONA_DRE3L_ENA_SHIFT, ARIZONA_DRE3R_ENA_SHIFT, 1, 0), + SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), -- cgit v1.2.3-70-g09d2 From fbedc8cbc3c40281cf52ed0e2e5998dea98e2992 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 Dec 2013 09:30:12 +0000 Subject: ASoC: wm5110: Add FSH for ISRCs Currently, the driver only supports configuration of the lower sample rate (FSL) on the ISRCs. With the higher rate being fixed a SYSCLK, this patch adds support for configuring the higher sample rate (FSH). Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 16 ++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 3 +++ 3 files changed, 20 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6bfd8031c0c..56d3ff570ad 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -498,6 +498,22 @@ const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { EXPORT_SYMBOL_GPL(arizona_rate_val); +const struct soc_enum arizona_isrc_fsh[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_1, + ARIZONA_ISRC1_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_2_CTRL_1, + ARIZONA_ISRC2_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_3_CTRL_1, + ARIZONA_ISRC3_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), +}; +EXPORT_SYMBOL_GPL(arizona_isrc_fsh); + const struct soc_enum arizona_isrc_fsl[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_2, ARIZONA_ISRC1_FSL_SHIFT, 0xf, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 6641f3d6419..99a97c06d46 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -186,6 +186,7 @@ extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; extern const struct soc_enum arizona_isrc_fsl[]; +extern const struct soc_enum arizona_isrc_fsh[]; extern const struct soc_enum arizona_in_vi_ramp; extern const struct soc_enum arizona_in_vd_ramp; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 3487ffa2e95..57be82261c2 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -280,6 +280,9 @@ SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), SOC_VALUE_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]), +SOC_VALUE_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]), +SOC_VALUE_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), +SOC_VALUE_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]), ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), -- cgit v1.2.3-70-g09d2 From 56d37d85438df38e150282baafe52dcd588854c7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 Dec 2013 09:30:13 +0000 Subject: ASoC: wm5110: Add support for ASRC RATE 1 Add support for configuring the sample rate on the SYSCLK side of the ASRC. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 7 +++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 1 + 3 files changed, 9 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 56d3ff570ad..e4295fee8f1 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -530,6 +530,13 @@ const struct soc_enum arizona_isrc_fsl[] = { }; EXPORT_SYMBOL_GPL(arizona_isrc_fsl); +const struct soc_enum arizona_asrc_rate1 = + SOC_VALUE_ENUM_SINGLE(ARIZONA_ASRC_RATE1, + ARIZONA_ASRC_RATE1_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE - 1, + arizona_rate_text, arizona_rate_val); +EXPORT_SYMBOL_GPL(arizona_asrc_rate1); + static const char *arizona_vol_ramp_text[] = { "0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB", "15ms/6dB", "30ms/6dB", diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 99a97c06d46..10b39847720 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -187,6 +187,7 @@ extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; extern const struct soc_enum arizona_isrc_fsl[]; extern const struct soc_enum arizona_isrc_fsh[]; +extern const struct soc_enum arizona_asrc_rate1; extern const struct soc_enum arizona_in_vi_ramp; extern const struct soc_enum arizona_in_vd_ramp; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 57be82261c2..f3d96eae031 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -283,6 +283,7 @@ SOC_VALUE_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]), SOC_VALUE_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]), SOC_VALUE_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), SOC_VALUE_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]), +SOC_VALUE_ENUM("ASRC RATE 1", arizona_asrc_rate1), ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), -- cgit v1.2.3-70-g09d2 From e1acb40a3addc9aceb4600f04c9c86b50770b9b8 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Thu, 19 Dec 2013 11:59:54 +0800 Subject: ASoC: simple-card: Use devm_snd_soc_register_card() Makes the code slightly shorter. Signed-off-by: Xiubo Li Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7a9b6b4898c..3d190d05ad4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -234,14 +234,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_card.num_links = 1; cinfo->snd_card.dev = &pdev->dev; - return snd_soc_register_card(&cinfo->snd_card); -} - -static int asoc_simple_card_remove(struct platform_device *pdev) -{ - struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; - - return snd_soc_unregister_card(&cinfo->snd_card); + return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); } static const struct of_device_id asoc_simple_of_match[] = { @@ -257,7 +250,6 @@ static struct platform_driver asoc_simple_card = { .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, - .remove = asoc_simple_card_remove, }; module_platform_driver(asoc_simple_card); -- cgit v1.2.3-70-g09d2 From 05004cb4cd06127bb8ff70d5ab5a915103828e9d Mon Sep 17 00:00:00 2001 From: Matthew Garrett Date: Wed, 18 Dec 2013 13:50:10 +0000 Subject: ASoC: fsl/mxs: Remove unnecessarily gendered language The kernel as a number of cases of gendered language. The majority of these refer to objects that don't have gender in English, and so I've replaced them with "it" and "its". Some refer to people (developers or users), and I've replaced these with the singular "they" variant. Some are simply typos that I've fixed up. I've left cases where gendered language was used to refer to specific individuals, was a quote or is part of license text. Signed-off-by: Matthew Garrett Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/mxs/mxs-saif.c | 10 +++++----- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index fb9bb9eb5ca..d570f8c81dc 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -852,7 +852,7 @@ static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm) } /** - * find_ssi_node -- returns the SSI node that points to his DMA channel node + * find_ssi_node -- returns the SSI node that points to its DMA channel node * * Although this DMA driver attempts to operate independently of the other * devices, it still needs to determine some information about the SSI device diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 54e622acac3..d6cb9a51dd5 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -50,9 +50,9 @@ static struct mxs_saif *mxs_saif[2]; * This also means that both SAIFs must operate at the same sample rate. * * We abstract this as each saif has a master, the master could be - * himself or other saifs. In the generic saif driver, saif does not need - * to know the different clkmux. Saif only needs to know who is his master - * and operating his master to generate the proper clock rate for him. + * itself or other saifs. In the generic saif driver, saif does not need + * to know the different clkmux. Saif only needs to know who is its master + * and operating its master to generate the proper clock rate for it. * The master id is provided in mach-specific layer according to different * clkmux setting. */ @@ -76,7 +76,7 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK * is provided by other SAIF, we provide a interface here to get its master * from its master_id. - * Note that the master could be himself. + * Note that the master could be itself. */ static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) { @@ -516,7 +516,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, } /* - * If the saif's master is not himself, we also need to enable + * If the saif's master is not itself, we also need to enable * itself clk for its internal basic logic to work. */ if (saif != master_saif) { -- cgit v1.2.3-70-g09d2 From aafa85e71a75fdea9076c5e0f7fe09e12568c9a4 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 12 Dec 2013 18:44:45 +0800 Subject: ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series This patch adds three main functions for DAI master mode: set_dai_fmt(), set_dai_sysclk() and set_dai_tdm_slot(), and one essential baud clock accordingly. After appending this patch, the fsl_ssi driver on i.MX series has the ability to derive LRCLK and BCLK from baud clock source so as to support some audio Codecs which can only be used in slave mode. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 280 +++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_ssi.h | 2 + 2 files changed, 278 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f9f4569417e..b2ebaf81159 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -38,6 +38,7 @@ #include #include #include +#include #include #include #include @@ -139,7 +140,10 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + bool baudclk_locked; u8 i2s_mode; + spinlock_t baudclk_lock; + struct clk *baudclk; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -434,13 +438,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + unsigned long flags; /* First, we only do fsl_ssi_setup() when SSI is going to be active. * Second, fsl_ssi_setup was already called by ac97_init earlier if * the driver is in ac97 mode. */ - if (!dai->active && !ssi_private->imx_ac97) + if (!dai->active && !ssi_private->imx_ac97) { fsl_ssi_setup(ssi_private); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } return 0; } @@ -501,6 +510,243 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, return 0; } +/** + * fsl_ssi_set_dai_fmt - configure Digital Audio Interface Format. + */ +static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 strcr = 0, stcr, srcr, scr, mask; + + scr = read_ssi(&ssi->scr) & ~(CCSR_SSI_SCR_SYN | CCSR_SSI_SCR_I2S_MODE_MASK); + scr |= CCSR_SSI_SCR_NET; + + mask = CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR | + CCSR_SSI_STCR_TSCKP | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TFSL | + CCSR_SSI_STCR_TEFS; + stcr = read_ssi(&ssi->stcr) & ~mask; + srcr = read_ssi(&ssi->srcr) & ~mask; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFM: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + break; + default: + return -EINVAL; + } + scr |= ssi_private->i2s_mode; + + /* Data on rising edge of bclk, frame low, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + strcr ^= CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + strcr ^= CCSR_SSI_STCR_TFSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + strcr ^= CCSR_SSI_STCR_TSCKP; + strcr ^= CCSR_SSI_STCR_TFSI; + break; + default: + return -EINVAL; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + strcr |= CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR; + scr |= CCSR_SSI_SCR_SYS_CLK_EN; + break; + case SND_SOC_DAIFMT_CBM_CFM: + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + stcr |= strcr; + srcr |= strcr; + + if (ssi_private->cpu_dai_drv.symmetric_rates) { + /* Need to clear RXDIR when using SYNC mode */ + srcr &= ~CCSR_SSI_SRCR_RXDIR; + scr |= CCSR_SSI_SCR_SYN; + } + + write_ssi(stcr, &ssi->stcr); + write_ssi(srcr, &ssi->srcr); + write_ssi(scr, &ssi->scr); + + return 0; +} + +/** + * fsl_ssi_set_dai_sysclk - configure Digital Audio Interface bit clock + * + * Note: This function can be only called when using SSI as DAI master + * + * Quick instruction for parameters: + * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels + * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + */ +static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + unsigned long flags, clkrate, baudrate, tmprate; + u64 sub, savesub = 100000; + + /* Don't apply it to any non-baudclk circumstance */ + if (IS_ERR(ssi_private->baudclk)) + return -EINVAL; + + /* It should be already enough to divide clock by setting pm alone */ + psr = 0; + div2 = 0; + + factor = (div2 + 1) * (7 * psr + 1) * 2; + + for (i = 0; i < 255; i++) { + /* The bclk rate must be smaller than 1/5 sysclk rate */ + if (factor * (i + 1) < 5) + continue; + + tmprate = freq * factor * (i + 2); + clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + + do_div(clkrate, factor); + afreq = (u32)clkrate / (i + 1); + + if (freq == afreq) + sub = 0; + else if (freq / afreq == 1) + sub = freq - afreq; + else if (afreq / freq == 1) + sub = afreq - freq; + else + continue; + + /* Calculate the fraction */ + sub *= 100000; + do_div(sub, freq); + + if (sub < savesub) { + baudrate = tmprate; + savesub = sub; + pm = i; + } + + /* We are lucky */ + if (savesub == 0) + break; + } + + /* No proper pm found if it is still remaining the initial value */ + if (pm == 999) { + dev_err(cpu_dai->dev, "failed to handle the required sysclk\n"); + return -EINVAL; + } + + stccr = CCSR_SSI_SxCCR_PM(pm + 1) | (div2 ? CCSR_SSI_SxCCR_DIV2 : 0) | + (psr ? CCSR_SSI_SxCCR_PSR : 0); + mask = CCSR_SSI_SxCCR_PM_MASK | CCSR_SSI_SxCCR_DIV2 | CCSR_SSI_SxCCR_PSR; + + if (dir == SND_SOC_CLOCK_OUT || synchronous) + write_ssi_mask(&ssi->stccr, mask, stccr); + else + write_ssi_mask(&ssi->srccr, mask, stccr); + + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + if (!ssi_private->baudclk_locked) { + ret = clk_set_rate(ssi_private->baudclk, baudrate); + if (ret) { + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + dev_err(cpu_dai->dev, "failed to set baudclk rate\n"); + return -EINVAL; + } + ssi_private->baudclk_locked = true; + } + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + + return 0; +} + +/** + * fsl_ssi_set_dai_tdm_slot - set TDM slot number + * + * Note: This function can be only called when using SSI as DAI master + */ +static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 val; + + /* The slot number should be >= 2 if using Network mode or I2S mode */ + val = read_ssi(&ssi->scr) & (CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET); + if (val && slots < 2) { + dev_err(cpu_dai->dev, "slot number should be >= 2 in I2S or NET\n"); + return -EINVAL; + } + + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + + /* The register SxMSKs needs SSI to provide essential clock due to + * hardware design. So we here temporarily enable SSI to set them. + */ + val = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN; + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN); + + write_ssi(tx_mask, &ssi->stmsk); + write_ssi(rx_mask, &ssi->srmsk); + + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, val); + + return 0; +} + /** * fsl_ssi_trigger: start and stop the DMA transfer. * @@ -517,6 +763,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; unsigned int sier_bits; + unsigned long flags; /* * Enable only the interrupts and DMA requests @@ -557,8 +804,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) & - (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) { write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } break; default: @@ -585,6 +836,9 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, + .set_fmt = fsl_ssi_set_dai_fmt, + .set_sysclk = fsl_ssi_set_dai_sysclk, + .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; @@ -897,6 +1151,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + ssi_private->baudclk_locked = false; + spin_lock_init(&ssi_private->baudclk_lock); + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { u32 dma_events[2]; ssi_private->ssi_on_imx = true; @@ -914,6 +1171,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) goto error_irqmap; } + /* For those SLAVE implementations, we ingore non-baudclk cases + * and, instead, abandon MASTER mode that needs baud clock. + */ + ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); + if (IS_ERR(ssi_private->baudclk)) + dev_warn(&pdev->dev, "could not get baud clock: %d\n", ret); + else + clk_prepare_enable(ssi_private->baudclk); + /* * We have burstsize be "fifo_depth - 2" to match the SSI * watermark setting in fsl_ssi_startup(). @@ -1059,8 +1325,11 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } error_irqmap: irq_dispose_mapping(ssi_private->irq); @@ -1076,8 +1345,11 @@ static int fsl_ssi_remove(struct platform_device *pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } irq_dispose_mapping(ssi_private->irq); return 0; diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index e6b9a69e2a6..e6b63240a3d 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -125,7 +125,9 @@ struct ccsr_ssi { #define CCSR_SSI_SRCR_REFS 0x00000001 /* STCCR and SRCCR */ +#define CCSR_SSI_SxCCR_DIV2_SHIFT 18 #define CCSR_SSI_SxCCR_DIV2 0x00040000 +#define CCSR_SSI_SxCCR_PSR_SHIFT 17 #define CCSR_SSI_SxCCR_PSR 0x00020000 #define CCSR_SSI_SxCCR_WL_SHIFT 13 #define CCSR_SSI_SxCCR_WL_MASK 0x0001E000 -- cgit v1.2.3-70-g09d2 From 0d35d080ac93f317b6c47180d75c8e1a8109b4c4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Nov 2013 16:03:43 +0200 Subject: ASoC: twl6040: Rename twl6040_is_path_unmuted -> twl6040_can_write_to_chip Matches more precisely of the functionality. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index f2f4bcb2ff7..ef13a501a7b 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -219,8 +219,8 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, return value; } -static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, - unsigned int reg) +static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec, + unsigned int reg) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); @@ -250,7 +250,7 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - if (twl6040_is_path_unmuted(codec, reg)) + if (twl6040_can_write_to_chip(codec, reg)) return twl6040_reg_write(twl6040, reg, value); else return 0; -- cgit v1.2.3-70-g09d2 From 53509108f7372f786576d7d43f8f881cdf51d38a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Nov 2013 16:03:44 +0200 Subject: ASoC: twl6040: Custom caching for sensitive DL1/2 path registers Introduce a small register cache for registers which needs special caching to reduce pop noise: TWL6040_REG_HSLCTL, TWL6040_REG_HSRCTL, TWL6040_REG_EARCTL, TWL6040_REG_HFLCTL and TWL6040_REG_HFRCTL. Switch over and use the new small cache for these registers instead of the main reg_cache. This is in preparation to remove the local ASoC reg_cache from the driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 51 ++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 49 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index ef13a501a7b..fb8c65bd6e5 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -72,6 +72,7 @@ struct twl6040_data { int hs_power_mode_locked; bool dl1_unmuted; bool dl2_unmuted; + u8 dl12_cache[TWL6040_REG_HFRCTL - TWL6040_REG_HSLCTL + 1]; unsigned int clk_in; unsigned int sysclk; struct twl6040_jack_data hs_jack; @@ -174,18 +175,62 @@ static struct snd_pcm_hw_constraint_list sysclk_constraints[] = { { .count = ARRAY_SIZE(hp_rates), .list = hp_rates, }, }; +static inline int twl6040_read_dl12_cache(struct snd_soc_codec *codec, + u8 reg, u8 *value) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (reg) { + case TWL6040_REG_HSLCTL: + case TWL6040_REG_HSRCTL: + case TWL6040_REG_EARCTL: + case TWL6040_REG_HFLCTL: + case TWL6040_REG_HFRCTL: + *value = priv->dl12_cache[reg - TWL6040_REG_HSLCTL]; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + /* * read twl6040 register cache */ static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) + unsigned int reg) { u8 *cache = codec->reg_cache; + u8 value; if (reg >= TWL6040_CACHEREGNUM) return -EIO; - return cache[reg]; + if (twl6040_read_dl12_cache(codec, reg, &value)) + value = cache[reg]; + + return value; +} + +static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); + + switch (reg) { + case TWL6040_REG_HSLCTL: + case TWL6040_REG_HSRCTL: + case TWL6040_REG_EARCTL: + case TWL6040_REG_HFLCTL: + case TWL6040_REG_HFRCTL: + priv->dl12_cache[reg - TWL6040_REG_HSLCTL] = value; + break; + default: + break; + } } /* @@ -199,6 +244,8 @@ static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return; cache[reg] = value; + + twl6040_update_dl12_cache(codec, reg, value); } /* -- cgit v1.2.3-70-g09d2 From 79ae5130381cb7d3e74c162123392ba6067c218e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Nov 2013 16:03:46 +0200 Subject: ASoC: twl6040: Remove register restore functionality The MFD core takes care of the restore via standard regmap API, no need to do this anymore here. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 29 ----------------------------- 1 file changed, 29 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index fb8c65bd6e5..b07839b2d91 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -133,22 +133,6 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x00, /* REG_STATUS 0x2E (ro) */ }; -/* List of registers to be restored after power up */ -static const int twl6040_restore_list[] = { - TWL6040_REG_MICLCTL, - TWL6040_REG_MICRCTL, - TWL6040_REG_MICGAIN, - TWL6040_REG_LINEGAIN, - TWL6040_REG_HSLCTL, - TWL6040_REG_HSRCTL, - TWL6040_REG_HSGAIN, - TWL6040_REG_EARCTL, - TWL6040_REG_HFLCTL, - TWL6040_REG_HFLGAIN, - TWL6040_REG_HFRCTL, - TWL6040_REG_HFRGAIN, -}; - /* set of rates for each pll: low-power and high-performance */ static unsigned int lp_rates[] = { 8000, @@ -335,17 +319,6 @@ static void twl6040_init_chip(struct snd_soc_codec *codec) twl6040_write_reg_cache(codec, TWL6040_REG_LINEGAIN, 0); } -static void twl6040_restore_regs(struct snd_soc_codec *codec) -{ - u8 *cache = codec->reg_cache; - int reg, i; - - for (i = 0; i < ARRAY_SIZE(twl6040_restore_list); i++) { - reg = twl6040_restore_list[i]; - twl6040_write(codec, reg, cache[reg]); - } -} - /* set headset dac and driver power mode */ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) { @@ -978,8 +951,6 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, priv->codec_powered = 1; - twl6040_restore_regs(codec); - /* Set external boost GPO */ twl6040_write(codec, TWL6040_REG_GPOCTL, 0x02); break; -- cgit v1.2.3-70-g09d2 From 626bcacb89f93b2145f3a705a342067a77347a99 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Nov 2013 16:03:47 +0200 Subject: ASoC: twl6040: Remove self managed local reg_cache support We can rely on mfd driver to manage the register caching via regmap. The driver still need to cache some registers associated with DL1/2 routes. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 211 +++++++++++---------------------------------- 1 file changed, 49 insertions(+), 162 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b07839b2d91..0afe8bef676 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -80,59 +80,6 @@ struct twl6040_data { struct mutex mutex; }; -/* - * twl6040 register cache & default register settings - */ -static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { - 0x00, /* not used 0x00 */ - 0x4B, /* REG_ASICID 0x01 (ro) */ - 0x00, /* REG_ASICREV 0x02 (ro) */ - 0x00, /* REG_INTID 0x03 */ - 0x00, /* REG_INTMR 0x04 */ - 0x00, /* REG_NCPCTRL 0x05 */ - 0x00, /* REG_LDOCTL 0x06 */ - 0x60, /* REG_HPPLLCTL 0x07 */ - 0x00, /* REG_LPPLLCTL 0x08 */ - 0x4A, /* REG_LPPLLDIV 0x09 */ - 0x00, /* REG_AMICBCTL 0x0A */ - 0x00, /* REG_DMICBCTL 0x0B */ - 0x00, /* REG_MICLCTL 0x0C */ - 0x00, /* REG_MICRCTL 0x0D */ - 0x00, /* REG_MICGAIN 0x0E */ - 0x1B, /* REG_LINEGAIN 0x0F */ - 0x00, /* REG_HSLCTL 0x10 */ - 0x00, /* REG_HSRCTL 0x11 */ - 0x00, /* REG_HSGAIN 0x12 */ - 0x00, /* REG_EARCTL 0x13 */ - 0x00, /* REG_HFLCTL 0x14 */ - 0x00, /* REG_HFLGAIN 0x15 */ - 0x00, /* REG_HFRCTL 0x16 */ - 0x00, /* REG_HFRGAIN 0x17 */ - 0x00, /* REG_VIBCTLL 0x18 */ - 0x00, /* REG_VIBDATL 0x19 */ - 0x00, /* REG_VIBCTLR 0x1A */ - 0x00, /* REG_VIBDATR 0x1B */ - 0x00, /* REG_HKCTL1 0x1C */ - 0x00, /* REG_HKCTL2 0x1D */ - 0x00, /* REG_GPOCTL 0x1E */ - 0x00, /* REG_ALB 0x1F */ - 0x00, /* REG_DLB 0x20 */ - 0x00, /* not used 0x21 */ - 0x00, /* not used 0x22 */ - 0x00, /* not used 0x23 */ - 0x00, /* not used 0x24 */ - 0x00, /* not used 0x25 */ - 0x00, /* not used 0x26 */ - 0x00, /* not used 0x27 */ - 0x00, /* REG_TRIM1 0x28 */ - 0x00, /* REG_TRIM2 0x29 */ - 0x00, /* REG_TRIM3 0x2A */ - 0x00, /* REG_HSOTRIM 0x2B */ - 0x00, /* REG_HFOTRIM 0x2C */ - 0x09, /* REG_ACCCTL 0x2D */ - 0x00, /* REG_STATUS 0x2E (ro) */ -}; - /* set of rates for each pll: low-power and high-performance */ static unsigned int lp_rates[] = { 8000, @@ -159,11 +106,14 @@ static struct snd_pcm_hw_constraint_list sysclk_constraints[] = { { .count = ARRAY_SIZE(hp_rates), .list = hp_rates, }, }; -static inline int twl6040_read_dl12_cache(struct snd_soc_codec *codec, - u8 reg, u8 *value) +static unsigned int twl6040_read(struct snd_soc_codec *codec, unsigned int reg) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - int ret = 0; + struct twl6040 *twl6040 = codec->control_data; + u8 value; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; switch (reg) { case TWL6040_REG_HSLCTL: @@ -171,36 +121,18 @@ static inline int twl6040_read_dl12_cache(struct snd_soc_codec *codec, case TWL6040_REG_EARCTL: case TWL6040_REG_HFLCTL: case TWL6040_REG_HFRCTL: - *value = priv->dl12_cache[reg - TWL6040_REG_HSLCTL]; + value = priv->dl12_cache[reg - TWL6040_REG_HSLCTL]; break; default: - ret = -EINVAL; + value = twl6040_reg_read(twl6040, reg); break; } - return ret; -} - -/* - * read twl6040 register cache - */ -static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u8 *cache = codec->reg_cache; - u8 value; - - if (reg >= TWL6040_CACHEREGNUM) - return -EIO; - - if (twl6040_read_dl12_cache(codec, reg, &value)) - value = cache[reg]; - return value; } -static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec, - u8 reg, u8 value) +static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec, + unsigned int reg) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); @@ -208,50 +140,18 @@ static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec, case TWL6040_REG_HSLCTL: case TWL6040_REG_HSRCTL: case TWL6040_REG_EARCTL: + /* DL1 path */ + return priv->dl1_unmuted; case TWL6040_REG_HFLCTL: case TWL6040_REG_HFRCTL: - priv->dl12_cache[reg - TWL6040_REG_HSLCTL] = value; - break; + return priv->dl2_unmuted; default: - break; + return 1; } } -/* - * write twl6040 register cache - */ -static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec, - u8 reg, u8 value) -{ - u8 *cache = codec->reg_cache; - - if (reg >= TWL6040_CACHEREGNUM) - return; - cache[reg] = value; - - twl6040_update_dl12_cache(codec, reg, value); -} - -/* - * read from twl6040 hardware register - */ -static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct twl6040 *twl6040 = codec->control_data; - u8 value; - - if (reg >= TWL6040_CACHEREGNUM) - return -EIO; - - value = twl6040_reg_read(twl6040, reg); - twl6040_write_reg_cache(codec, reg, value); - - return value; -} - -static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec, - unsigned int reg) +static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); @@ -259,19 +159,15 @@ static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec, case TWL6040_REG_HSLCTL: case TWL6040_REG_HSRCTL: case TWL6040_REG_EARCTL: - /* DL1 path */ - return priv->dl1_unmuted; case TWL6040_REG_HFLCTL: case TWL6040_REG_HFRCTL: - return priv->dl2_unmuted; + priv->dl12_cache[reg - TWL6040_REG_HSLCTL] = value; + break; default: - return 1; + break; } } -/* - * write to the twl6040 register space - */ static int twl6040_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -280,7 +176,7 @@ static int twl6040_write(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return -EIO; - twl6040_write_reg_cache(codec, reg, value); + twl6040_update_dl12_cache(codec, reg, value); if (twl6040_can_write_to_chip(codec, reg)) return twl6040_reg_write(twl6040, reg, value); else @@ -289,34 +185,27 @@ static int twl6040_write(struct snd_soc_codec *codec, static void twl6040_init_chip(struct snd_soc_codec *codec) { - struct twl6040 *twl6040 = codec->control_data; - u8 val; - - /* Update reg_cache: ASICREV, and TRIM values */ - val = twl6040_get_revid(twl6040); - twl6040_write_reg_cache(codec, TWL6040_REG_ASICREV, val); - - twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM1); - twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM2); - twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM3); - twl6040_read_reg_volatile(codec, TWL6040_REG_HSOTRIM); - twl6040_read_reg_volatile(codec, TWL6040_REG_HFOTRIM); + twl6040_read(codec, TWL6040_REG_TRIM1); + twl6040_read(codec, TWL6040_REG_TRIM2); + twl6040_read(codec, TWL6040_REG_TRIM3); + twl6040_read(codec, TWL6040_REG_HSOTRIM); + twl6040_read(codec, TWL6040_REG_HFOTRIM); /* Change chip defaults */ /* No imput selected for microphone amplifiers */ - twl6040_write_reg_cache(codec, TWL6040_REG_MICLCTL, 0x18); - twl6040_write_reg_cache(codec, TWL6040_REG_MICRCTL, 0x18); + twl6040_write(codec, TWL6040_REG_MICLCTL, 0x18); + twl6040_write(codec, TWL6040_REG_MICRCTL, 0x18); /* * We need to lower the default gain values, so the ramp code * can work correctly for the first playback. * This reduces the pop noise heard at the first playback. */ - twl6040_write_reg_cache(codec, TWL6040_REG_HSGAIN, 0xff); - twl6040_write_reg_cache(codec, TWL6040_REG_EARCTL, 0x1e); - twl6040_write_reg_cache(codec, TWL6040_REG_HFLGAIN, 0x1d); - twl6040_write_reg_cache(codec, TWL6040_REG_HFRGAIN, 0x1d); - twl6040_write_reg_cache(codec, TWL6040_REG_LINEGAIN, 0); + twl6040_write(codec, TWL6040_REG_HSGAIN, 0xff); + twl6040_write(codec, TWL6040_REG_EARCTL, 0x1e); + twl6040_write(codec, TWL6040_REG_HFLGAIN, 0x1d); + twl6040_write(codec, TWL6040_REG_HFRGAIN, 0x1d); + twl6040_write(codec, TWL6040_REG_LINEGAIN, 0); } /* set headset dac and driver power mode */ @@ -325,8 +214,8 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) int hslctl, hsrctl; int mask = TWL6040_HSDRVMODE | TWL6040_HSDACMODE; - hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); - hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + hslctl = twl6040_read(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read(codec, TWL6040_REG_HSRCTL); if (high_perf) { hslctl &= ~mask; @@ -353,8 +242,8 @@ static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, * Both HS DAC need to be turned on (before the HS driver) and off at * the same time. */ - hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); - hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + hslctl = twl6040_read(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read(codec, TWL6040_REG_HSRCTL); if (SND_SOC_DAPM_EVENT_ON(event)) { hslctl |= TWL6040_HSDACENA; hsrctl |= TWL6040_HSDACENA; @@ -399,7 +288,7 @@ static void twl6040_hs_jack_report(struct snd_soc_codec *codec, mutex_lock(&priv->mutex); /* Sync status */ - status = twl6040_read_reg_volatile(codec, TWL6040_REG_STATUS); + status = twl6040_read(codec, TWL6040_REG_STATUS); if (status & TWL6040_PLUGCOMP) snd_soc_jack_report(jack, report, report); else @@ -451,7 +340,7 @@ static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol, unsigned int val; /* Do not allow changes while Input/FF efect is running */ - val = twl6040_read_reg_volatile(codec, e->reg); + val = twl6040_read(codec, e->reg); if (val & TWL6040_VIBENA && !(val & TWL6040_VIBSEL)) return -EBUSY; @@ -676,7 +565,7 @@ int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim) if (unlikely(trim >= TWL6040_TRIM_INVAL)) return -EINVAL; - return twl6040_read_reg_cache(codec, TWL6040_REG_TRIM1 + trim); + return twl6040_read(codec, TWL6040_REG_TRIM1 + trim); } EXPORT_SYMBOL_GPL(twl6040_get_trim_value); @@ -1071,9 +960,9 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i switch (id) { case TWL6040_DAI_DL1: - hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); - hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); - earctl = twl6040_read_reg_cache(codec, TWL6040_REG_EARCTL); + hslctl = twl6040_read(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read(codec, TWL6040_REG_HSRCTL); + earctl = twl6040_read(codec, TWL6040_REG_EARCTL); if (mute) { /* Power down drivers and DACs */ @@ -1089,8 +978,8 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i priv->dl1_unmuted = !mute; break; case TWL6040_DAI_DL2: - hflctl = twl6040_read_reg_cache(codec, TWL6040_REG_HFLCTL); - hfrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HFRCTL); + hflctl = twl6040_read(codec, TWL6040_REG_HFLCTL); + hfrctl = twl6040_read(codec, TWL6040_REG_HFRCTL); if (mute) { /* Power down drivers and DACs */ @@ -1227,6 +1116,7 @@ static int twl6040_resume(struct snd_soc_codec *codec) static int twl6040_probe(struct snd_soc_codec *codec) { struct twl6040_data *priv; + struct twl6040 *twl6040 = dev_get_drvdata(codec->dev->parent); struct platform_device *pdev = container_of(codec->dev, struct platform_device, dev); int ret = 0; @@ -1238,7 +1128,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, priv); priv->codec = codec; - codec->control_data = dev_get_drvdata(codec->dev->parent); + codec->control_data = twl6040; priv->plug_irq = platform_get_irq(pdev, 0); if (priv->plug_irq < 0) { @@ -1258,10 +1148,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) return ret; } + twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); twl6040_init_chip(codec); - /* power on device */ - return twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; } static int twl6040_remove(struct snd_soc_codec *codec) @@ -1279,12 +1169,9 @@ static struct snd_soc_codec_driver soc_codec_dev_twl6040 = { .remove = twl6040_remove, .suspend = twl6040_suspend, .resume = twl6040_resume, - .read = twl6040_read_reg_cache, + .read = twl6040_read, .write = twl6040_write, .set_bias_level = twl6040_set_bias_level, - .reg_cache_size = ARRAY_SIZE(twl6040_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = twl6040_reg, .ignore_pmdown_time = true, .controls = twl6040_snd_controls, -- cgit v1.2.3-70-g09d2 From f467a0f513ad81998f0cad1022684a273d5743f7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:08 +0100 Subject: ASoC: au1x: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Tested-by: Manuel Lauss Signed-off-by: Mark Brown --- sound/soc/au1x/dbdma2.c | 9 --------- sound/soc/au1x/dma.c | 14 -------------- 2 files changed, 23 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 3b4eafaf30d..17a24d80473 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -65,19 +65,10 @@ struct au1xpsc_audio_dmadata { #define AU1XPSC_PERIOD_MIN_BYTES 1024 #define AU1XPSC_BUFFER_MIN_BYTES 65536 -#define AU1XPSC_PCM_FMTS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ - SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ - SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \ - SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \ - SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \ - 0) - /* PCM hardware DMA capabilities - platform specific */ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, - .formats = AU1XPSC_PCM_FMTS, .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, .period_bytes_max = 4096 * 1024 - 1, .periods_min = 2, diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index befd1074f9b..e920b60bf6c 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -21,14 +21,6 @@ #include "psc.h" -#define ALCHEMY_PCM_FMTS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ - SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ - SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \ - SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \ - SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \ - 0) - struct pcm_period { u32 start; u32 relative_end; /* relative to start of buffer */ @@ -171,12 +163,6 @@ static irqreturn_t au1000_dma_interrupt(int irq, void *ptr) static const struct snd_pcm_hardware alchemy_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, - .formats = ALCHEMY_PCM_FMTS, - .rates = SNDRV_PCM_RATE_8000_192000, - .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_192000, - .channels_min = 2, - .channels_max = 2, .period_bytes_min = 1024, .period_bytes_max = 16 * 1024 - 1, .periods_min = 4, -- cgit v1.2.3-70-g09d2 From 5e8154332f48f92f37824577c88e400b5e0cd56d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:22 +0100 Subject: ASoC: tegra: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Tested-by: Stephen Warren Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_pcm.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 7b2d23ba69b..c09ffd18791 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -42,9 +42,6 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .channels_min = 2, - .channels_max = 2, .period_bytes_min = 1024, .period_bytes_max = PAGE_SIZE, .periods_min = 2, -- cgit v1.2.3-70-g09d2 From a6af47ae5399baf4f5a2426b2121c1bcb9da4019 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Fri, 20 Dec 2013 12:17:38 +0800 Subject: ASoC: fsl-sai: Remove fsl_sai_remove() There is no need of this function and makes the code slightly shorter Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 11 ----------- 1 file changed, 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 50a797e6578..1868ec34be1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -385,19 +385,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } -static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai) -{ - cpu_dai->playback_dma_data = NULL; - cpu_dai->capture_dma_data = NULL; - - snd_soc_dai_set_drvdata(cpu_dai, NULL); - - return 0; -} - static struct snd_soc_dai_driver fsl_sai_dai = { .probe = fsl_sai_dai_probe, - .remove = fsl_sai_dai_remove, .playback = { .channels_min = 1, .channels_max = 2, -- cgit v1.2.3-70-g09d2 From e5180df3960b6130f17f3c5ab50d23674cdb2b5a Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Fri, 20 Dec 2013 12:30:26 +0800 Subject: ASoC: fsl-sai: Use devm_snd_dmaengine_pcm_register() Makes the code slightly shorter Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 15 +-------------- 1 file changed, 1 insertion(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 1868ec34be1..262d3107892 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -443,19 +443,8 @@ static int fsl_sai_probe(struct platform_device *pdev) if (ret) return ret; - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); - if (ret) - return ret; - - return 0; -} - -static int fsl_sai_remove(struct platform_device *pdev) -{ - snd_dmaengine_pcm_unregister(&pdev->dev); - - return 0; } static const struct of_device_id fsl_sai_ids[] = { @@ -465,8 +454,6 @@ static const struct of_device_id fsl_sai_ids[] = { static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, - .remove = fsl_sai_remove, - .driver = { .name = "fsl-sai", .owner = THIS_MODULE, -- cgit v1.2.3-70-g09d2 From dd9f40602e96353c210805a99abd9af6abd28473 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Fri, 20 Dec 2013 12:35:33 +0800 Subject: ASoC: fsl-sai: Use snd_soc_dai_init_dma_data() Makes the code slightly shorter Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 262d3107892..b8cdbf8660f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -377,8 +377,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - cpu_dai->playback_dma_data = &sai->dma_params_tx; - cpu_dai->capture_dma_data = &sai->dma_params_rx; + snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, + &sai->dma_params_rx); snd_soc_dai_set_drvdata(cpu_dai, sai); -- cgit v1.2.3-70-g09d2 From dd41e0c4e35abc5a5679762ab108a6fcd9a0ead9 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Fri, 20 Dec 2013 14:39:50 +0800 Subject: ASoC: simple-card: Add cpu_dai and codec_dai names NULL check The name of cpu DAI maybe omitted, and then strlen() will lead kernel panic. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7a9b6b4898c..be7c1db5388 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -142,6 +142,9 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (ret < 0) return ret; + if (!info->cpu_dai.name || !info->codec_dai.name) + return -EINVAL; + /* card name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, strlen(info->cpu_dai.name) + -- cgit v1.2.3-70-g09d2 From 1fb2d9d7465bcbb519c582fa4a3bd04ff4fce2d2 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:00 +0800 Subject: ASoC: fsl_sai: Keep symmetry for clk_enable() and clk_disable() There are two functions haven't clk_disable_unprepare() if having error. Thus fix them. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index b8cdbf8660f..69a375f48ef 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -111,7 +111,7 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, dev_err(cpu_dai->dev, "Cannot set SAI's transmitter sysclk: %d\n", ret); - return ret; + goto err_clk; } ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, @@ -120,12 +120,13 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, dev_err(cpu_dai->dev, "Cannot set SAI's receiver sysclk: %d\n", ret); - return ret; + goto err_clk; } +err_clk: clk_disable_unprepare(sai->clk); - return 0; + return ret; } static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, @@ -222,7 +223,7 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) dev_err(cpu_dai->dev, "Cannot set SAI's transmitter format: %d\n", ret); - return ret; + goto err_clk; } ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER); @@ -230,12 +231,13 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) dev_err(cpu_dai->dev, "Cannot set SAI's receiver format: %d\n", ret); - return ret; + goto err_clk; } +err_clk: clk_disable_unprepare(sai->clk); - return 0; + return ret; } static int fsl_sai_hw_params(struct snd_pcm_substream *substream, -- cgit v1.2.3-70-g09d2 From 1d7003092771bd2feec30e2f3e5a06aa33479e08 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:01 +0800 Subject: ASoC: fsl_sai: Use snd_pcm_format_width() Use common helper function snd_pcm_format_width() to make code neater. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 17 ++--------------- 1 file changed, 2 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 69a375f48ef..e68102e6352 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -244,9 +244,10 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr, word_width; + u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr; unsigned int channels = params_channels(params); struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 word_width = snd_pcm_format_width(params_format(params)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { reg_cr4 = FSL_SAI_TCR4; @@ -267,20 +268,6 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - word_width = 16; - break; - case SNDRV_PCM_FORMAT_S20_3LE: - word_width = 20; - break; - case SNDRV_PCM_FORMAT_S24_LE: - word_width = 24; - break; - default: - return -EINVAL; - } - val_cr4 |= FSL_SAI_CR4_SYWD(word_width); val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); -- cgit v1.2.3-70-g09d2 From d22e28cce80a93578787d273bf1fa26a2be2636b Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:02 +0800 Subject: ASoC: fsl_sai: Drop useless channels check in hw_params() SAi only supports two data channels on hardware level and the driver also does register the min->1 and max->2, so no need to check channels. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index e68102e6352..8450bff6fb1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -278,10 +278,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_FBT(0); val_cr4 |= FSL_SAI_CR4_FRSZ(channels); - if (channels == 2 || channels == 1) - val_mr = ~0UL - ((1 << channels) - 1); - else - return -EINVAL; + val_mr = ~0UL - ((1 << channels) - 1); sai_writel(sai, val_cr4, sai->base + reg_cr4); sai_writel(sai, val_cr5, sai->base + reg_cr5); -- cgit v1.2.3-70-g09d2 From 15b29dae6604d2d2daf586429ff12f26272a868a Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:03 +0800 Subject: ASoC: fsl_sai: Drop useless ret in startup() We can save this ret to make the code neater. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 8450bff6fb1..fc4cd95ad0a 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -334,12 +334,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { - int ret; struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - ret = clk_prepare_enable(sai->clk); - - return ret; + return clk_prepare_enable(sai->clk); } static void fsl_sai_shutdown(struct snd_pcm_substream *substream, -- cgit v1.2.3-70-g09d2 From 190af12dad975f2ea7d69d1c5c9d36fec64da767 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:04 +0800 Subject: ASoC: fsl_sai: Make dev_err information neater Since using dev_err() there's no need to mention SAI any more, it will print the full name of the driver -- fsl_sai. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fc4cd95ad0a..68d666b491d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -108,18 +108,14 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_TRANSMITTER); if (ret) { - dev_err(cpu_dai->dev, - "Cannot set SAI's transmitter sysclk: %d\n", - ret); + dev_err(cpu_dai->dev, "Cannot set tx sysclk: %d\n", ret); goto err_clk; } ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_RECEIVER); if (ret) { - dev_err(cpu_dai->dev, - "Cannot set SAI's receiver sysclk: %d\n", - ret); + dev_err(cpu_dai->dev, "Cannot set rx sysclk: %d\n", ret); goto err_clk; } @@ -220,17 +216,13 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER); if (ret) { - dev_err(cpu_dai->dev, - "Cannot set SAI's transmitter format: %d\n", - ret); + dev_err(cpu_dai->dev, "Cannot set tx format: %d\n", ret); goto err_clk; } ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER); if (ret) { - dev_err(cpu_dai->dev, - "Cannot set SAI's receiver format: %d\n", - ret); + dev_err(cpu_dai->dev, "Cannot set rx format: %d\n", ret); goto err_clk; } -- cgit v1.2.3-70-g09d2 From 4e3a99f5b004b30bc604d82e5498700649148e0d Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 20 Dec 2013 16:41:05 +0800 Subject: ASoC: fsl_sai: Sort local variable in general way Generally we would write code for local variable like: static new_func() { struct xxx *yyy; ... int ret; } But this driver only follows this pattern for some functions, not all. Thus this patch sorts the local variable in the general way. Signed-off-by: Nicolin Chen Reviewed-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 68d666b491d..b72132fa70d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -53,8 +53,8 @@ static inline void sai_writel(struct fsl_sai *sai, static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { - u32 val_cr2, reg_cr2; struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 val_cr2, reg_cr2; if (fsl_dir == FSL_FMT_TRANSMITTER) reg_cr2 = FSL_SAI_TCR2; @@ -90,8 +90,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - int ret; struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + int ret; if (dir == SND_SOC_CLOCK_IN) return 0; @@ -128,8 +128,8 @@ err_clk: static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, unsigned int fmt, int fsl_dir) { - u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4; struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4; if (fsl_dir == FSL_FMT_TRANSMITTER) { reg_cr2 = FSL_SAI_TCR2; @@ -207,8 +207,8 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - int ret; struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + int ret; ret = clk_prepare_enable(sai->clk); if (ret) @@ -236,9 +236,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr; unsigned int channels = params_channels(params); - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); u32 word_width = snd_pcm_format_width(params_format(params)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -383,10 +383,10 @@ static const struct snd_soc_component_driver fsl_component = { static int fsl_sai_probe(struct platform_device *pdev) { - int ret; + struct device_node *np = pdev->dev.of_node; struct fsl_sai *sai; struct resource *res; - struct device_node *np = pdev->dev.of_node; + int ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) -- cgit v1.2.3-70-g09d2 From d754fa9ad18d16209c276fc6241aa2d10f819ede Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:09 +0100 Subject: ASoC: blackfin: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.c | 3 --- 2 files changed, 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 1d4c676eb6c..cdb8ee75ded 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -107,7 +107,6 @@ static const struct snd_pcm_hardware bf5xx_pcm_hardware = { #endif SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = SNDRV_PCM_FMTBIT_S16_LE, .period_bytes_min = 32, .period_bytes_max = 0x10000, .periods_min = 1, diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 2a5b43417fd..a3881c4381c 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -52,9 +52,6 @@ static const struct snd_pcm_hardware bf5xx_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 0x10000, .periods_min = 1, -- cgit v1.2.3-70-g09d2 From f52c91921553be26c7a0de13daa0d18ef46655ff Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:10 +0100 Subject: ASoC: davinci: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-pcm.c | 28 ---------------------------- 1 file changed, 28 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index fb5d107f560..14145cdf8a1 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -46,33 +46,11 @@ static void print_buf_info(int slot, char *name) } #endif -#define DAVINCI_PCM_FMTBITS (\ - SNDRV_PCM_FMTBIT_S8 |\ - SNDRV_PCM_FMTBIT_U8 |\ - SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S16_BE |\ - SNDRV_PCM_FMTBIT_U16_LE |\ - SNDRV_PCM_FMTBIT_U16_BE |\ - SNDRV_PCM_FMTBIT_S24_LE |\ - SNDRV_PCM_FMTBIT_S24_BE |\ - SNDRV_PCM_FMTBIT_U24_LE |\ - SNDRV_PCM_FMTBIT_U24_BE |\ - SNDRV_PCM_FMTBIT_S32_LE |\ - SNDRV_PCM_FMTBIT_S32_BE |\ - SNDRV_PCM_FMTBIT_U32_LE |\ - SNDRV_PCM_FMTBIT_U32_BE) - static struct snd_pcm_hardware pcm_hardware_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME| SNDRV_PCM_INFO_BATCH), - .formats = DAVINCI_PCM_FMTBITS, - .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 384, .buffer_bytes_max = 128 * 1024, .period_bytes_min = 32, .period_bytes_max = 8 * 1024, @@ -86,12 +64,6 @@ static struct snd_pcm_hardware pcm_hardware_capture = { SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_BATCH), - .formats = DAVINCI_PCM_FMTBITS, - .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 384, .buffer_bytes_max = 128 * 1024, .period_bytes_min = 32, .period_bytes_max = 8 * 1024, -- cgit v1.2.3-70-g09d2 From 3cec159cfb3fc69190f3ccdc2d1329c66775529f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:19 +0100 Subject: ASoC: s6000: Don't set unused struct snd_pcm_hardware fields MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Acked-by: Daniel Glöckner Signed-off-by: Mark Brown --- sound/soc/s6000/s6000-pcm.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index d219880815c..fb8461e1b1f 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -33,13 +33,6 @@ static struct snd_pcm_hardware s6000_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_JOINT_DUPLEX), - .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), - .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_192000), - .rate_min = 0, - .rate_max = 1562500, - .channels_min = 2, - .channels_max = 8, .buffer_bytes_max = 0x7ffffff0, .period_bytes_min = 16, .period_bytes_max = 0xfffff0, -- cgit v1.2.3-70-g09d2 From 60e21d2873440fc005c88970960bed678138217e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:17 +0100 Subject: ASoC: omap: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b8fa9862e54..07b8b7bc9d2 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -45,8 +45,6 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 64 * 1024, .periods_min = 2, -- cgit v1.2.3-70-g09d2 From e6dc12d7198eddba2e3e7a13feab5c7edde7ba1d Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Wed, 25 Dec 2013 11:20:14 +0800 Subject: ASoC: fsl_sai: Move the global registers setting to _dai_probe() Because we cannot make sure which one of _dai_fmt() and _dai_sysclk() will be firstly called. So move the RCSR/TCSR and TCR1/RCR1's initialization to _dai_probe(), and this can make sure that before any of {T,R}CR{1~5} register to be set the RCSR/TCSR's RE/TE bit has been cleared for the hareware limitation. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index b72132fa70d..596aabbf903 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -100,11 +100,6 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, if (ret) return ret; - sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); - sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); - sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); - sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); - ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_TRANSMITTER); if (ret) { @@ -351,6 +346,18 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); + int ret; + + ret = clk_prepare_enable(sai->clk); + if (ret) + return ret; + + sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); + sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); + sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); + sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); + + clk_disable_unprepare(sai->clk); snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, &sai->dma_params_rx); -- cgit v1.2.3-70-g09d2 From 61c66c60c75bd7f9650e27fb0a017bc39f2629a2 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 26 Dec 2013 11:41:29 +0530 Subject: ASoC: samsung: Trivial cleanups in header files commit 5d229ce569be ("ASoC: samsung: move plat/ headers to local directory") moved the header files but forgot to clean the pointers to their old locaton. Remove them now. Signed-off-by: Sachin Kamat Reviewed-by: Jingoo Han Signed-off-by: Mark Brown --- sound/soc/samsung/regs-ac97.h | 9 ++++----- sound/soc/samsung/regs-iis.h | 9 ++++----- 2 files changed, 8 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/regs-ac97.h b/sound/soc/samsung/regs-ac97.h index c3878f7acb8..a71be45bbff 100644 --- a/sound/soc/samsung/regs-ac97.h +++ b/sound/soc/samsung/regs-ac97.h @@ -1,5 +1,4 @@ -/* arch/arm/mach-s3c2410/include/mach/regs-ac97.h - * +/* * Copyright (c) 2006 Simtec Electronics * http://www.simtec.co.uk/products/SWLINUX/ * @@ -10,8 +9,8 @@ * S3C2440 AC97 Controller */ -#ifndef __ASM_ARCH_REGS_AC97_H -#define __ASM_ARCH_REGS_AC97_H __FILE__ +#ifndef __SAMSUNG_REGS_AC97_H__ +#define __SAMSUNG_REGS_AC97_H__ #define S3C_AC97_GLBCTRL (0x00) @@ -64,4 +63,4 @@ #define S3C_AC97_PCM_DATA (0x18) #define S3C_AC97_MIC_DATA (0x1C) -#endif /* __ASM_ARCH_REGS_AC97_H */ +#endif /* __SAMSUNG_REGS_AC97_H__ */ diff --git a/sound/soc/samsung/regs-iis.h b/sound/soc/samsung/regs-iis.h index a18d35e7a73..dc6cbbe9c4f 100644 --- a/sound/soc/samsung/regs-iis.h +++ b/sound/soc/samsung/regs-iis.h @@ -1,5 +1,4 @@ -/* arch/arm/plat-samsung/include/plat/regs-iis.h - * +/* * Copyright (c) 2003 Simtec Electronics * http://www.simtec.co.uk/products/SWLINUX/ * @@ -10,8 +9,8 @@ * S3C2410 IIS register definition */ -#ifndef __ASM_ARCH_REGS_IIS_H -#define __ASM_ARCH_REGS_IIS_H +#ifndef __SAMSUNG_REGS_IIS_H__ +#define __SAMSUNG_REGS_IIS_H__ #define S3C2410_IISCON (0x00) @@ -67,4 +66,4 @@ #define S3C2410_IISFIFO (0x10) -#endif /* __ASM_ARCH_REGS_IIS_H */ +#endif /* __SAMSUNG_REGS_IIS_H__ */ -- cgit v1.2.3-70-g09d2 From e5d0fa9c3ec59a40e0285d96b65b7f62875acd42 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Wed, 25 Dec 2013 12:40:04 +0800 Subject: ASoC: fsl_sai: Add disable operation for the corresponding data channel. Enables/Disables the corresponding data channel for tx/rx operation. A channel must be enabled before its FIFO is accessed, and then disable it when tx/rx is stopped or idle. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 596aabbf903..af802465456 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -124,20 +124,17 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, unsigned int fmt, int fsl_dir) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4; + u32 val_cr2, val_cr4, reg_cr2, reg_cr4; if (fsl_dir == FSL_FMT_TRANSMITTER) { reg_cr2 = FSL_SAI_TCR2; - reg_cr3 = FSL_SAI_TCR3; reg_cr4 = FSL_SAI_TCR4; } else { reg_cr2 = FSL_SAI_RCR2; - reg_cr3 = FSL_SAI_RCR3; reg_cr4 = FSL_SAI_RCR4; } val_cr2 = sai_readl(sai, sai->base + reg_cr2); - val_cr3 = sai_readl(sai, sai->base + reg_cr3); val_cr4 = sai_readl(sai, sai->base + reg_cr4); if (sai->big_endian_data) @@ -188,13 +185,10 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - val_cr3 |= FSL_SAI_CR3_TRCE; - if (fsl_dir == FSL_FMT_RECEIVER) val_cr2 |= FSL_SAI_CR2_SYNC; sai_writel(sai, val_cr2, sai->base + reg_cr2); - sai_writel(sai, val_cr3, sai->base + reg_cr3); sai_writel(sai, val_cr4, sai->base + reg_cr4); return 0; @@ -278,7 +272,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - unsigned int tcsr, rcsr; + u32 tcsr, rcsr, val_cr3, reg_cr3; tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); @@ -286,17 +280,24 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { tcsr |= FSL_SAI_CSR_FRDE; rcsr &= ~FSL_SAI_CSR_FRDE; + reg_cr3 = FSL_SAI_TCR3; } else { rcsr |= FSL_SAI_CSR_FRDE; tcsr &= ~FSL_SAI_CSR_FRDE; + reg_cr3 = FSL_SAI_RCR3; } + val_cr3 = sai_readl(sai, sai->base + reg_cr3); + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: tcsr |= FSL_SAI_CSR_TERE; rcsr |= FSL_SAI_CSR_TERE; + val_cr3 |= FSL_SAI_CR3_TRCE; + + sai_writel(sai, val_cr3, sai->base + reg_cr3); sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); break; @@ -308,8 +309,12 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, tcsr &= ~FSL_SAI_CSR_TERE; rcsr &= ~FSL_SAI_CSR_TERE; } + + val_cr3 &= ~FSL_SAI_CR3_TRCE; + sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); + sai_writel(sai, val_cr3, sai->base + reg_cr3); break; default: return -EINVAL; -- cgit v1.2.3-70-g09d2 From 95ab1297e7b1a57c2b777f210b74f3a2bd2ac269 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:25 +0100 Subject: ASoC: soc-utils: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 5e633659c1b..6ebdfd9a1a1 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -59,10 +59,6 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params) EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); static const struct snd_pcm_hardware dummy_dma_hardware = { - .formats = 0xffffffff, - .channels_min = 1, - .channels_max = UINT_MAX, - /* Random values to keep userspace happy when checking constraints */ .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, -- cgit v1.2.3-70-g09d2 From 328089a47112a4fc06071e2003ecd75cc6d31029 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:20 +0100 Subject: ASoC: samsung: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/dma.c | 6 ------ sound/soc/samsung/idma.c | 8 -------- 2 files changed, 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index ee23194f7ab..dc09b71b7d9 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -35,12 +35,6 @@ static const struct snd_pcm_hardware dma_hardware = { SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, .buffer_bytes_max = 128*1024, .period_bytes_min = PAGE_SIZE, .period_bytes_max = PAGE_SIZE*2, diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index e4f318fc2f8..3d5cf1530b6 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -35,14 +35,6 @@ static const struct snd_pcm_hardware idma_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_U24_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, .buffer_bytes_max = MAX_IDMA_BUFFER, .period_bytes_min = 128, .period_bytes_max = MAX_IDMA_PERIOD, -- cgit v1.2.3-70-g09d2 From 38136bde7691bdafa91c2320e014913aec6dbe6b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:23 +0100 Subject: ASoC: txx9: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/txx9/txx9aclc.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index fbd077f4de7..f0829de2870 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -40,11 +40,6 @@ static const struct snd_pcm_hardware txx9aclc_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_PAUSE, -#ifdef __BIG_ENDIAN - .formats = SNDRV_PCM_FMTBIT_S16_BE, -#else - .formats = SNDRV_PCM_FMTBIT_S16_LE, -#endif .period_bytes_min = 1024, .period_bytes_max = 8 * 1024, .periods_min = 2, -- cgit v1.2.3-70-g09d2 From 323702b4e06dfd1a4ee6cee5834a889b9663cccf Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:18 +0100 Subject: ASoC: mmp: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/mmp-pcm.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 682ee52942b..5e8d8133017 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -36,14 +36,9 @@ struct mmp_dma_data { SNDRV_PCM_INFO_PAUSE | \ SNDRV_PCM_INFO_RESUME) -#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) - static struct snd_pcm_hardware mmp_pcm_hardware[] = { { .info = MMP_PCM_INFO, - .formats = MMP_PCM_FORMATS, .period_bytes_min = 1024, .period_bytes_max = 2048, .periods_min = 2, @@ -53,7 +48,6 @@ static struct snd_pcm_hardware mmp_pcm_hardware[] = { }, { .info = MMP_PCM_INFO, - .formats = MMP_PCM_FORMATS, .period_bytes_min = 1024, .period_bytes_max = 2048, .periods_min = 2, -- cgit v1.2.3-70-g09d2 From 111bd7b18e13c66552e9a672000eeacacd414a65 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:11 +0100 Subject: ASoC: ep93xx: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-pcm.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index ca6698d7573..5f664471d99 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -30,15 +30,6 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), - - .rates = SNDRV_PCM_RATE_8000_192000, - .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_192000, - - .formats = (SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE), - .buffer_bytes_max = 131072, .period_bytes_min = 32, .period_bytes_max = 32768, -- cgit v1.2.3-70-g09d2 From 496a39d9ec238569fac6daceac8f5420c5edc2f1 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 31 Dec 2013 15:33:21 +0800 Subject: ASoC: fsl_sai: Fix one bug for hardware limitation. This is maybe one bug or a limitation of the hardware that the {T,R}CR2's Synchronous Mode bits must be set as late as possible, or the SAI device maybe hanged up, and there has not any explaination about this limitation in the SAI Data Sheet. And the {T,R}CR2's Synchronous Mode bits must be set at the same time whether for Tx or Rx stream. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index af802465456..2ece14716c6 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -145,7 +145,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: val_cr4 |= FSL_SAI_CR4_FSE; - val_cr4 |= FSL_SAI_CR4_FSP; break; default: return -EINVAL; @@ -185,9 +184,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - if (fsl_dir == FSL_FMT_RECEIVER) - val_cr2 |= FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + reg_cr2); sai_writel(sai, val_cr4, sai->base + reg_cr4); @@ -253,6 +249,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); + val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; if (sai->big_endian_data) val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); else @@ -272,7 +269,15 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 tcsr, rcsr, val_cr3, reg_cr3; + u32 tcsr, rcsr, val_cr2, val_cr3, reg_cr3; + + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2); + val_cr2 &= ~FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2); + + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2); + val_cr2 |= FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2); tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); -- cgit v1.2.3-70-g09d2 From 72aa62bed3ea30635156fad95f123a0b665072bf Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 31 Dec 2013 15:33:22 +0800 Subject: ASoC: fsl_sai: fix the endianess for SAI fifo data. Revert the SAI's endianess for fifo data to/from DMA engine. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 2ece14716c6..5d38a6749b9 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -138,9 +138,9 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 = sai_readl(sai, sai->base + reg_cr4); if (sai->big_endian_data) - val_cr4 |= FSL_SAI_CR4_MF; - else val_cr4 &= ~FSL_SAI_CR4_MF; + else + val_cr4 |= FSL_SAI_CR4_MF; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -251,9 +251,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; if (sai->big_endian_data) - val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); - else val_cr5 |= FSL_SAI_CR5_FBT(0); + else + val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); val_cr4 |= FSL_SAI_CR4_FRSZ(channels); val_mr = ~0UL - ((1 << channels) - 1); -- cgit v1.2.3-70-g09d2 From 192043cf608909eb5728a5fd68f5234b90d9415b Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 27 Nov 2013 18:05:10 +0800 Subject: ASoC: atmel: sam9x5_wm8731: remove platform_set_drvdata When call snd_soc_register_card, it will set driver data to this device through dev_set_drvdata, then in driver, no need to call platform_set_drvdata again, so remove it. Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/sam9x5_wm8731.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 992ae38d5a1..6f4e812d6e6 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -153,8 +153,6 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) of_node_put(codec_np); of_node_put(cpu_np); - platform_set_drvdata(pdev, card); - ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, -- cgit v1.2.3-70-g09d2 From d4c22094b256a7327346738721b89a78d4680b08 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 23 Dec 2013 12:57:01 +0800 Subject: ASoC: simple-card: Add DAPM routes parse from device tree Parses a simple DAPM route table from device tree. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/simple-card.txt | 13 +++++++++++-- sound/soc/generic/simple-card.c | 11 +++++++++-- 2 files changed, 20 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 769a346f890..2ee80c76ca6 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -9,8 +9,13 @@ Required properties: Optional properties: - simple-audio-card,format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" +- simple-audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. + Required subnodes: - simple-audio-card,cpu : CPU sub-node @@ -38,6 +43,10 @@ Example: sound { compatible = "simple-audio-card"; simple-audio-card,format = "left_j"; + simple-audio-routing = + "MIC_IN", "Mic Jack", + "Headphone Jack", "HP_OUT", + "Ext Spk", "LINE_OUT"; simple-audio-card,cpu { sound-dai = <&sh_fsi2 0>; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3d190d05ad4..6230efb05fd 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -116,12 +116,18 @@ static int asoc_simple_card_parse_of(struct device_node *node, { struct device_node *np; char *name; - int ret = 0; + int ret; /* get CPU/CODEC common format via simple-audio-card,format */ info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); + /* DAPM routes */ + ret = snd_soc_of_parse_audio_routing(&info->snd_card, + "simple-audio-routing"); + if (ret) + return ret; + /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); @@ -182,6 +188,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); if (cinfo) { int ret; + cinfo->snd_card.dev = &pdev->dev; ret = asoc_simple_card_parse_of(np, cinfo, dev, &of_cpu, &of_codec, @@ -193,6 +200,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } } } else { + cinfo->snd_card.dev = &pdev->dev; cinfo = pdev->dev.platform_data; } @@ -232,7 +240,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; - cinfo->snd_card.dev = &pdev->dev; return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); } -- cgit v1.2.3-70-g09d2 From e874ddead38996ec40c6a6be2347a69fac520126 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 23 Dec 2013 13:24:59 +0800 Subject: ASoC: simple-card: Cleanup __asoc_simple_card_dai_init() ret check The ret parameter is always equal to zero till here. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6230efb05fd..3ba65bb6343 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -25,7 +25,7 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, daifmt |= set->fmt; - if (!ret && daifmt) + if (daifmt) ret = snd_soc_dai_set_fmt(dai, daifmt); if (ret == -ENOTSUPP) { -- cgit v1.2.3-70-g09d2 From 0f7f3d1f17c2e4d73e449e6acb2007b13813c58e Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 20 Dec 2013 12:40:16 +0200 Subject: ASoC: hdmi-codec: Add devicetree binding with documentation Signed-off-by: Jyri Sarha cc: bcousson@baylibre.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/hdmi.txt | 17 +++++++++++++++++ sound/soc/codecs/hdmi.c | 10 ++++++++++ 2 files changed, 27 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/hdmi.txt (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/hdmi.txt b/Documentation/devicetree/bindings/sound/hdmi.txt new file mode 100644 index 00000000000..31af7bca309 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/hdmi.txt @@ -0,0 +1,17 @@ +Device-Tree bindings for dummy HDMI codec + +Required properties: + - compatible: should be "linux,hdmi-audio". + +CODEC output pins: + * TX + +CODEC input pins: + * RX + +Example node: + + hdmi_audio: hdmi_audio@0 { + compatible = "linux,hdmi-audio"; + status = "okay"; + }; diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index 32797a8e4ee..9cb1c7d3e1d 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -20,6 +20,7 @@ */ #include #include +#include #define DRV_NAME "hdmi-audio-codec" @@ -60,6 +61,14 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { }; +#ifdef CONFIG_OF +static const struct of_device_id hdmi_audio_codec_ids[] = { + { .compatible = "linux,hdmi-audio", }, + { } +}; +MODULE_DEVICE_TABLE(of, hdmi_audio_codec_ids); +#endif + static struct snd_soc_codec_driver hdmi_codec = { .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), @@ -83,6 +92,7 @@ static struct platform_driver hdmi_codec_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(hdmi_audio_codec_ids), }, .probe = hdmi_codec_probe, -- cgit v1.2.3-70-g09d2 From e337853ebb46d012c069ca47ba3ce9f4744305ea Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:26:31 -0800 Subject: ASoC: rsnd: add rsnd_adg_set_ssi_clk() and cleanup adg This patch adds rsnd_adg_set_ssi_clk() to access to AUDIO_CLK_SEL0/1/2, and removes last user of rsnd_write/read/bset which is very low level function. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 58 ++++++++++++++++++++----------------------------- 1 file changed, 24 insertions(+), 34 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 9430097979a..55d0394955b 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -30,41 +30,41 @@ struct rsnd_adg { i++, (pos) = adg->clk[i]) #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) -static enum rsnd_reg rsnd_adg_ssi_reg_get(int id) +static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val) { - enum rsnd_reg reg; + int id = rsnd_mod_id(mod); + int shift = (id % 4) * 8; + u32 mask = 0xFF << shift; + + val = val << shift; /* * SSI 8 is not connected to ADG. * it works with SSI 7 */ if (id == 8) - return RSND_REG_MAX; - - if (0 <= id && id <= 3) - reg = RSND_REG_AUDIO_CLK_SEL0; - else if (4 <= id && id <= 7) - reg = RSND_REG_AUDIO_CLK_SEL1; - else - reg = RSND_REG_AUDIO_CLK_SEL2; - - return reg; + return; + + switch (id / 4) { + case 0: + rsnd_mod_bset(mod, AUDIO_CLK_SEL0, mask, val); + break; + case 1: + rsnd_mod_bset(mod, AUDIO_CLK_SEL1, mask, val); + break; + case 2: + rsnd_mod_bset(mod, AUDIO_CLK_SEL2, mask, val); + break; + } } int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod) { - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - enum rsnd_reg reg; - int id; - /* * "mod" = "ssi" here. * we can get "ssi id" from mod */ - id = rsnd_mod_id(mod); - reg = rsnd_adg_ssi_reg_get(id); - - rsnd_write(priv, mod, reg, 0); + rsnd_adg_set_ssi_clk(mod, 0); return 0; } @@ -75,8 +75,7 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate) struct rsnd_adg *adg = rsnd_priv_to_adg(priv); struct device *dev = rsnd_priv_to_dev(priv); struct clk *clk; - enum rsnd_reg reg; - int id, shift, i; + int i; u32 data; int sel_table[] = { [CLKA] = 0x1, @@ -125,19 +124,10 @@ found_clock: * This "mod" = "ssi" here. * we can get "ssi id" from mod */ - id = rsnd_mod_id(mod); - reg = rsnd_adg_ssi_reg_get(id); - - dev_dbg(dev, "ADG: ssi%d selects clk%d = %d", id, i, rate); - - /* - * Enable SSIx clock - */ - shift = (id % 4) * 8; + rsnd_adg_set_ssi_clk(mod, data); - rsnd_bset(priv, mod, reg, - 0xFF << shift, - data << shift); + dev_dbg(dev, "ADG: ssi%d selects clk%d = %d", + rsnd_mod_id(mod), i, rate); return 0; } -- cgit v1.2.3-70-g09d2 From 729aca51a19f2e2b3404c29b82df61d714150a49 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:26:44 -0800 Subject: ASoC: rsnd: tidyup ssi comment we can check rsnd_ssi_init(), not, rsnd_ssi_start() Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 82b04c6f352..aff5b76d1fd 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -209,7 +209,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, /* * this driver is assuming that * system word is 64fs (= 2 x 32bit) - * see rsnd_ssi_start() + * see rsnd_ssi_init() */ main_rate = rate / adg_clk_div_table[i] * 32 * 2 * ssi_clk_mul_table[j]; -- cgit v1.2.3-70-g09d2 From 690ef81ebe02a43991b0fcb418d77b8420346cfd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:27:03 -0800 Subject: ASoC: rsnd: tidyup register naming Use correct register name which appears in the datasheet Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 4 ++-- sound/soc/sh/rcar/rsnd.h | 4 ++-- sound/soc/sh/rcar/scu.c | 4 ++-- 3 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index d0ab203836d..862758d3ec0 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -314,11 +314,11 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL0, 0x08), RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL1, 0x0c), RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL2, 0x10), - RSND_GEN1_S_REG(gen, SRU, SRC_CTRL, 0xc0), + RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_CTRL, 0xc0), RSND_GEN1_S_REG(gen, SRU, SSI_MODE0, 0xD0), RSND_GEN1_S_REG(gen, SRU, SSI_MODE1, 0xD4), RSND_GEN1_M_REG(gen, SRU, BUSIF_MODE, 0x20, 0x4), - RSND_GEN1_M_REG(gen, SRU, BUSIF_ADINR, 0x214, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_ADINR, 0x214, 0x40), RSND_GEN1_S_REG(gen, ADG, BRRA, 0x00), RSND_GEN1_S_REG(gen, ADG, BRRB, 0x04), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index bff7b9e5306..d5c0182c8e5 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -36,12 +36,12 @@ enum rsnd_reg { RSND_REG_SRC_TMG_SEL0, RSND_REG_SRC_TMG_SEL1, RSND_REG_SRC_TMG_SEL2, - RSND_REG_SRC_CTRL, + RSND_REG_SRC_ROUTE_CTRL, RSND_REG_SSI_MODE0, RSND_REG_SSI_MODE1, RSND_REG_BUSIF_MODE, - RSND_REG_BUSIF_ADINR, RSND_REG_INT_ENABLE, + RSND_REG_SRC_ADINR, /* ADG */ RSND_REG_BRRA, diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 1ab1bce6be7..187f7dc3036 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -115,7 +115,7 @@ static int rsnd_scu_set_mode(struct rsnd_priv *priv, if (rsnd_is_gen1(priv)) { val = (1 << id); - rsnd_mod_bset(mod, SRC_CTRL, val, val); + rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val); } return 0; @@ -141,7 +141,7 @@ static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, } rsnd_mod_write(mod, BUSIF_MODE, 1); - rsnd_mod_write(mod, BUSIF_ADINR, adinr); + rsnd_mod_write(mod, SRC_ADINR, adinr); return 0; } -- cgit v1.2.3-70-g09d2 From 7808aa30d6cf366e5f627dcbf7c84f9dc6e602ab Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:27:19 -0800 Subject: ASoC: rsnd: make sure variable name for 44.1kHz/48kHz This driver is assuming that RBGA is used as source clock of 44.1kHz category, and RBGB is used as source clock of 48kHz category. This patch clarifies the variable name. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 55d0394955b..2e71a7bda4c 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -19,8 +19,8 @@ struct rsnd_adg { struct clk *clk[CLKMAX]; - int rate_of_441khz_div_6; - int rate_of_48khz_div_6; + int rbga_rate_for_441khz_div_6; /* RBGA */ + int rbgb_rate_for_48khz_div_6; /* RBGB */ u32 ckr; }; @@ -101,12 +101,12 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate) /* * find 1/6 clock from BRGA/BRGB */ - if (rate == adg->rate_of_441khz_div_6) { + if (rate == adg->rbga_rate_for_441khz_div_6) { data = 0x10; goto found_clock; } - if (rate == adg->rate_of_48khz_div_6) { + if (rate == adg->rbgb_rate_for_48khz_div_6) { data = 0x20; goto found_clock; } @@ -156,8 +156,8 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) * rsnd_adg_ssi_clk_try_start() */ ckr = 0; - adg->rate_of_441khz_div_6 = 0; - adg->rate_of_48khz_div_6 = 0; + adg->rbga_rate_for_441khz_div_6 = 0; + adg->rbgb_rate_for_48khz_div_6 = 0; for_each_rsnd_clk(clk, adg, i) { rate = clk_get_rate(clk); @@ -165,14 +165,14 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) continue; /* RBGA */ - if (!adg->rate_of_441khz_div_6 && (0 == rate % 44100)) { - adg->rate_of_441khz_div_6 = rate / 6; + if (!adg->rbga_rate_for_441khz_div_6 && (0 == rate % 44100)) { + adg->rbga_rate_for_441khz_div_6 = rate / 6; ckr |= brg_table[i] << 20; } /* RBGB */ - if (!adg->rate_of_48khz_div_6 && (0 == rate % 48000)) { - adg->rate_of_48khz_div_6 = rate / 6; + if (!adg->rbgb_rate_for_48khz_div_6 && (0 == rate % 48000)) { + adg->rbgb_rate_for_48khz_div_6 = rate / 6; ckr |= brg_table[i] << 16; } } -- cgit v1.2.3-70-g09d2 From 2582718cb6bd620d37a54db885c75bfe4822db45 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:27:37 -0800 Subject: ASoC: rsnd: route setting is needed only Gen1 Renesas sound has SRC (= Sampling Rate Converter), but, the HW implementation depends on its generation. It was part of SRU on Gen1, and SCU on Gen2. This SCU needs DMA transfer to use it. Current rsnd driver is using it as DMA transfer buffer (= no rate convert), and Gen1 is only supported at this point. This patch cleanup it with focusing about SRC and Gen2 part. rsnd_scu_set_route() is needed only Gen1. This patch renames it to rsnd_scu_set_route_if_gen1() and it adds comment to rsnd_reg member in order to clarify it is used for Gen1. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/rsnd.h | 10 +++++----- sound/soc/sh/rcar/scu.c | 6 ++++-- 2 files changed, 9 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index d5c0182c8e5..a14bc9274b3 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -32,11 +32,11 @@ */ enum rsnd_reg { /* SRU/SCU/SSIU */ - RSND_REG_SRC_ROUTE_SEL, - RSND_REG_SRC_TMG_SEL0, - RSND_REG_SRC_TMG_SEL1, - RSND_REG_SRC_TMG_SEL2, - RSND_REG_SRC_ROUTE_CTRL, + RSND_REG_SRC_ROUTE_SEL, /* for Gen1 */ + RSND_REG_SRC_TMG_SEL0, /* for Gen1 */ + RSND_REG_SRC_TMG_SEL1, /* for Gen1 */ + RSND_REG_SRC_TMG_SEL2, /* for Gen1 */ + RSND_REG_SRC_ROUTE_CTRL, /* for Gen1 */ RSND_REG_SSI_MODE0, RSND_REG_SSI_MODE1, RSND_REG_BUSIF_MODE, diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 187f7dc3036..7642ec52b59 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -36,7 +36,8 @@ struct rsnd_scu { ((pos) = (struct rsnd_scu *)(priv)->scu + i); \ i++) -static int rsnd_scu_set_route(struct rsnd_priv *priv, +/* Gen1 only */ +static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) @@ -174,7 +175,8 @@ static int rsnd_scu_start(struct rsnd_mod *mod, } /* it use DMA transter */ - ret = rsnd_scu_set_route(priv, mod, rdai, io); + + ret = rsnd_src_set_route_if_gen1(priv, mod, rdai, io); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From af8a478821345fd264fd2294e80f5b0a28a518bc Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:28:04 -0800 Subject: ASoC: rsnd: add rsnd_scu_transfer_start() Renesas sound has SRC (= Sampling Rate Converter), but, the HW implementation depends on its generation. It was part of SRU on Gen1, and SCU on Gen2. This SCU needs DMA transfer to use it. Current rsnd driver is using it as DMA transfer buffer (= no rate convert), and Gen1 is only supported at this point. This patch cleanup it with focusing about SRC and Gen2 part. SRC_CTRL/BUSIF_MODE are used for transfer start. This patch adds rsnd_scu_transfer_start() and merge these Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/scu.c | 39 ++++++++++++++++++++------------------- 1 file changed, 20 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 7642ec52b59..3d8b57b1630 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -106,22 +106,6 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, return 0; } -static int rsnd_scu_set_mode(struct rsnd_priv *priv, - struct rsnd_mod *mod, - struct rsnd_dai *rdai, - struct rsnd_dai_stream *io) -{ - int id = rsnd_mod_id(mod); - u32 val; - - if (rsnd_is_gen1(priv)) { - val = (1 << id); - rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val); - } - - return 0; -} - static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_dai *rdai, @@ -141,12 +125,29 @@ static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, return -EIO; } - rsnd_mod_write(mod, BUSIF_MODE, 1); rsnd_mod_write(mod, SRC_ADINR, adinr); return 0; } +static int rsnd_scu_transfer_start(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + int id = rsnd_mod_id(mod); + u32 val; + + if (rsnd_is_gen1(priv)) { + val = (1 << id); + rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val); + } + + rsnd_mod_write(mod, BUSIF_MODE, 1); + + return 0; +} + bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod) { struct rsnd_scu *scu = rsnd_mod_to_scu(mod); @@ -180,11 +181,11 @@ static int rsnd_scu_start(struct rsnd_mod *mod, if (ret < 0) return ret; - ret = rsnd_scu_set_mode(priv, mod, rdai, io); + ret = rsnd_scu_set_hpbif(priv, mod, rdai, io); if (ret < 0) return ret; - ret = rsnd_scu_set_hpbif(priv, mod, rdai, io); + ret = rsnd_scu_transfer_start(priv, mod, rdai, io); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From 52ea2a79f440740b57925e729a59337414b4c300 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:28:19 -0800 Subject: ASoC: rsnd: INT_ENABLE is needed only Gen2 INT_ENABLE is needed only Gen2. rsnd_mod_write() do nothing on Gen1, but it is confusable. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/ssi.c | 5 +++-- 2 files changed, 4 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index a14bc9274b3..3774dfcfaf0 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -40,7 +40,7 @@ enum rsnd_reg { RSND_REG_SSI_MODE0, RSND_REG_SSI_MODE1, RSND_REG_BUSIF_MODE, - RSND_REG_INT_ENABLE, + RSND_REG_INT_ENABLE, /* for Gen2 */ RSND_REG_SRC_ADINR, /* ADG */ diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index aff5b76d1fd..01b5cf945d6 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -457,8 +457,9 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod, /* enable PIO IRQ */ ssi->cr_etc = UIEN | OIEN | DIEN; - /* enable PIO interrupt */ - rsnd_mod_write(&ssi->mod, INT_ENABLE, 0x0f000000); + /* enable PIO interrupt if gen2 */ + if (rsnd_is_gen2(priv)) + rsnd_mod_write(&ssi->mod, INT_ENABLE, 0x0f000000); rsnd_ssi_hw_start(ssi, rdai, io); -- cgit v1.2.3-70-g09d2 From 99feec32f26a3c267f89ce48db4bd36650a95f7f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:28:31 -0800 Subject: ASoC: rsnd: scu cleanup: add rsnd_scu_rate_ctrl() Renesas sound has SRC (= Sampling Rate Converter), but, the HW implementation depends on its generation. It was part of SRU on Gen1, and SCU on Gen2. This SCU needs DMA transfer to use it. Current rsnd driver is using it as DMA transfer buffer (= no rate convert), and Gen1 is only supported at this point. This patch cleanup it with focusing about SRC and Gen2 part. rsnd_scu_set_hpbif() is renamed to rsnd_scu_rate_ctrl(), since its naming doesn't indicate the function meaning. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/scu.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 3d8b57b1630..5f4f57206fa 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -106,7 +106,7 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, return 0; } -static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, +static int rsnd_scu_rate_ctrl(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) @@ -181,7 +181,7 @@ static int rsnd_scu_start(struct rsnd_mod *mod, if (ret < 0) return ret; - ret = rsnd_scu_set_hpbif(priv, mod, rdai, io); + ret = rsnd_scu_rate_ctrl(priv, mod, rdai, io); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From adcf7d5e7605e8134a99d415b7afd13f03c4bf23 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:28:39 -0800 Subject: ASoC: rsnd: tidyup rsnd_ssi_master_clk_start() parameter Renesas sound has SRC (= Sampling Rate Converter), but, the HW implementation depends on its generation. It was part of SRU on Gen1, and SCU on Gen2. This SCU needs DMA transfer to use it. Current rsnd driver is using it as DMA transfer buffer (= no rate convert), and Gen1 is only supported at this point. This patch cleanup it with focusing about SRC and Gen2 part. ssi clock which is calculated from rsnd_ssi_master_clk_start() should have flexibility since Renesas sound has SRC (= Sampling Rate Converter). But current implementation is using runtime->rate directly. This patch tidyup rsnd_ssi_master_clk_start() parameter as preparation of future SRC support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 01b5cf945d6..2db9711549f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -187,9 +187,10 @@ static void rsnd_ssi_status_check(struct rsnd_mod *mod, } static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, - unsigned int rate) + struct rsnd_dai_stream *io) { struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct device *dev = rsnd_priv_to_dev(priv); int i, j, ret; int adg_clk_div_table[] = { @@ -199,6 +200,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, 1, 2, 4, 8, 16, 6, 12, }; unsigned int main_rate; + unsigned int rate = runtime->rate; /* * Find best clock, and try to start ADG @@ -251,14 +253,10 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, clk_enable(ssi->clk); if (rsnd_rdai_is_clk_master(rdai)) { - struct snd_pcm_runtime *runtime; - - runtime = rsnd_io_to_runtime(io); - if (rsnd_ssi_clk_from_parent(ssi)) rsnd_ssi_hw_start(ssi->parent, rdai, io); else - rsnd_ssi_master_clk_start(ssi, runtime->rate); + rsnd_ssi_master_clk_start(ssi, io); } } -- cgit v1.2.3-70-g09d2 From ef749400434cefd14fe02fe3de9e9f0125b2256d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Dec 2013 19:28:51 -0800 Subject: ASoC: rsnd: add SRC (Sampling Rate Converter) support This patch adds SRC support to Renesas sound driver. SRC converts sampling rate between codec <-> cpu. It needs special codec chip, or very simple DA/AD converter to use it. This patch was tested via ak4554 codec, and supports Gen1 only at this point. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 1 + sound/soc/sh/rcar/adg.c | 73 +++++++++++++++++++++++ sound/soc/sh/rcar/gen.c | 10 ++++ sound/soc/sh/rcar/rsnd.h | 18 ++++++ sound/soc/sh/rcar/scu.c | 152 ++++++++++++++++++++++++++++++++++++++++++++--- sound/soc/sh/rcar/ssi.c | 2 +- 6 files changed, 248 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index a818ff76b13..e147498abe5 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -58,6 +58,7 @@ struct rsnd_ssi_platform_info { struct rsnd_scu_platform_info { u32 flags; + u32 convert_rate; /* sampling rate convert */ }; /* diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 2e71a7bda4c..a53235c4d1b 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -30,6 +30,79 @@ struct rsnd_adg { i++, (pos) = adg->clk[i]) #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) +static int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv, + struct rsnd_mod *mod, + unsigned int src_rate, + unsigned int dst_rate) +{ + struct rsnd_adg *adg = rsnd_priv_to_adg(priv); + struct device *dev = rsnd_priv_to_dev(priv); + int idx, sel, div, shift; + u32 mask, val; + int id = rsnd_mod_id(mod); + unsigned int sel_rate [] = { + clk_get_rate(adg->clk[CLKA]), /* 000: CLKA */ + clk_get_rate(adg->clk[CLKB]), /* 001: CLKB */ + clk_get_rate(adg->clk[CLKC]), /* 010: CLKC */ + 0, /* 011: MLBCLK (not used) */ + adg->rbga_rate_for_441khz_div_6,/* 100: RBGA */ + adg->rbgb_rate_for_48khz_div_6, /* 101: RBGB */ + }; + + /* find div (= 1/128, 1/256, 1/512, 1/1024, 1/2048 */ + for (sel = 0; sel < ARRAY_SIZE(sel_rate); sel++) { + for (div = 128, idx = 0; + div <= 2048; + div *= 2, idx++) { + if (src_rate == sel_rate[sel] / div) { + val = (idx << 4) | sel; + goto find_rate; + } + } + } + dev_err(dev, "can't find convert src clk\n"); + return -EINVAL; + +find_rate: + shift = (id % 4) * 8; + mask = 0xFF << shift; + val = val << shift; + + dev_dbg(dev, "adg convert src clk = %02x\n", val); + + switch (id / 4) { + case 0: + rsnd_mod_bset(mod, AUDIO_CLK_SEL3, mask, val); + break; + case 1: + rsnd_mod_bset(mod, AUDIO_CLK_SEL4, mask, val); + break; + case 2: + rsnd_mod_bset(mod, AUDIO_CLK_SEL5, mask, val); + break; + } + + /* + * Gen1 doesn't need dst_rate settings, + * since it uses SSI WS pin. + * see also rsnd_src_set_route_if_gen1() + */ + + return 0; +} + +int rsnd_adg_set_convert_clk(struct rsnd_priv *priv, + struct rsnd_mod *mod, + unsigned int src_rate, + unsigned int dst_rate) +{ + if (rsnd_is_gen1(priv)) + return rsnd_adg_set_convert_clk_gen1(priv, mod, + src_rate, dst_rate); + + return -EINVAL; +} + static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val) { int id = rsnd_mod_id(mod); diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 862758d3ec0..add088bd4b2 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -318,13 +318,23 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) RSND_GEN1_S_REG(gen, SRU, SSI_MODE0, 0xD0), RSND_GEN1_S_REG(gen, SRU, SSI_MODE1, 0xD4), RSND_GEN1_M_REG(gen, SRU, BUSIF_MODE, 0x20, 0x4), + RSND_GEN1_M_REG(gen, SRU, SRC_ROUTE_MODE0,0x50, 0x8), + RSND_GEN1_M_REG(gen, SRU, SRC_SWRSR, 0x200, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_SRCIR, 0x204, 0x40), RSND_GEN1_M_REG(gen, SRU, SRC_ADINR, 0x214, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_IFSCR, 0x21c, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_IFSVR, 0x220, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_SRCCR, 0x224, 0x40), + RSND_GEN1_M_REG(gen, SRU, SRC_MNFSR, 0x228, 0x40), RSND_GEN1_S_REG(gen, ADG, BRRA, 0x00), RSND_GEN1_S_REG(gen, ADG, BRRB, 0x04), RSND_GEN1_S_REG(gen, ADG, SSICKR, 0x08), RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL3, 0x18), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL4, 0x1c), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL5, 0x20), RSND_GEN1_M_REG(gen, SSI, SSICR, 0x00, 0x40), RSND_GEN1_M_REG(gen, SSI, SSISR, 0x04, 0x40), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3774dfcfaf0..4ca66cd899c 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -41,7 +41,14 @@ enum rsnd_reg { RSND_REG_SSI_MODE1, RSND_REG_BUSIF_MODE, RSND_REG_INT_ENABLE, /* for Gen2 */ + RSND_REG_SRC_ROUTE_MODE0, + RSND_REG_SRC_SWRSR, + RSND_REG_SRC_SRCIR, RSND_REG_SRC_ADINR, + RSND_REG_SRC_IFSCR, + RSND_REG_SRC_IFSVR, + RSND_REG_SRC_SRCCR, + RSND_REG_SRC_MNFSR, /* ADG */ RSND_REG_BRRA, @@ -50,6 +57,9 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL0, RSND_REG_AUDIO_CLK_SEL1, RSND_REG_AUDIO_CLK_SEL2, + RSND_REG_AUDIO_CLK_SEL3, /* for Gen1 */ + RSND_REG_AUDIO_CLK_SEL4, /* for Gen1 */ + RSND_REG_AUDIO_CLK_SEL5, /* for Gen1 */ /* SSI */ RSND_REG_SSICR, @@ -227,6 +237,10 @@ int rsnd_adg_probe(struct platform_device *pdev, struct rsnd_priv *priv); void rsnd_adg_remove(struct platform_device *pdev, struct rsnd_priv *priv); +int rsnd_adg_set_convert_clk(struct rsnd_priv *priv, + struct rsnd_mod *mod, + unsigned int src_rate, + unsigned int dst_rate); /* * R-Car sound priv @@ -280,6 +294,10 @@ void rsnd_scu_remove(struct platform_device *pdev, struct rsnd_priv *priv); struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id); bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod); +unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv, + struct rsnd_mod *ssi_mod, + struct snd_pcm_runtime *runtime); + #define rsnd_scu_nr(priv) ((priv)->scu_nr) /* diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 5f4f57206fa..1406dd8d9ed 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -13,9 +13,13 @@ struct rsnd_scu { struct rsnd_scu_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; + struct clk *clk; }; #define rsnd_scu_mode_flags(p) ((p)->info->flags) +#define rsnd_scu_convert_rate(p) ((p)->info->convert_rate) + +#define RSND_SCU_NAME_SIZE 16 /* * ADINR @@ -26,6 +30,15 @@ struct rsnd_scu { #define OTBL_18 (6 << 16) #define OTBL_16 (8 << 16) +/* + * image of SRC (Sampling Rate Converter) + * + * 96kHz <-> +-----+ 48kHz +-----+ 48kHz +-------+ + * 48kHz <-> | SRC | <------> | SSI | <-----> | codec | + * 44.1kHz <-> +-----+ +-----+ +-------+ + * ... + * + */ #define rsnd_mod_to_scu(_mod) \ container_of((_mod), struct rsnd_scu, mod) @@ -56,7 +69,7 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, { 0x3, 28, }, /* 7 */ { 0x3, 30, }, /* 8 */ }; - + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); u32 mask; u32 val; int shift; @@ -86,9 +99,18 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, */ shift = (id % 4) * 8; mask = 0x1F << shift; - if (8 == id) /* SRU8 is very special */ + + /* + * ADG is used as source clock if SRC was used, + * then, SSI WS is used as destination clock. + * SSI WS is used as source clock if SRC is not used + * (when playback, source/destination become reverse when capture) + */ + if (rsnd_scu_convert_rate(scu)) /* use ADG */ + val = 0; + else if (8 == id) /* use SSI WS, but SRU8 is special */ val = id << shift; - else + else /* use SSI WS */ val = (id + 1) << shift; switch (id / 4) { @@ -106,14 +128,45 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv, return 0; } -static int rsnd_scu_rate_ctrl(struct rsnd_priv *priv, +unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv, + struct rsnd_mod *ssi_mod, + struct snd_pcm_runtime *runtime) +{ + struct rsnd_scu *scu; + unsigned int rate; + + /* this function is assuming SSI id = SCU id here */ + scu = rsnd_mod_to_scu(rsnd_scu_mod_get(priv, rsnd_mod_id(ssi_mod))); + + /* + * return convert rate if SRC is used, + * otherwise, return runtime->rate as usual + */ + rate = rsnd_scu_convert_rate(scu); + if (!rate) + rate = runtime->rate; + + return rate; +} + +static int rsnd_scu_convert_rate_ctrl(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); + u32 convert_rate = rsnd_scu_convert_rate(scu); u32 adinr = runtime->channels; + /* set/clear soft reset */ + rsnd_mod_write(mod, SRC_SWRSR, 0); + rsnd_mod_write(mod, SRC_SWRSR, 1); + + /* Initialize the operation of the SRC internal circuits */ + rsnd_mod_write(mod, SRC_SRCIR, 1); + + /* Set channel number and output bit length */ switch (runtime->sample_bits) { case 16: adinr |= OTBL_16; @@ -124,9 +177,42 @@ static int rsnd_scu_rate_ctrl(struct rsnd_priv *priv, default: return -EIO; } - rsnd_mod_write(mod, SRC_ADINR, adinr); + if (convert_rate) { + u32 fsrate = 0x0400000 / convert_rate * runtime->rate; + int ret; + + /* Enable the initial value of IFS */ + rsnd_mod_write(mod, SRC_IFSCR, 1); + + /* Set initial value of IFS */ + rsnd_mod_write(mod, SRC_IFSVR, fsrate); + + /* Select SRC mode (fixed value) */ + rsnd_mod_write(mod, SRC_SRCCR, 0x00010110); + + /* Set the restriction value of the FS ratio (98%) */ + rsnd_mod_write(mod, SRC_MNFSR, fsrate / 100 * 98); + + if (rsnd_is_gen1(priv)) { + /* no SRC_BFSSR settings, since SRC_SRCCR::BUFMD is 0 */ + } + + /* set convert clock */ + ret = rsnd_adg_set_convert_clk(priv, mod, + runtime->rate, + convert_rate); + if (ret < 0) + return ret; + } + + /* Cancel the initialization and operate the SRC function */ + rsnd_mod_write(mod, SRC_SRCIR, 0); + + /* use DMA transfer */ + rsnd_mod_write(mod, BUSIF_MODE, 1); + return 0; } @@ -135,6 +221,7 @@ static int rsnd_scu_transfer_start(struct rsnd_priv *priv, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) { + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); int id = rsnd_mod_id(mod); u32 val; @@ -143,7 +230,28 @@ static int rsnd_scu_transfer_start(struct rsnd_priv *priv, rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val); } - rsnd_mod_write(mod, BUSIF_MODE, 1); + if (rsnd_scu_convert_rate(scu)) + rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1); + + return 0; +} + +static int rsnd_scu_transfer_stop(struct rsnd_priv *priv, + struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); + int id = rsnd_mod_id(mod); + u32 mask; + + if (rsnd_is_gen1(priv)) { + mask = (1 << id); + rsnd_mod_bset(mod, SRC_ROUTE_CTRL, mask, 0); + } + + if (rsnd_scu_convert_rate(scu)) + rsnd_mod_write(mod, SRC_ROUTE_MODE0, 0); return 0; } @@ -161,6 +269,7 @@ static int rsnd_scu_start(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); struct device *dev = rsnd_priv_to_dev(priv); int ret; @@ -175,13 +284,15 @@ static int rsnd_scu_start(struct rsnd_mod *mod, return 0; } + clk_enable(scu->clk); + /* it use DMA transter */ ret = rsnd_src_set_route_if_gen1(priv, mod, rdai, io); if (ret < 0) return ret; - ret = rsnd_scu_rate_ctrl(priv, mod, rdai, io); + ret = rsnd_scu_convert_rate_ctrl(priv, mod, rdai, io); if (ret < 0) return ret; @@ -194,9 +305,27 @@ static int rsnd_scu_start(struct rsnd_mod *mod, return 0; } +static int rsnd_scu_stop(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct rsnd_dai_stream *io) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); + + if (!rsnd_scu_hpbif_is_enable(mod)) + return 0; + + rsnd_scu_transfer_stop(priv, mod, rdai, io); + + clk_disable(scu->clk); + + return 0; +} + static struct rsnd_mod_ops rsnd_scu_ops = { .name = "scu", .start = rsnd_scu_start, + .stop = rsnd_scu_stop, }; struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id) @@ -212,6 +341,8 @@ int rsnd_scu_probe(struct platform_device *pdev, { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_scu *scu; + struct clk *clk; + char name[RSND_SCU_NAME_SIZE]; int i, nr; /* @@ -228,9 +359,16 @@ int rsnd_scu_probe(struct platform_device *pdev, priv->scu = scu; for_each_rsnd_scu(scu, priv, i) { + snprintf(name, RSND_SCU_NAME_SIZE, "scu.%d", i); + + clk = devm_clk_get(dev, name); + if (IS_ERR(clk)) + return PTR_ERR(clk); + rsnd_mod_init(priv, &scu->mod, &rsnd_scu_ops, i); scu->info = &info->scu_info[i]; + scu->clk = clk; dev_dbg(dev, "SCU%d probed\n", i); } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 2db9711549f..b7cd06be943 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -200,7 +200,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, 1, 2, 4, 8, 16, 6, 12, }; unsigned int main_rate; - unsigned int rate = runtime->rate; + unsigned int rate = rsnd_scu_get_ssi_rate(priv, &ssi->mod, runtime); /* * Find best clock, and try to start ADG -- cgit v1.2.3-70-g09d2 From e41975edc73d2c16d0784e5fa87a6162e2fcab80 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Fri, 20 Dec 2013 14:39:51 +0800 Subject: ASoC: core: Fix the DAI name getting. From "ASoC: make snd_soc_dai_link more symmetrical", can we see that the name of CPU DAI maybe omitted. If the DAI name is omitted, try to use the component name instead. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e53d87e881..03c779ebd72 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4617,10 +4617,14 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, if (id < 0 || id >= pos->num_dai) { ret = -EINVAL; - } else { - *dai_name = pos->dai_drv[id].name; - ret = 0; + break; } + + ret = 0; + + *dai_name = pos->dai_drv[id].name; + if (!*dai_name) + *dai_name = pos->name; } break; -- cgit v1.2.3-70-g09d2 From 14c3aa98397b3b4e67b042f388f4ad0f9b18250e Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Mon, 23 Dec 2013 17:28:53 +0800 Subject: ASoC: imx-spdif: Use snd-soc-dummy CODEC driver to link card This is a quick fix for the below two issues when building spdif as modules. 1) If modprobing modules in order: (Step 1) snd-soc-fsl-spdif -> (Step 2) snd-soc-imx-spdif -> (Step 3) snd-soc-spdif-tx/rx, we will fail to create imx-spdif card and dai link unless we rmmod snd-soc-imx-spdif and modprobe it again due to the execution platform_driver_unregister() in probe() when meeting -EPROBE_DEFER at Step 2. 2) After "imx-spdif sound-spdif.17: dit-hifi <-> 2004000.spdif mapping ok", 'rmmod snd-soc-imx-spdif' would cause kernel dump with warning: WARNING: CPU: 0 PID: 1301 at /home/rmk/git/linux-rmk/fs/sysfs/dir.c:915 sysfs_hash_and_remove+0x84/0x90() sysfs: can not remove 'dapm_widget', no directory This should be caused by disordered resourse releasing of the whole link. And trying to unregister the card and then CODEC dev can't fix this issue. Thus this patch just provides a simple fix to these two bugs by using the snd-soc-dummy in the core instead of seperate snd-soc-spdif-tx/rx so that there's no need to handle the registering and unregistering of CODEC or CODEC dai any more. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 - sound/soc/fsl/imx-spdif.c | 84 ++++++++++++----------------------------------- 2 files changed, 21 insertions(+), 64 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index b7ab71f2ccc..8c10b8f80ac 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -197,7 +197,6 @@ config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select SND_SOC_SPDIF select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 8499d5292f0..e656245d24c 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -14,17 +14,15 @@ #include struct imx_spdif_data { - struct snd_soc_dai_link dai[2]; + struct snd_soc_dai_link dai; struct snd_soc_card card; - struct platform_device *txdev; - struct platform_device *rxdev; }; static int imx_spdif_audio_probe(struct platform_device *pdev) { struct device_node *spdif_np, *np = pdev->dev.of_node; struct imx_spdif_data *data; - int ret = 0, num_links = 0; + int ret = 0; spdif_np = of_parse_phandle(np, "spdif-controller", 0); if (!spdif_np) { @@ -40,69 +38,42 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) goto end; } - if (of_property_read_bool(np, "spdif-out")) { - data->dai[num_links].name = "S/PDIF TX"; - data->dai[num_links].stream_name = "S/PDIF PCM Playback"; - data->dai[num_links].codec_dai_name = "dit-hifi"; - data->dai[num_links].codec_name = "spdif-dit"; - data->dai[num_links].cpu_of_node = spdif_np; - data->dai[num_links].platform_of_node = spdif_np; - num_links++; - - data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); - if (IS_ERR(data->txdev)) { - ret = PTR_ERR(data->txdev); - dev_err(&pdev->dev, "register dit failed: %d\n", ret); - goto end; - } - } + data->dai.name = "S/PDIF PCM"; + data->dai.stream_name = "S/PDIF PCM"; + data->dai.codec_dai_name = "snd-soc-dummy-dai"; + data->dai.codec_name = "snd-soc-dummy"; + data->dai.cpu_of_node = spdif_np; + data->dai.platform_of_node = spdif_np; + data->dai.playback_only = true; + data->dai.capture_only = true; - if (of_property_read_bool(np, "spdif-in")) { - data->dai[num_links].name = "S/PDIF RX"; - data->dai[num_links].stream_name = "S/PDIF PCM Capture"; - data->dai[num_links].codec_dai_name = "dir-hifi"; - data->dai[num_links].codec_name = "spdif-dir"; - data->dai[num_links].cpu_of_node = spdif_np; - data->dai[num_links].platform_of_node = spdif_np; - num_links++; - - data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); - if (IS_ERR(data->rxdev)) { - ret = PTR_ERR(data->rxdev); - dev_err(&pdev->dev, "register dir failed: %d\n", ret); - goto error_dit; - } - } + if (of_property_read_bool(np, "spdif-out")) + data->dai.capture_only = false; + + if (of_property_read_bool(np, "spdif-in")) + data->dai.playback_only = false; - if (!num_links) { + if (data->dai.playback_only && data->dai.capture_only) { dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); - goto error_dir; + goto end; } data->card.dev = &pdev->dev; - data->card.num_links = num_links; - data->card.dai_link = data->dai; + data->card.dai_link = &data->dai; + data->card.num_links = 1; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto error_dir; + goto end; ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); - goto error_dir; + goto end; } platform_set_drvdata(pdev, data); - goto end; - -error_dir: - if (data->rxdev) - platform_device_unregister(data->rxdev); -error_dit: - if (data->txdev) - platform_device_unregister(data->txdev); end: if (spdif_np) of_node_put(spdif_np); @@ -110,18 +81,6 @@ end: return ret; } -static int imx_spdif_audio_remove(struct platform_device *pdev) -{ - struct imx_spdif_data *data = platform_get_drvdata(pdev); - - if (data->rxdev) - platform_device_unregister(data->rxdev); - if (data->txdev) - platform_device_unregister(data->txdev); - - return 0; -} - static const struct of_device_id imx_spdif_dt_ids[] = { { .compatible = "fsl,imx-audio-spdif", }, { /* sentinel */ } @@ -135,7 +94,6 @@ static struct platform_driver imx_spdif_driver = { .of_match_table = imx_spdif_dt_ids, }, .probe = imx_spdif_audio_probe, - .remove = imx_spdif_audio_remove, }; module_platform_driver(imx_spdif_driver); -- cgit v1.2.3-70-g09d2 From 71467e46414d3bab220de77d3d085be0c0aa03e1 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 23 Dec 2013 15:25:38 +0800 Subject: ASoC: simple-card: Add device's module clock selection. Try to get the device's module clock if the dt has no clocks and system-clock-frequency properties. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3ba65bb6343..58c217e403a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -90,14 +90,29 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * dai->sysclk come from * "clocks = <&xxx>" (if system has common clock) * or "system-clock-frequency = " + * or device's module clock. */ - clk = of_clk_get(np, 0); - if (IS_ERR(clk)) + if (of_property_read_bool(np, "clocks")) { + clk = of_clk_get(np, 0); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + goto parse_error; + } + + dai->sysclk = clk_get_rate(clk); + } else if (of_property_read_bool(np, "system-clock-frequency")) { of_property_read_u32(np, "system-clock-frequency", &dai->sysclk); - else + } else { + clk = of_clk_get(*node, 0); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + goto parse_error; + } + dai->sysclk = clk_get_rate(clk); + } ret = 0; -- cgit v1.2.3-70-g09d2 From f60e5473e6788f93849a61198bec4e02fea31e51 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Dec 2013 11:42:55 +0100 Subject: ASoC: ssm2518: Fix off-by-one error by ffs() ffs() returns the bit position from 1, while the ssm2158 driver code assumes it being 0-based. Also, the bit mask computation of the two channel slots are incorrect; it must have worked just casually. Signed-off-by: Takashi Iwai Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2518.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 95aed552139..cc8debce752 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -549,13 +549,13 @@ static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, right_slot = 0; } else { /* We assume the left channel < right channel */ - left_slot = ffs(tx_mask); - tx_mask &= ~(1 << tx_mask); + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); if (tx_mask == 0) { right_slot = left_slot; } else { - right_slot = ffs(tx_mask); - tx_mask &= ~(1 << tx_mask); + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); } } -- cgit v1.2.3-70-g09d2 From c097d5fdf3b51cdb2521c3cffab0a8cf03c68cc6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Dec 2013 12:41:39 +0000 Subject: ASoC: ad1836: Reject unsupported bit sizes Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/ad1836.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 9a92b7962f4..d7c983862cf 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -179,6 +179,8 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S32_LE: word_len = AD1836_WORD_LEN_24; break; + default: + return -EINVAL; } regmap_update_bits(ad1836->regmap, AD1836_DAC_CTRL1, -- cgit v1.2.3-70-g09d2 From 6873ee464a9fd23f0b7c2ab38e4ab8cea02cb50d Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Sun, 5 Jan 2014 10:21:16 +0400 Subject: ASoC: fsl_ssi: Fix printing return code on clk error Signed-off-by: Alexander Shiyan Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b2ebaf81159..6e3d38a8528 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1176,7 +1176,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); if (IS_ERR(ssi_private->baudclk)) - dev_warn(&pdev->dev, "could not get baud clock: %d\n", ret); + dev_warn(&pdev->dev, "could not get baud clock: %d\n", + PTR_ERR(ssi_private->baudclk)); else clk_prepare_enable(ssi_private->baudclk); -- cgit v1.2.3-70-g09d2 From e2a19ac6c5b27ac93fe744c0ff0823cde52c9cbb Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 6 Jan 2014 12:34:36 +0800 Subject: ASoC: simple-card: Fix the sysclk selection. For spdif there is no need to do the sysclk setting. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 58c217e403a..d4402fb5725 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -106,12 +106,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np, &dai->sysclk); } else { clk = of_clk_get(*node, 0); - if (IS_ERR(clk)) { - ret = PTR_ERR(clk); - goto parse_error; - } - - dai->sysclk = clk_get_rate(clk); + if (!IS_ERR(clk)) + dai->sysclk = clk_get_rate(clk); } ret = 0; -- cgit v1.2.3-70-g09d2 From a5d3f6abbf0f8be882d752da33b3e204c2d76f59 Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Sun, 5 Jan 2014 11:38:31 +0400 Subject: ASoC: mc13783: Use module_platform_driver_probe() mc13783-codec is probed only by MC13XXX MFD core driver so use module_platform_driver_probe(). Signed-off-by: Alexander Shiyan Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index bae60164c7b..8ab96686022 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -750,7 +750,7 @@ static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { .num_dapm_routes = ARRAY_SIZE(mc13783_routes), }; -static int mc13783_codec_probe(struct platform_device *pdev) +static int __init mc13783_codec_probe(struct platform_device *pdev) { struct mc13xxx *mc13xxx; struct mc13783_priv *priv; @@ -804,11 +804,9 @@ static struct platform_driver mc13783_codec_driver = { .name = "mc13783-codec", .owner = THIS_MODULE, }, - .probe = mc13783_codec_probe, .remove = mc13783_codec_remove, }; - -module_platform_driver(mc13783_codec_driver); +module_platform_driver_probe(mc13783_codec_driver, mc13783_codec_probe); MODULE_DESCRIPTION("ASoC MC13783 driver"); MODULE_AUTHOR("Sascha Hauer, Pengutronix "); -- cgit v1.2.3-70-g09d2 From 0acb26a6c716ef2f8ab550475c5da4d187995cca Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Sun, 5 Jan 2014 11:38:32 +0400 Subject: ASoC: mc13783: Use core error messages if registration fails Signed-off-by: Alexander Shiyan Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8ab96686022..c2def5d188e 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -781,14 +781,6 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); - if (ret) - goto err_register_codec; - - return 0; - -err_register_codec: - dev_err(&pdev->dev, "register codec failed with %d\n", ret); - return ret; } -- cgit v1.2.3-70-g09d2 From 295b84237b4ec2e1f148c8f6d7f59a7d06fda624 Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Sun, 5 Jan 2014 11:38:33 +0400 Subject: ASoC: mc13783: Drop fixed ADC & DAC ports usage There are no users of this driver without pdata, so stop using constant assignment of ADC and DAC ports. Signed-off-by: Alexander Shiyan Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index c2def5d188e..997f708afc7 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -770,8 +770,7 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) priv->adc_ssi_port = pdata->adc_ssi_port; priv->dac_ssi_port = pdata->dac_ssi_port; } else { - priv->adc_ssi_port = MC13783_SSI1_PORT; - priv->dac_ssi_port = MC13783_SSI2_PORT; + return -ENOSYS; } if (priv->adc_ssi_port == priv->dac_ssi_port) -- cgit v1.2.3-70-g09d2 From 2b32098f74ad6e8e3c0dbc714aa0f14c2f7df20a Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Sun, 5 Jan 2014 11:38:34 +0400 Subject: ASoC: mc13783: trivial: Cleanup module This is a trivial cleanup: remove useless variable mc13xxx and extra spaces. No functional changes. Signed-off-by: Alexander Shiyan Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 997f708afc7..582c2bbd42c 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -752,20 +752,14 @@ static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { static int __init mc13783_codec_probe(struct platform_device *pdev) { - struct mc13xxx *mc13xxx; struct mc13783_priv *priv; struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data; int ret; - mc13xxx = dev_get_drvdata(pdev->dev.parent); - - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (priv == NULL) + if (!priv) return -ENOMEM; - dev_set_drvdata(&pdev->dev, priv); - priv->mc13xxx = mc13xxx; if (pdata) { priv->adc_ssi_port = pdata->adc_ssi_port; priv->dac_ssi_port = pdata->dac_ssi_port; @@ -773,6 +767,9 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) return -ENOSYS; } + dev_set_drvdata(&pdev->dev, priv); + priv->mc13xxx = dev_get_drvdata(pdev->dev.parent); + if (priv->adc_ssi_port == priv->dac_ssi_port) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync)); @@ -792,9 +789,9 @@ static int mc13783_codec_remove(struct platform_device *pdev) static struct platform_driver mc13783_codec_driver = { .driver = { - .name = "mc13783-codec", - .owner = THIS_MODULE, - }, + .name = "mc13783-codec", + .owner = THIS_MODULE, + }, .remove = mc13783_codec_remove, }; module_platform_driver_probe(mc13783_codec_driver, mc13783_codec_probe); -- cgit v1.2.3-70-g09d2 From 6ed54f08bab0a93d53fddcd37b69d6b15fbef500 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 20 Dec 2013 14:20:07 +0100 Subject: ASoC: atmel: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Acked-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm-dma.c | 1 - sound/soc/atmel/atmel-pcm-pdc.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 06082e5e5dc..b79a2a86451 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -50,7 +50,6 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE, .period_bytes_min = 256, /* lighting DMA overhead */ .period_bytes_max = 2 * 0xffff, /* if 2 bytes format */ .periods_min = 8, diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index 054ea4d9326..33ec592ecd7 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -58,7 +58,6 @@ static const struct snd_pcm_hardware atmel_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE, .period_bytes_min = 32, .period_bytes_max = 8192, .periods_min = 2, -- cgit v1.2.3-70-g09d2 From 08ae9b456d393dfd1bbe7619b994189be6a26449 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 14:19:06 +0100 Subject: ASoC: dpcm: Add helper function for initializing runtime pcm We have the same code for initializing the runtime pcm on both the playback and the capture path. Factor this out into a common helper function. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 10f29a0ad5a..b649e32791d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1228,6 +1228,17 @@ unwind: return err; } +static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, + struct snd_soc_pcm_stream *stream) +{ + runtime->hw.rate_min = stream->rate_min; + runtime->hw.rate_max = stream->rate_max; + runtime->hw.channels_min = stream->channels_min; + runtime->hw.channels_max = stream->channels_max; + runtime->hw.formats &= stream->formats; + runtime->hw.rates = stream->rates; +} + static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1235,21 +1246,10 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - runtime->hw.rate_min = cpu_dai_drv->playback.rate_min; - runtime->hw.rate_max = cpu_dai_drv->playback.rate_max; - runtime->hw.channels_min = cpu_dai_drv->playback.channels_min; - runtime->hw.channels_max = cpu_dai_drv->playback.channels_max; - runtime->hw.formats &= cpu_dai_drv->playback.formats; - runtime->hw.rates = cpu_dai_drv->playback.rates; - } else { - runtime->hw.rate_min = cpu_dai_drv->capture.rate_min; - runtime->hw.rate_max = cpu_dai_drv->capture.rate_max; - runtime->hw.channels_min = cpu_dai_drv->capture.channels_min; - runtime->hw.channels_max = cpu_dai_drv->capture.channels_max; - runtime->hw.formats &= cpu_dai_drv->capture.formats; - runtime->hw.rates = cpu_dai_drv->capture.rates; - } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); + else + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); } static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) -- cgit v1.2.3-70-g09d2 From 002220a90db8ab9a6313887934dec25b54404cbd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 14:19:07 +0100 Subject: ASoC: dpcm: Allow PCMs to omit the set of supported formats Allow PCMs that do not impose any restrictions on the supported formats to set the formats field to 0, Instead of assuming that this means that the PCM does not support any formats (which doesn't make much sense), assume that it supports all formats. This brings the behavior of DPCM closer to that of non-DPCM. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b649e32791d..feb0f284302 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1235,7 +1235,10 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, runtime->hw.rate_max = stream->rate_max; runtime->hw.channels_min = stream->channels_min; runtime->hw.channels_max = stream->channels_max; - runtime->hw.formats &= stream->formats; + if (runtime->hw.formats) + runtime->hw.formats &= stream->formats; + else + runtime->hw.formats = stream->formats; runtime->hw.rates = stream->rates; } -- cgit v1.2.3-70-g09d2 From ff1b15acb44398f1a23e804fc0e178c952ee7fde Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 7 Jan 2014 08:00:13 -0200 Subject: ASoC: fsl: fsl_ssi: Use '%ld' to print 'long int' Commit 6873ee464a (ASoC: fsl_ssi: Fix printing return code on clk error) caused the following build warning: sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_probe': sound/soc/fsl/fsl_ssi.c:1196:6: warning: format '%d' expects argument of type 'int', but argument 3 has type 'long int' [-Wformat] Fix it by using '%ld' to print the 'long int' format. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6e3d38a8528..816ae4b28a5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1176,7 +1176,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); if (IS_ERR(ssi_private->baudclk)) - dev_warn(&pdev->dev, "could not get baud clock: %d\n", + dev_warn(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); else clk_prepare_enable(ssi_private->baudclk); -- cgit v1.2.3-70-g09d2 From b0a23b8b36e1fb754dcbdfe622e5ca5ded2f188b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 14:19:09 +0100 Subject: ASoC: fsl: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 7 ------- sound/soc/fsl/imx-pcm-dma.c | 3 --- sound/soc/fsl/imx-pcm-fiq.c | 3 --- sound/soc/fsl/mpc5200_dma.c | 4 ---- 4 files changed, 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index d570f8c81dc..6bb0ea59284 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -55,10 +55,6 @@ SNDRV_PCM_FMTBIT_S32_BE | \ SNDRV_PCM_FMTBIT_U32_LE | \ SNDRV_PCM_FMTBIT_U32_BE) - -#define FSLDMA_PCM_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ - SNDRV_PCM_RATE_CONTINUOUS) - struct dma_object { struct snd_soc_platform_driver dai; dma_addr_t ssi_stx_phys; @@ -140,9 +136,6 @@ static const struct snd_pcm_hardware fsl_dma_hardware = { SNDRV_PCM_INFO_JOINT_DUPLEX | SNDRV_PCM_INFO_PAUSE, .formats = FSLDMA_PCM_FORMATS, - .rates = FSLDMA_PCM_RATES, - .rate_min = 5512, - .rate_max = 192000, .period_bytes_min = 512, /* A reasonable limit */ .period_bytes_max = (u32) -1, .periods_min = NUM_DMA_LINKS, diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index c5e47f866b4..2585ae44e63 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -41,9 +41,6 @@ static const struct snd_pcm_hardware imx_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .channels_min = 2, - .channels_max = 2, .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, .period_bytes_min = 128, .period_bytes_max = 65535, /* Limited by SDMA engine */ diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index c75d43bb2e9..6553202dd48 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -162,9 +162,6 @@ static struct snd_pcm_hardware snd_imx_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .channels_min = 2, - .channels_max = 2, .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, .period_bytes_min = 128, .period_bytes_max = 16 * 1024, diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 71bf2f248cd..f2b5d756b1f 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -200,10 +200,6 @@ static const struct snd_pcm_hardware psc_dma_hardware = { SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, .period_bytes_max = 1024 * 1024, .period_bytes_min = 32, .periods_min = 2, -- cgit v1.2.3-70-g09d2 From f3b6079683e371eff8772882448020c29913cab1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 14:19:12 +0100 Subject: ASoC: mxs: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-pcm.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 04a6b0d6094..2e0863a70da 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -36,11 +36,6 @@ static const struct snd_pcm_hardware snd_mxs_hardware = { SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_HALF_DUPLEX, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | - SNDRV_PCM_FMTBIT_S24_LE, - .channels_min = 2, - .channels_max = 2, .period_bytes_min = 32, .period_bytes_max = 8192, .periods_min = 1, -- cgit v1.2.3-70-g09d2 From 96ae0f08ac574f3dac17cff9afdeee5562f61cbb Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 17:19:22 +0100 Subject: ASoC: mxs: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag Since commit 7b11304 ("dma: mxs-dma: Report correct residue for cyclic DMA") the mxs dmaengine driver has support for residue reporting. So there is no need to specify the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag anymore. This allows a finer grained resolution for the PCM pointer as well as avoids the race condition that can occur with the period counting that is used when the dmaengine driver does not support residue reporting. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-pcm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 2e0863a70da..a371b4f91c5 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -52,7 +52,6 @@ static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { int mxs_pcm_platform_register(struct device *dev) { return devm_snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); -- cgit v1.2.3-70-g09d2 From 0475680b5c2ef4bbdc3af1f6cfd014ea08c8d981 Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 19 Dec 2013 15:54:58 +0000 Subject: ARM: ux500: Don't use enums for MSP IDs - for easy DT conversion Signed-off-by: Lee Jones Acked-by: Linus Walleij Signed-off-by: Mark Brown --- arch/arm/mach-ux500/board-mop500-audio.c | 8 ++++---- include/linux/platform_data/asoc-ux500-msp.h | 9 +-------- sound/soc/ux500/ux500_msp_i2s.h | 2 +- 3 files changed, 6 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/arch/arm/mach-ux500/board-mop500-audio.c b/arch/arm/mach-ux500/board-mop500-audio.c index 154e15f5970..43d6cb8c381 100644 --- a/arch/arm/mach-ux500/board-mop500-audio.c +++ b/arch/arm/mach-ux500/board-mop500-audio.c @@ -31,7 +31,7 @@ static struct stedma40_chan_cfg msp0_dma_tx = { }; struct msp_i2s_platform_data msp0_platform_data = { - .id = MSP_I2S_0, + .id = 0, .msp_i2s_dma_rx = &msp0_dma_rx, .msp_i2s_dma_tx = &msp0_dma_tx, }; @@ -49,7 +49,7 @@ static struct stedma40_chan_cfg msp1_dma_tx = { }; struct msp_i2s_platform_data msp1_platform_data = { - .id = MSP_I2S_1, + .id = 1, .msp_i2s_dma_rx = NULL, .msp_i2s_dma_tx = &msp1_dma_tx, }; @@ -69,13 +69,13 @@ static struct stedma40_chan_cfg msp2_dma_tx = { }; struct msp_i2s_platform_data msp2_platform_data = { - .id = MSP_I2S_2, + .id = 2, .msp_i2s_dma_rx = &msp2_dma_rx, .msp_i2s_dma_tx = &msp2_dma_tx, }; struct msp_i2s_platform_data msp3_platform_data = { - .id = MSP_I2S_3, + .id = 3, .msp_i2s_dma_rx = &msp1_dma_rx, .msp_i2s_dma_tx = NULL, }; diff --git a/include/linux/platform_data/asoc-ux500-msp.h b/include/linux/platform_data/asoc-ux500-msp.h index 9991aea3d57..2f34bb98fe2 100644 --- a/include/linux/platform_data/asoc-ux500-msp.h +++ b/include/linux/platform_data/asoc-ux500-msp.h @@ -10,16 +10,9 @@ #include -enum msp_i2s_id { - MSP_I2S_0 = 0, - MSP_I2S_1, - MSP_I2S_2, - MSP_I2S_3, -}; - /* Platform data structure for a MSP I2S-device */ struct msp_i2s_platform_data { - enum msp_i2s_id id; + int id; struct stedma40_chan_cfg *msp_i2s_dma_rx; struct stedma40_chan_cfg *msp_i2s_dma_tx; }; diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 258d0bcee0b..875de0f68b8 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -475,7 +475,7 @@ struct ux500_msp_dma_params { }; struct ux500_msp { - enum msp_i2s_id id; + int id; void __iomem *registers; struct device *dev; struct ux500_msp_dma_params playback_dma_data; -- cgit v1.2.3-70-g09d2 From a61f9e314ad8ab9434ddd989b857ed93fdc725e2 Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 19 Dec 2013 15:55:00 +0000 Subject: ASoC: ux500: Provide better checking for Device Tree and/or Platform Data These drivers will not work without platform specific data, which is passed in via Device Tree or Platform Data. To avoid the chance of NULL pointer dereferencing and alike, let's ensure we have at least one of the methods in play before continuing. Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index c6fb5cce980..bc042cce115 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -771,10 +771,14 @@ static const struct snd_soc_component_driver ux500_msp_component = { static int ux500_msp_drv_probe(struct platform_device *pdev) { struct ux500_msp_i2s_drvdata *drvdata; + struct msp_i2s_platform_data *pdata = pdev->dev.platform_data; + struct device_node *np = pdev->dev.of_node; int ret = 0; - dev_dbg(&pdev->dev, "%s: Enter (pdev->name = %s).\n", __func__, - pdev->name); + if (!pdata && !np) { + dev_err(&pdev->dev, "No platform data or Device Tree found\n"); + return -ENODEV; + } drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp_i2s_drvdata), -- cgit v1.2.3-70-g09d2 From ae276e93b8ccb933c8cfca368427d1eafd07128d Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 19 Dec 2013 15:55:01 +0000 Subject: ASoC: Ux500: Match platform by device node when booting Device Tree We're getting closer to fully enabling the Ux500 ASoC driver for Device Tree. When we switch over from using AUXDATA we'll need to match platform by only Device Tree nodes. In this patch we NULL out the platform_name, and supply nodes for each platform device. Acked-by: Linus Walleij Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/mop500.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 178d1bad625..b3b66aa98dc 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -91,6 +91,8 @@ static int mop500_of_probe(struct platform_device *pdev, for (i = 0; i < 2; i++) { mop500_dai_links[i].cpu_of_node = msp_np[i]; mop500_dai_links[i].cpu_dai_name = NULL; + mop500_dai_links[i].platform_of_node = msp_np[i]; + mop500_dai_links[i].platform_name = NULL; mop500_dai_links[i].codec_of_node = codec_np; mop500_dai_links[i].codec_name = NULL; } -- cgit v1.2.3-70-g09d2 From 609a3050b8a516d12cf6dc0e8beb5875ededad3d Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 19 Dec 2013 15:55:02 +0000 Subject: ASoC: ux500_pcm: Stop pretending that we support varying address widths The Slave Config's addr_width attribute is populated by data_width of dma_cfg, which in turn is derived from dma_params' data_size attribute and that comes from the slot_width which is always 16 bits (2 Bytes). We're cutting out the middle man here and just setting the DMA Slave Config directly. Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_pcm.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index ce554de5d9d..32d45723211 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -109,20 +109,19 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ux500_msp_dma_params *dma_params; - struct stedma40_chan_cfg *dma_cfg; int ret; dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - dma_cfg = dma_params->dma_cfg; ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); if (ret) return ret; slave_config->dst_maxburst = 4; - slave_config->dst_addr_width = dma_cfg->dst_info.data_width; slave_config->src_maxburst = 4; - slave_config->src_addr_width = dma_cfg->src_info.data_width; + + slave_config->src_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) slave_config->dst_addr = dma_params->tx_rx_addr; -- cgit v1.2.3-70-g09d2 From f6c377520c26297cc870173df3cd0acdef08bc1c Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 19 Dec 2013 15:55:03 +0000 Subject: ASoC: ux500_pcm: Expect different saved DMA data when obtaining from DAI store In preparation for full Device Tree enablement we must differentiate between the two varying ways DMA data can be held in the DAI store. If we're booting with Device Tree the provided 'snd_dmaengine_dai_dma_data' data structure shall be used, whereas in order to avoid breaking legacy platform data we also need to be able to translate DMA data stored using the UX500 specific 'ux500_msp_dma_params' method. Once we move over to solely use Device Tree, we can enforce the use of 'snd_dmaengine_dai_dma_data' and this code can be removed altogether. Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_pcm.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 32d45723211..8b53f22edca 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -108,10 +108,21 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct dma_slave_config *slave_config) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct ux500_msp_dma_params *dma_params; + struct msp_i2s_platform_data *pdata = rtd->cpu_dai->dev->platform_data; + struct snd_dmaengine_dai_dma_data *snd_dma_params; + struct ux500_msp_dma_params *ste_dma_params; + dma_addr_t dma_addr; int ret; - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (pdata) { + ste_dma_params = + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_addr = ste_dma_params->tx_rx_addr; + } else { + snd_dma_params = + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_addr = snd_dma_params->addr; + } ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); if (ret) @@ -124,9 +135,9 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - slave_config->dst_addr = dma_params->tx_rx_addr; + slave_config->dst_addr = dma_addr; else - slave_config->src_addr = dma_params->tx_rx_addr; + slave_config->src_addr = dma_addr; return 0; } -- cgit v1.2.3-70-g09d2 From 05c56c24137273de3460872b6121a2bd762d11e8 Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 19 Dec 2013 15:55:04 +0000 Subject: ASoC: ux500_pcm: Extend Device Tree support to deal with DMA data Soon we will strip out pdata support from the Ux500 set of ASoC drivers. When this happens it will have to supply a DMA slave_config to the dmaengine. At the moment a great deal of this comes from pdata via AUXDATA. We need to become independent of this soon. This patch starts the process by allocating memory for the associated data structures and fetches the MSP id used for const struct indexing. Acked-by: Linus Walleij Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_i2s.c | 56 ++++++++++++++++++++++++++++++----------- 1 file changed, 41 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 1ca8b08ae99..7f2a4acddcd 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -646,6 +646,34 @@ int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) } +int ux500_msp_i2s_of_init_msp(struct platform_device *pdev, + struct ux500_msp *msp, + struct msp_i2s_platform_data **platform_data) +{ + struct msp_i2s_platform_data *pdata; + + *platform_data = devm_kzalloc(&pdev->dev, + sizeof(struct msp_i2s_platform_data), + GFP_KERNEL); + pdata = *platform_data; + if (!pdata) + return -ENOMEM; + + msp->playback_dma_data.dma_cfg = devm_kzalloc(&pdev->dev, + sizeof(struct stedma40_chan_cfg), + GFP_KERNEL); + if (!msp->playback_dma_data.dma_cfg) + return -ENOMEM; + + msp->capture_dma_data.dma_cfg = devm_kzalloc(&pdev->dev, + sizeof(struct stedma40_chan_cfg), + GFP_KERNEL); + if (!msp->capture_dma_data.dma_cfg) + return -ENOMEM; + + return 0; +} + int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct ux500_msp **msp_p, struct msp_i2s_platform_data *platform_data) @@ -653,30 +681,28 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct resource *res = NULL; struct device_node *np = pdev->dev.of_node; struct ux500_msp *msp; + int ret; *msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL); msp = *msp_p; if (!msp) return -ENOMEM; - if (np) { - if (!platform_data) { - platform_data = devm_kzalloc(&pdev->dev, - sizeof(struct msp_i2s_platform_data), GFP_KERNEL); - if (!platform_data) - return -ENOMEM; - } - } else - if (!platform_data) + if (!platform_data) { + if (np) { + ret = ux500_msp_i2s_of_init_msp(pdev, msp, + &platform_data); + if (ret) + return ret; + } else return -EINVAL; + } else { + msp->playback_dma_data.dma_cfg = platform_data->msp_i2s_dma_tx; + msp->capture_dma_data.dma_cfg = platform_data->msp_i2s_dma_rx; + msp->id = platform_data->id; + } - dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, - pdev->name, platform_data->id); - - msp->id = platform_data->id; msp->dev = &pdev->dev; - msp->playback_dma_data.dma_cfg = platform_data->msp_i2s_dma_tx; - msp->capture_dma_data.dma_cfg = platform_data->msp_i2s_dma_rx; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res == NULL) { -- cgit v1.2.3-70-g09d2 From f382acbe163a6faebd7cafe57800306970e241d4 Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 19 Dec 2013 15:55:05 +0000 Subject: ASoC: ux500: Store DMA data in the DAI differently in the pdata and DT case In this patch we do two things. Firstly, instead of open coding the store of DMA data in to the DAI for later use, we use the API provided. Secondly we create and store similar DMA data for the DT case, only this time we use 'struct snd_dmaengine_dai_dma_data' which is provided by the core for this very reason. Acked-by: Linus Walleij Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.c | 42 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 40 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index bc042cce115..f4d607a7266 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -17,12 +17,14 @@ #include #include #include +#include #include #include #include #include #include +#include #include "ux500_msp_i2s.h" #include "ux500_msp_dai.h" @@ -654,16 +656,52 @@ static int ux500_msp_dai_trigger(struct snd_pcm_substream *substream, return ret; } +static int ux500_msp_dai_of_probe(struct snd_soc_dai *dai) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct snd_dmaengine_dai_dma_data *playback_dma_data; + struct snd_dmaengine_dai_dma_data *capture_dma_data; + + playback_dma_data = devm_kzalloc(dai->dev, + sizeof(*playback_dma_data), + GFP_KERNEL); + if (!playback_dma_data) + return -ENOMEM; + + capture_dma_data = devm_kzalloc(dai->dev, + sizeof(*capture_dma_data), + GFP_KERNEL); + if (!capture_dma_data) + return -ENOMEM; + + playback_dma_data->addr = drvdata->msp->playback_dma_data.tx_rx_addr; + capture_dma_data->addr = drvdata->msp->capture_dma_data.tx_rx_addr; + + playback_dma_data->maxburst = 4; + capture_dma_data->maxburst = 4; + + snd_soc_dai_init_dma_data(dai, playback_dma_data, capture_dma_data); + + return 0; +} + static int ux500_msp_dai_probe(struct snd_soc_dai *dai) { struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct msp_i2s_platform_data *pdata = dai->dev->platform_data; + int ret; - dai->playback_dma_data = &drvdata->msp->playback_dma_data; - dai->capture_dma_data = &drvdata->msp->capture_dma_data; + if (!pdata) { + ret = ux500_msp_dai_of_probe(dai); + return ret; + } drvdata->msp->playback_dma_data.data_size = drvdata->slot_width; drvdata->msp->capture_dma_data.data_size = drvdata->slot_width; + snd_soc_dai_init_dma_data(dai, + &drvdata->msp->playback_dma_data, + &drvdata->msp->capture_dma_data); return 0; } -- cgit v1.2.3-70-g09d2 From ead20611a212db8ab4392cfc28092c9c849c69a4 Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 19 Dec 2013 15:55:06 +0000 Subject: ASoC: ux500_pcm: Take out pointless dev_dbg() call Acked-by: Linus Walleij Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_pcm.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 8b53f22edca..3d1c342245f 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -65,14 +65,10 @@ static struct dma_chan *ux500_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_substream *substream) { struct snd_soc_dai *dai = rtd->cpu_dai; - struct device *dev = dai->dev; u16 per_data_width, mem_data_width; struct stedma40_chan_cfg *dma_cfg; struct ux500_msp_dma_params *dma_params; - dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, - snd_pcm_stream_str(substream)); - dma_params = snd_soc_dai_get_dma_data(dai, substream); dma_cfg = dma_params->dma_cfg; -- cgit v1.2.3-70-g09d2 From 86a3fdfc63402ffbcee226c4a2503eee14a41afe Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 19 Dec 2013 15:55:07 +0000 Subject: ASoC: ux500_pcm: Differentiate between pdata and DT initialisation If booting with full DT support (i.e. DMA too, the last piece of the puzzle), then we don't need to use the compatible_request_channel call back or require some of the historical bumph which probably isn't required by a platform data start-up now either. This will also be ripped out in upcoming commits. Acked-by: Linus Walleij Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_pcm.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 3d1c342245f..55a8634cc3d 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -145,15 +145,25 @@ static const struct snd_dmaengine_pcm_config ux500_dmaengine_pcm_config = { .prepare_slave_config = ux500_pcm_prepare_slave_config, }; +static const struct snd_dmaengine_pcm_config ux500_dmaengine_of_pcm_config = { + .compat_request_channel = ux500_pcm_request_chan, + .prepare_slave_config = ux500_pcm_prepare_slave_config, +}; + int ux500_pcm_register_platform(struct platform_device *pdev) { + const struct snd_dmaengine_pcm_config *pcm_config; + struct device_node *np = pdev->dev.of_node; int ret; - ret = snd_dmaengine_pcm_register(&pdev->dev, - &ux500_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_COMPAT | - SND_DMAENGINE_PCM_FLAG_NO_DT); + if (np) + pcm_config = &ux500_dmaengine_of_pcm_config; + else + pcm_config = &ux500_dmaengine_pcm_config; + + ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_COMPAT); if (ret < 0) { dev_err(&pdev->dev, "%s: ERROR: Failed to register platform '%s' (%d)!\n", -- cgit v1.2.3-70-g09d2 From 33899b19851db3d5baf1bcde49fe90cd5f68c82c Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 19 Dec 2013 15:55:08 +0000 Subject: ASoC: ux500: Dramatically reduce the size of the DAI driver data struct We no longer have a means to differentiate between MSP devices at probe time, mainly because we don't really have to. So rather than have an over- sized static data structure in place, where the only difference between devices is the ID and name (which are unused), we'll just create one succinct, statically assigned and shared one instead. Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.c | 96 ++++++----------------------------------- 1 file changed, 14 insertions(+), 82 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index f4d607a7266..5f4807b2c00 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -718,87 +718,19 @@ static struct snd_soc_dai_ops ux500_msp_dai_ops[] = { } }; -static struct snd_soc_dai_driver ux500_msp_dai_drv[UX500_NBR_OF_DAI] = { - { - .name = "ux500-msp-i2s.0", - .probe = ux500_msp_dai_probe, - .id = 0, - .suspend = NULL, - .resume = NULL, - .playback = { - .channels_min = UX500_MSP_MIN_CHANNELS, - .channels_max = UX500_MSP_MAX_CHANNELS, - .rates = UX500_I2S_RATES, - .formats = UX500_I2S_FORMATS, - }, - .capture = { - .channels_min = UX500_MSP_MIN_CHANNELS, - .channels_max = UX500_MSP_MAX_CHANNELS, - .rates = UX500_I2S_RATES, - .formats = UX500_I2S_FORMATS, - }, - .ops = ux500_msp_dai_ops, - }, - { - .name = "ux500-msp-i2s.1", - .probe = ux500_msp_dai_probe, - .id = 1, - .suspend = NULL, - .resume = NULL, - .playback = { - .channels_min = UX500_MSP_MIN_CHANNELS, - .channels_max = UX500_MSP_MAX_CHANNELS, - .rates = UX500_I2S_RATES, - .formats = UX500_I2S_FORMATS, - }, - .capture = { - .channels_min = UX500_MSP_MIN_CHANNELS, - .channels_max = UX500_MSP_MAX_CHANNELS, - .rates = UX500_I2S_RATES, - .formats = UX500_I2S_FORMATS, - }, - .ops = ux500_msp_dai_ops, - }, - { - .name = "ux500-msp-i2s.2", - .id = 2, - .probe = ux500_msp_dai_probe, - .suspend = NULL, - .resume = NULL, - .playback = { - .channels_min = UX500_MSP_MIN_CHANNELS, - .channels_max = UX500_MSP_MAX_CHANNELS, - .rates = UX500_I2S_RATES, - .formats = UX500_I2S_FORMATS, - }, - .capture = { - .channels_min = UX500_MSP_MIN_CHANNELS, - .channels_max = UX500_MSP_MAX_CHANNELS, - .rates = UX500_I2S_RATES, - .formats = UX500_I2S_FORMATS, - }, - .ops = ux500_msp_dai_ops, - }, - { - .name = "ux500-msp-i2s.3", - .probe = ux500_msp_dai_probe, - .id = 3, - .suspend = NULL, - .resume = NULL, - .playback = { - .channels_min = UX500_MSP_MIN_CHANNELS, - .channels_max = UX500_MSP_MAX_CHANNELS, - .rates = UX500_I2S_RATES, - .formats = UX500_I2S_FORMATS, - }, - .capture = { - .channels_min = UX500_MSP_MIN_CHANNELS, - .channels_max = UX500_MSP_MAX_CHANNELS, - .rates = UX500_I2S_RATES, - .formats = UX500_I2S_FORMATS, - }, - .ops = ux500_msp_dai_ops, - }, +static struct snd_soc_dai_driver ux500_msp_dai_drv = { + .probe = ux500_msp_dai_probe, + .suspend = NULL, + .resume = NULL, + .playback.channels_min = UX500_MSP_MIN_CHANNELS, + .playback.channels_max = UX500_MSP_MAX_CHANNELS, + .playback.rates = UX500_I2S_RATES, + .playback.formats = UX500_I2S_FORMATS, + .capture.channels_min = UX500_MSP_MIN_CHANNELS, + .capture.channels_max = UX500_MSP_MAX_CHANNELS, + .capture.rates = UX500_I2S_RATES, + .capture.formats = UX500_I2S_FORMATS, + .ops = ux500_msp_dai_ops, }; static const struct snd_soc_component_driver ux500_msp_component = { @@ -868,7 +800,7 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, drvdata); ret = snd_soc_register_component(&pdev->dev, &ux500_msp_component, - &ux500_msp_dai_drv[drvdata->msp->id], 1); + &ux500_msp_dai_drv, 1); if (ret < 0) { dev_err(&pdev->dev, "Error: %s: Failed to register MSP%d!\n", __func__, drvdata->msp->id); -- cgit v1.2.3-70-g09d2 From f87a3e825cb0f7d4d51556ece147f1a6299ac1af Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 7 Jan 2014 09:13:42 +0800 Subject: ASoC: simple-card: fix the DAPM routes map parsing The simple-card's DAPM route maping is optional. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d4402fb5725..eb95beb25d4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -134,10 +134,12 @@ static int asoc_simple_card_parse_of(struct device_node *node, (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* DAPM routes */ - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + if (of_property_read_bool(node, "simple-audio-routing")) { + ret = snd_soc_of_parse_audio_routing(&info->snd_card, "simple-audio-routing"); - if (ret) - return ret; + if (ret) + return ret; + } /* CPU sub-node */ ret = -EINVAL; -- cgit v1.2.3-70-g09d2 From 8c0b8230b2d9708eed5b50f9f8442aaa879a3c57 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 7 Jan 2014 09:15:16 +0800 Subject: ASoC: simple-card: keep the property's name the same pattern Even though we might not have rigor rule for the simple card property names, according to the existing ones, they are all in a same pattern: [simple-audio-card,]XXX; Rename simple-audio-routing to simple-audio-card,routing, and make the simple card's properties has one unified name. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/simple-card.txt | 2 +- sound/soc/generic/simple-card.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 2ee80c76ca6..e9e20ec67d6 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -11,7 +11,7 @@ Optional properties: - simple-audio-card,format : CPU/CODEC common audio format. "i2s", "right_j", "left_j" , "dsp_a" "dsp_b", "ac97", "pdm", "msb", "lsb" -- simple-audio-routing : A list of the connections between audio components. +- simple-audio-card,routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index eb95beb25d4..0430be85f23 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -134,9 +134,9 @@ static int asoc_simple_card_parse_of(struct device_node *node, (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* DAPM routes */ - if (of_property_read_bool(node, "simple-audio-routing")) { + if (of_property_read_bool(node, "simple-audio-card,routing")) { ret = snd_soc_of_parse_audio_routing(&info->snd_card, - "simple-audio-routing"); + "simple-audio-card,routing"); if (ret) return ret; } -- cgit v1.2.3-70-g09d2 From dcf0fa27a56025793a700e81edd261ee3369e294 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Fri, 3 Jan 2014 09:19:18 +0100 Subject: ASoC: pcm: Fix lack of platform bespoke_trigger() call When the platform driver has no ops, the platform function bespoke_trigger() is no more called. The problem was introduced by the commit c5914b0aaea6494aaa9e415cbd32f8b7eb604af0 "ASoC: pcm: Check for ops before deferencing them" Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index feb0f284302..d70eecd9e16 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -769,7 +769,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, return ret; } - if (platform->driver->ops && platform->driver->bespoke_trigger) { + if (platform->driver->bespoke_trigger) { ret = platform->driver->bespoke_trigger(substream, cmd); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From 1e9de42f4324b91ce2e9da0939cab8fc6ae93893 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 7 Jan 2014 17:51:42 +0000 Subject: ASoC: dpcm: Explicitly set BE DAI link supported stream directions Some BE DAIs can be "dummy" (when the DSP is controlling the DAI) and as such wont have set a minimum number of playback or capture channels required for BE DAI registration (to establish supported stream directions). Force machine drivers to explicitly set whether they support playback and capture stream directions for every BE DAIs. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++++ sound/soc/soc-pcm.c | 6 ++---- 2 files changed, 6 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 1f741cb24f3..a5ef14f0612 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -886,6 +886,10 @@ struct snd_soc_dai_link { /* This DAI link can route to other DAI links at runtime (Frontend)*/ unsigned int dynamic:1; + /* DPCM capture and Playback support */ + unsigned int dpcm_capture:1; + unsigned int dpcm_playback:1; + /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 42782c01e41..141a302e4e7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2026,10 +2026,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int ret = 0, playback = 0, capture = 0; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - if (cpu_dai->driver->playback.channels_min) - playback = 1; - if (cpu_dai->driver->capture.channels_min) - capture = 1; + playback = rtd->dai_link->dpcm_playback; + capture = rtd->dai_link->dpcm_capture; } else { if (codec_dai->driver->playback.channels_min && cpu_dai->driver->playback.channels_min) -- cgit v1.2.3-70-g09d2 From bece9e957cbfb37f12488b24166364307e39f5b0 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 8 Jan 2014 10:40:18 +0000 Subject: ASoC: utils: Add internal call to determine if DAI is dummy. Provide a quick way to tell if a DAI is a dummy DAI or a regular DAI. This is for internal DAPM usage only and is used to determine whether to insert a DAI link connection into the DAPM graph. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-utils.c | 7 +++++++ 2 files changed, 9 insertions(+) (limited to 'sound/soc') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 800c101bb09..c42864b3458 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -123,6 +123,8 @@ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction); +int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); + struct snd_soc_dai_ops { /* * DAI clocking configuration, all optional. diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 5e633659c1b..d14bdb3c52d 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -123,6 +123,13 @@ static struct snd_soc_dai_driver dummy_dai = { }, }; +int snd_soc_dai_is_dummy(struct snd_soc_dai *dai) +{ + if (dai->driver == &dummy_dai) + return 1; + return 0; +} + static int snd_soc_dummy_probe(struct platform_device *pdev) { int ret; -- cgit v1.2.3-70-g09d2 From b893ea5f1cd1adbbd7e0794d16d47bbb46f80733 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 8 Jan 2014 10:40:19 +0000 Subject: ASoC: sapm: Automatically connect DAI link widgets in DAPM graph. Connect the DAPM graph through each BE DAI link to the componnent(s) on the other side of the BE DAI link. This allows the graph to be walked on both sides of the link when graph changes are made. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 49 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 51 insertions(+) (limited to 'sound/soc') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 2037c45adfe..a5de124d2f9 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -411,6 +411,7 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, struct snd_soc_dai *dai); int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); +void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, struct snd_soc_dapm_widget *source, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e53d87e881..7d9c0660ab2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1728,6 +1728,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } snd_soc_dapm_link_dai_widgets(card); + snd_soc_dapm_connect_dai_link_widgets(card); if (card->controls) snd_soc_add_card_controls(card, card->controls, card->num_controls); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 67e63ab1f11..51b4c192f41 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3634,6 +3634,55 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } +void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd = card->rtd; + struct snd_soc_dai *cpu_dai, *codec_dai; + struct snd_soc_dapm_route r; + int i; + + memset(&r, 0, sizeof(r)); + + /* for each BE DAI link... */ + for (i = 0; i < card->num_rtd; i++) { + rtd = &card->rtd[i]; + cpu_dai = rtd->cpu_dai; + codec_dai = rtd->codec_dai; + + /* dynamic FE links have no fixed DAI mapping */ + if (rtd->dai_link->dynamic) + continue; + + /* there is no point in connecting BE DAI links with dummies */ + if (snd_soc_dai_is_dummy(codec_dai) || + snd_soc_dai_is_dummy(cpu_dai)) + continue; + + /* connect BE DAI playback if widgets are valid */ + if (codec_dai->playback_widget && cpu_dai->playback_widget) { + r.source = cpu_dai->playback_widget->name; + r.sink = codec_dai->playback_widget->name; + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", + cpu_dai->codec->name, r.source, + codec_dai->platform->name, r.sink); + + snd_soc_dapm_add_route(&card->dapm, &r); + } + + /* connect BE DAI capture if widgets are valid */ + if (codec_dai->capture_widget && cpu_dai->capture_widget) { + r.source = codec_dai->capture_widget->name; + r.sink = cpu_dai->capture_widget->name; + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", + codec_dai->codec->name, r.source, + cpu_dai->platform->name, r.sink); + + snd_soc_dapm_add_route(&card->dapm, &r); + } + + } +} + static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { -- cgit v1.2.3-70-g09d2 From 7ee4518ab75164533e282eb8f2827a74920a2a19 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Wed, 8 Jan 2014 18:13:35 +0800 Subject: ASoC: ux500: Fix sparse non static symbol warning Fixes the following sparse warning: sound/soc/ux500/ux500_msp_i2s.c:649:5: warning: symbol 'ux500_msp_i2s_of_init_msp' was not declared. Should it be static? Signed-off-by: Wei Yongjun Acked-by: Arnd Bergmann Acked-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 7f2a4acddcd..959d7b4edf5 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -646,9 +646,9 @@ int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) } -int ux500_msp_i2s_of_init_msp(struct platform_device *pdev, - struct ux500_msp *msp, - struct msp_i2s_platform_data **platform_data) +static int ux500_msp_i2s_of_init_msp(struct platform_device *pdev, + struct ux500_msp *msp, + struct msp_i2s_platform_data **platform_data) { struct msp_i2s_platform_data *pdata; -- cgit v1.2.3-70-g09d2 From 633ff8f8a4393b4a13b94eddd2613198c32035e6 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Wed, 8 Jan 2014 16:13:05 +0800 Subject: ASoC: fsl-sai: Clean up the code Makes the code slightly shorter. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5d38a6749b9..cdd3fa83070 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -62,26 +62,25 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, reg_cr2 = FSL_SAI_RCR2; val_cr2 = sai_readl(sai, sai->base + reg_cr2); + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + switch (clk_id) { case FSL_SAI_CLK_BUS: - val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; val_cr2 |= FSL_SAI_CR2_MSEL_BUS; break; case FSL_SAI_CLK_MAST1: - val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; val_cr2 |= FSL_SAI_CR2_MSEL_MCLK1; break; case FSL_SAI_CLK_MAST2: - val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; val_cr2 |= FSL_SAI_CR2_MSEL_MCLK2; break; case FSL_SAI_CLK_MAST3: - val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; val_cr2 |= FSL_SAI_CR2_MSEL_MCLK3; break; default: return -EINVAL; } + sai_writel(sai, val_cr2, sai->base + reg_cr2); return 0; -- cgit v1.2.3-70-g09d2 From a8fc415c29a62e4f0a7a932110ee9d8423e2cc52 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 Jan 2014 15:27:49 +0200 Subject: ASoC: twl4030: Separate write condition checking from I/O function Simplifies the code a bit and prepares it to the removal of local caching. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 40 +++++++++++++++++++++++----------------- 1 file changed, 23 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index dfc51bb425d..419108ae31d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -181,50 +181,56 @@ static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec, cache[reg] = value; } -/* - * write to the twl4030 register space - */ -static int twl4030_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec, + unsigned int reg) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int write_to_reg = 0; + bool write_to_reg = false; - twl4030_write_reg_cache(codec, reg, value); /* Decide if the given register can be written */ switch (reg) { case TWL4030_REG_EAR_CTL: if (twl4030->earpiece_enabled) - write_to_reg = 1; + write_to_reg = true; break; case TWL4030_REG_PREDL_CTL: if (twl4030->predrivel_enabled) - write_to_reg = 1; + write_to_reg = true; break; case TWL4030_REG_PREDR_CTL: if (twl4030->predriver_enabled) - write_to_reg = 1; + write_to_reg = true; break; case TWL4030_REG_PRECKL_CTL: if (twl4030->carkitl_enabled) - write_to_reg = 1; + write_to_reg = true; break; case TWL4030_REG_PRECKR_CTL: if (twl4030->carkitr_enabled) - write_to_reg = 1; + write_to_reg = true; break; case TWL4030_REG_HS_GAIN_SET: if (twl4030->hsl_enabled || twl4030->hsr_enabled) - write_to_reg = 1; + write_to_reg = true; break; default: /* All other register can be written */ - write_to_reg = 1; + write_to_reg = true; break; } - if (write_to_reg) - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - value, reg); + + return write_to_reg; +} + +/* + * write to the twl4030 register space + */ +static int twl4030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + twl4030_write_reg_cache(codec, reg, value); + if (twl4030_can_write_to_chip(codec, reg)) + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); return 0; } -- cgit v1.2.3-70-g09d2 From 7bfbdfea576e3ae109fa182519b6f004c6024952 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 Jan 2014 15:27:50 +0200 Subject: ASoC: twl4030: Remove check defaults functionality No need to keep the check defaults functionality anymore. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- include/linux/i2c/twl.h | 1 - sound/soc/codecs/twl4030.c | 23 ----------------------- 2 files changed, 24 deletions(-) (limited to 'sound/soc') diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index a09da091033..2937a9472b9 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -670,7 +670,6 @@ struct twl4030_codec_data { unsigned int digimic_delay; /* in ms */ unsigned int ramp_delay_value; unsigned int offset_cncl_path; - unsigned int check_defaults:1; unsigned int reset_registers:1; unsigned int hs_extmute:1; int hs_extmute_gpio; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 419108ae31d..7b732ab70d2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -268,25 +268,6 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) udelay(10); } -static inline void twl4030_check_defaults(struct snd_soc_codec *codec) -{ - int i, difference = 0; - u8 val; - - dev_dbg(codec->dev, "Checking TWL audio default configuration\n"); - for (i = 1; i <= TWL4030_REG_MISC_SET_2; i++) { - twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &val, i); - if (val != twl4030_reg[i]) { - difference++; - dev_dbg(codec->dev, - "Reg 0x%02x: chip: 0x%02x driver: 0x%02x\n", - i, val, twl4030_reg[i]); - } - } - dev_dbg(codec->dev, "Found %d non-matching registers. %s\n", - difference, difference ? "Not OK" : "OK"); -} - static inline void twl4030_reset_registers(struct snd_soc_codec *codec) { int i; @@ -378,10 +359,6 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } } - /* Check defaults, if instructed before anything else */ - if (pdata && pdata->check_defaults) - twl4030_check_defaults(codec); - /* Reset registers, if no setup data or if instructed to do so */ if (!pdata || (pdata && pdata->reset_registers)) twl4030_reset_registers(codec); -- cgit v1.2.3-70-g09d2 From 0dc41562a44c9e1012bb810c2a84e81c425867b0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 Jan 2014 15:27:51 +0200 Subject: ASoC: twl4030: Remove reset registers functionality The register states now tracked by the regmap implementation in the core which makes the reset registers functionality 'redundant' since we know the state of the registers now all the time. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- include/linux/i2c/twl.h | 1 - sound/soc/codecs/twl4030.c | 17 ----------------- 2 files changed, 18 deletions(-) (limited to 'sound/soc') diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index 2937a9472b9..ade1c06d4ce 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -670,7 +670,6 @@ struct twl4030_codec_data { unsigned int digimic_delay; /* in ms */ unsigned int ramp_delay_value; unsigned int offset_cncl_path; - unsigned int reset_registers:1; unsigned int hs_extmute:1; int hs_extmute_gpio; }; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 7b732ab70d2..ab2f22299db 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -268,17 +268,6 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) udelay(10); } -static inline void twl4030_reset_registers(struct snd_soc_codec *codec) -{ - int i; - - /* set all audio section registers to reasonable defaults */ - for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) - if (i != TWL4030_REG_APLL_CTL) - twl4030_write(codec, i, twl4030_reg[i]); - -} - static void twl4030_setup_pdata_of(struct twl4030_codec_data *pdata, struct device_node *node) { @@ -359,10 +348,6 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } } - /* Reset registers, if no setup data or if instructed to do so */ - if (!pdata || (pdata && pdata->reset_registers)) - twl4030_reset_registers(codec); - /* Refresh APLL_CTL register from HW */ twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, TWL4030_REG_APLL_CTL); @@ -2293,8 +2278,6 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec) struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); struct twl4030_codec_data *pdata = twl4030->pdata; - /* Reset registers to their chip default before leaving */ - twl4030_reset_registers(codec); twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); if (pdata && pdata->hs_extmute && gpio_is_valid(pdata->hs_extmute_gpio)) -- cgit v1.2.3-70-g09d2 From 8b3bca2966985f559f9ace1effc98955006f2b05 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 Jan 2014 15:27:52 +0200 Subject: ASoC: twl4030: Introduce local ctl register cache Few registers need to be cached in the codec driver level. These registers should only be written when the path is active to avoid pop noise on the given path. This patch adds an array which covers the range where the sensitive registers are located and uppon loadinf the driver the ctl cache will be initialized. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ab2f22299db..f88207712d3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -150,10 +150,22 @@ struct twl4030_priv { u8 earpiece_enabled; u8 predrivel_enabled, predriver_enabled; u8 carkitl_enabled, carkitr_enabled; + u8 ctl_cache[TWL4030_REG_PRECKR_CTL - TWL4030_REG_EAR_CTL + 1]; struct twl4030_codec_data *pdata; }; +static void tw4030_init_ctl_cache(struct twl4030_priv *twl4030) +{ + int i; + u8 byte; + + for (i = TWL4030_REG_EAR_CTL; i <= TWL4030_REG_PRECKR_CTL; i++) { + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, i); + twl4030->ctl_cache[i - TWL4030_REG_EAR_CTL] = byte; + } +} + /* * read twl4030 register cache */ @@ -348,6 +360,9 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } } + /* Initialize the local ctl register cache */ + tw4030_init_ctl_cache(twl4030); + /* Refresh APLL_CTL register from HW */ twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, TWL4030_REG_APLL_CTL); -- cgit v1.2.3-70-g09d2 From efc8acff1ffe18b981d70da7ab2525e5b3e5de85 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 Jan 2014 15:27:53 +0200 Subject: ASoC: twl4030: Remove local reg cache Depend on the regmap reg cache implementation for register caching done in the twl-core driver. The local register cache can be removed and we can keep only shadow copies of certain ctl registers for pop noise reduction. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 207 ++++++++++++++------------------------------- 1 file changed, 63 insertions(+), 144 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f88207712d3..dda53e8c51e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -48,86 +48,6 @@ #define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) -/* - * twl4030 register cache & default register settings - */ -static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { - 0x00, /* this register not used */ - 0x00, /* REG_CODEC_MODE (0x1) */ - 0x00, /* REG_OPTION (0x2) */ - 0x00, /* REG_UNKNOWN (0x3) */ - 0x00, /* REG_MICBIAS_CTL (0x4) */ - 0x00, /* REG_ANAMICL (0x5) */ - 0x00, /* REG_ANAMICR (0x6) */ - 0x00, /* REG_AVADC_CTL (0x7) */ - 0x00, /* REG_ADCMICSEL (0x8) */ - 0x00, /* REG_DIGMIXING (0x9) */ - 0x0f, /* REG_ATXL1PGA (0xA) */ - 0x0f, /* REG_ATXR1PGA (0xB) */ - 0x0f, /* REG_AVTXL2PGA (0xC) */ - 0x0f, /* REG_AVTXR2PGA (0xD) */ - 0x00, /* REG_AUDIO_IF (0xE) */ - 0x00, /* REG_VOICE_IF (0xF) */ - 0x3f, /* REG_ARXR1PGA (0x10) */ - 0x3f, /* REG_ARXL1PGA (0x11) */ - 0x3f, /* REG_ARXR2PGA (0x12) */ - 0x3f, /* REG_ARXL2PGA (0x13) */ - 0x25, /* REG_VRXPGA (0x14) */ - 0x00, /* REG_VSTPGA (0x15) */ - 0x00, /* REG_VRX2ARXPGA (0x16) */ - 0x00, /* REG_AVDAC_CTL (0x17) */ - 0x00, /* REG_ARX2VTXPGA (0x18) */ - 0x32, /* REG_ARXL1_APGA_CTL (0x19) */ - 0x32, /* REG_ARXR1_APGA_CTL (0x1A) */ - 0x32, /* REG_ARXL2_APGA_CTL (0x1B) */ - 0x32, /* REG_ARXR2_APGA_CTL (0x1C) */ - 0x00, /* REG_ATX2ARXPGA (0x1D) */ - 0x00, /* REG_BT_IF (0x1E) */ - 0x55, /* REG_BTPGA (0x1F) */ - 0x00, /* REG_BTSTPGA (0x20) */ - 0x00, /* REG_EAR_CTL (0x21) */ - 0x00, /* REG_HS_SEL (0x22) */ - 0x00, /* REG_HS_GAIN_SET (0x23) */ - 0x00, /* REG_HS_POPN_SET (0x24) */ - 0x00, /* REG_PREDL_CTL (0x25) */ - 0x00, /* REG_PREDR_CTL (0x26) */ - 0x00, /* REG_PRECKL_CTL (0x27) */ - 0x00, /* REG_PRECKR_CTL (0x28) */ - 0x00, /* REG_HFL_CTL (0x29) */ - 0x00, /* REG_HFR_CTL (0x2A) */ - 0x05, /* REG_ALC_CTL (0x2B) */ - 0x00, /* REG_ALC_SET1 (0x2C) */ - 0x00, /* REG_ALC_SET2 (0x2D) */ - 0x00, /* REG_BOOST_CTL (0x2E) */ - 0x00, /* REG_SOFTVOL_CTL (0x2F) */ - 0x13, /* REG_DTMF_FREQSEL (0x30) */ - 0x00, /* REG_DTMF_TONEXT1H (0x31) */ - 0x00, /* REG_DTMF_TONEXT1L (0x32) */ - 0x00, /* REG_DTMF_TONEXT2H (0x33) */ - 0x00, /* REG_DTMF_TONEXT2L (0x34) */ - 0x79, /* REG_DTMF_TONOFF (0x35) */ - 0x11, /* REG_DTMF_WANONOFF (0x36) */ - 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */ - 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */ - 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */ - 0x06, /* REG_APLL_CTL (0x3A) */ - 0x00, /* REG_DTMF_CTL (0x3B) */ - 0x44, /* REG_DTMF_PGA_CTL2 (0x3C) */ - 0x69, /* REG_DTMF_PGA_CTL1 (0x3D) */ - 0x00, /* REG_MISC_SET_1 (0x3E) */ - 0x00, /* REG_PCMBTMUX (0x3F) */ - 0x00, /* not used (0x40) */ - 0x00, /* not used (0x41) */ - 0x00, /* not used (0x42) */ - 0x00, /* REG_RX_PATH_SEL (0x43) */ - 0x32, /* REG_VDL_APGA_CTL (0x44) */ - 0x00, /* REG_VIBRA_CTL (0x45) */ - 0x00, /* REG_VIBRA_SET (0x46) */ - 0x00, /* REG_VIBRA_PWM_SET (0x47) */ - 0x00, /* REG_ANAMIC_GAIN (0x48) */ - 0x00, /* REG_MISC_SET_2 (0x49) */ -}; - /* codec private data */ struct twl4030_priv { unsigned int codec_powered; @@ -166,31 +86,48 @@ static void tw4030_init_ctl_cache(struct twl4030_priv *twl4030) } } -/* - * read twl4030 register cache - */ -static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) +static void twl4030_update_ctl_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) { - u8 *cache = codec->reg_cache; - - if (reg >= TWL4030_CACHEREGNUM) - return -EIO; + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - return cache[reg]; + switch (reg) { + case TWL4030_REG_EAR_CTL: + case TWL4030_REG_PREDL_CTL: + case TWL4030_REG_PREDR_CTL: + case TWL4030_REG_PRECKL_CTL: + case TWL4030_REG_PRECKR_CTL: + case TWL4030_REG_HS_GAIN_SET: + twl4030->ctl_cache[reg - TWL4030_REG_EAR_CTL] = value; + break; + default: + break; + } } -/* - * write twl4030 register cache - */ -static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec, - u8 reg, u8 value) +static unsigned int twl4030_read(struct snd_soc_codec *codec, unsigned int reg) { - u8 *cache = codec->reg_cache; + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); + u8 value = 0; if (reg >= TWL4030_CACHEREGNUM) - return; - cache[reg] = value; + return -EIO; + + switch (reg) { + case TWL4030_REG_EAR_CTL: + case TWL4030_REG_PREDL_CTL: + case TWL4030_REG_PREDR_CTL: + case TWL4030_REG_PRECKL_CTL: + case TWL4030_REG_PRECKR_CTL: + case TWL4030_REG_HS_GAIN_SET: + value = twl4030->ctl_cache[reg - TWL4030_REG_EAR_CTL]; + break; + default: + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + break; + } + + return value; } static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec, @@ -234,13 +171,10 @@ static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec, return write_to_reg; } -/* - * write to the twl4030 register space - */ static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - twl4030_write_reg_cache(codec, reg, value); + twl4030_update_ctl_cache(codec, reg, value); if (twl4030_can_write_to_chip(codec, reg)) return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); @@ -270,10 +204,8 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) else mode = twl4030_audio_disable_resource(TWL4030_AUDIO_RES_POWER); - if (mode >= 0) { - twl4030_write_reg_cache(codec, TWL4030_REG_CODEC_MODE, mode); + if (mode >= 0) twl4030->codec_powered = enable; - } /* REVISIT: this delay is present in TI sample drivers */ /* but there seems to be no TRM requirement for it */ @@ -363,13 +295,8 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) /* Initialize the local ctl register cache */ tw4030_init_ctl_cache(twl4030); - /* Refresh APLL_CTL register from HW */ - twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, - TWL4030_REG_APLL_CTL); - twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, byte); - /* anti-pop when changing analog gain */ - reg = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); + reg = twl4030_read(codec, TWL4030_REG_MISC_SET_1); twl4030_write(codec, TWL4030_REG_MISC_SET_1, reg | TWL4030_SMOOTH_ANAVOL_EN); @@ -386,15 +313,15 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) twl4030->pdata = pdata; - reg = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + reg = twl4030_read(codec, TWL4030_REG_HS_POPN_SET); reg &= ~TWL4030_RAMP_DELAY; reg |= (pdata->ramp_delay_value << 2); - twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, reg); + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, reg); /* initiate offset cancellation */ twl4030_codec_enable(codec, 1); - reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + reg = twl4030_read(codec, TWL4030_REG_ANAMICL); reg &= ~TWL4030_OFFSET_CNCL_SEL; reg |= pdata->offset_cncl_path; twl4030_write(codec, TWL4030_REG_ANAMICL, @@ -408,15 +335,14 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) msleep(20); do { usleep_range(1000, 2000); + twl_set_regcache_bypass(TWL4030_MODULE_AUDIO_VOICE, true); twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, TWL4030_REG_ANAMICL); + twl_set_regcache_bypass(TWL4030_MODULE_AUDIO_VOICE, false); } while ((i++ < 100) && ((byte & TWL4030_CNCL_OFFSET_START) == TWL4030_CNCL_OFFSET_START)); - /* Make sure that the reg_cache has the same value as the HW */ - twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte); - twl4030_codec_enable(codec, 0); } @@ -436,9 +362,6 @@ static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) status = twl4030_audio_disable_resource( TWL4030_AUDIO_RES_APLL); } - - if (status >= 0) - twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); } /* Earpiece */ @@ -661,8 +584,7 @@ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ switch (event) { \ case SND_SOC_DAPM_POST_PMU: \ twl4030->pin_name##_enabled = 1; \ - twl4030_write(w->codec, reg, \ - twl4030_read_reg_cache(w->codec, reg)); \ + twl4030_write(w->codec, reg, twl4030_read(w->codec, reg)); \ break; \ case SND_SOC_DAPM_POST_PMD: \ twl4030->pin_name##_enabled = 0; \ @@ -683,7 +605,7 @@ static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp) { unsigned char hs_ctl; - hs_ctl = twl4030_read_reg_cache(codec, reg); + hs_ctl = twl4030_read(codec, reg); if (ramp) { /* HF ramp-up */ @@ -763,7 +685,7 @@ static int aif_event(struct snd_soc_dapm_widget *w, { u8 audio_if; - audio_if = twl4030_read_reg_cache(w->codec, TWL4030_REG_AUDIO_IF); + audio_if = twl4030_read(w->codec, TWL4030_REG_AUDIO_IF); switch (event) { case SND_SOC_DAPM_PRE_PMU: /* Enable AIF */ @@ -793,8 +715,8 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) 8388608, 16777216, 33554432, 67108864}; unsigned int delay; - hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET); - hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + hs_gain = twl4030_read(codec, TWL4030_REG_HS_GAIN_SET); + hs_pop = twl4030_read(codec, TWL4030_REG_HS_POPN_SET); delay = (ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] / twl4030->sysclk) + 1; @@ -1738,7 +1660,7 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, { u8 reg, mask; - reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); + reg = twl4030_read(codec, TWL4030_REG_OPTION); if (direction == SNDRV_PCM_STREAM_PLAYBACK) mask = TWL4030_ARXL1_VRX_EN | TWL4030_ARXR1_EN; @@ -1767,7 +1689,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream, if (twl4030->configured) twl4030_constraints(twl4030, twl4030->master_substream); } else { - if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & + if (!(twl4030_read(codec, TWL4030_REG_CODEC_MODE) & TWL4030_OPTION_1)) { /* In option2 4 channel is not supported, set the * constraint for the first stream for channels, the @@ -1815,8 +1737,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, /* If the substream has 4 channel, do the necessary setup */ if (params_channels(params) == 4) { - format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); - mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); + format = twl4030_read(codec, TWL4030_REG_AUDIO_IF); + mode = twl4030_read(codec, TWL4030_REG_CODEC_MODE); /* Safety check: are we in the correct operating mode and * the interface is in TDM mode? */ @@ -1832,8 +1754,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, return 0; /* bit rate */ - old_mode = twl4030_read_reg_cache(codec, - TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; + old_mode = twl4030_read(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; mode = old_mode & ~TWL4030_APLL_RATE; switch (params_rate(params)) { @@ -1874,7 +1796,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, } /* sample size */ - old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + old_format = twl4030_read(codec, TWL4030_REG_AUDIO_IF); format = old_format; format &= ~TWL4030_DATA_WIDTH; switch (params_format(params)) { @@ -1957,7 +1879,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, u8 old_format, format; /* get format */ - old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + old_format = twl4030_read(codec, TWL4030_REG_AUDIO_IF); format = old_format; /* set master/slave audio interface */ @@ -2007,7 +1929,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, static int twl4030_set_tristate(struct snd_soc_dai *dai, int tristate) { struct snd_soc_codec *codec = dai->codec; - u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + u8 reg = twl4030_read(codec, TWL4030_REG_AUDIO_IF); if (tristate) reg |= TWL4030_AIF_TRI_EN; @@ -2024,7 +1946,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, { u8 reg, mask; - reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); + reg = twl4030_read(codec, TWL4030_REG_OPTION); if (direction == SNDRV_PCM_STREAM_PLAYBACK) mask = TWL4030_ARXL1_VRX_EN; @@ -2059,7 +1981,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, /* If the codec mode is not option2, the voice PCM interface is not * available. */ - mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) + mode = twl4030_read(codec, TWL4030_REG_CODEC_MODE) & TWL4030_OPT_MODE; if (mode != TWL4030_OPTION_2) { @@ -2091,7 +2013,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, twl4030_voice_enable(codec, substream->stream, 1); /* bit rate */ - old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) + old_mode = twl4030_read(codec, TWL4030_REG_CODEC_MODE) & ~(TWL4030_CODECPDZ); mode = old_mode; @@ -2154,7 +2076,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, u8 old_format, format; /* get format */ - old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + old_format = twl4030_read(codec, TWL4030_REG_VOICE_IF); format = old_format; /* set master/slave audio interface */ @@ -2201,7 +2123,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate) { struct snd_soc_codec *codec = dai->codec; - u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + u8 reg = twl4030_read(codec, TWL4030_REG_VOICE_IF); if (tristate) reg |= TWL4030_VIF_TRI_EN; @@ -2304,13 +2226,10 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .probe = twl4030_soc_probe, .remove = twl4030_soc_remove, - .read = twl4030_read_reg_cache, + .read = twl4030_read, .write = twl4030_write, .set_bias_level = twl4030_set_bias_level, .idle_bias_off = true, - .reg_cache_size = sizeof(twl4030_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = twl4030_reg, .controls = twl4030_snd_controls, .num_controls = ARRAY_SIZE(twl4030_snd_controls), -- cgit v1.2.3-70-g09d2 From 7ded5fe020e670befeab6777e7b8bc4bec272a3f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 Jan 2014 15:27:54 +0200 Subject: ASoC: twl4030: Parameter alignment fixes (for code consistency) Over time the multi line alignment got messed up. Correct them in one go so the code will look consistent. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 93 ++++++++++++++++++++++------------------------ 1 file changed, 45 insertions(+), 48 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index dda53e8c51e..7a5b91e70f9 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -171,8 +171,8 @@ static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec, return write_to_reg; } -static int twl4030_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) { twl4030_update_ctl_cache(codec, reg, value); if (twl4030_can_write_to_chip(codec, reg)) @@ -298,11 +298,11 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) /* anti-pop when changing analog gain */ reg = twl4030_read(codec, TWL4030_REG_MISC_SET_1); twl4030_write(codec, TWL4030_REG_MISC_SET_1, - reg | TWL4030_SMOOTH_ANAVOL_EN); + reg | TWL4030_SMOOTH_ANAVOL_EN); twl4030_write(codec, TWL4030_REG_OPTION, - TWL4030_ATXL1_EN | TWL4030_ATXR1_EN | - TWL4030_ARXL2_EN | TWL4030_ARXR2_EN); + TWL4030_ATXL1_EN | TWL4030_ATXR1_EN | + TWL4030_ARXL2_EN | TWL4030_ARXR2_EN); /* REG_ARXR2_APGA_CTL reset according to the TRM: 0dB, DA_EN */ twl4030_write(codec, TWL4030_REG_ARXR2_APGA_CTL, 0x32); @@ -325,7 +325,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) reg &= ~TWL4030_OFFSET_CNCL_SEL; reg |= pdata->offset_cncl_path; twl4030_write(codec, TWL4030_REG_ANAMICL, - reg | TWL4030_CNCL_OFFSET_START); + reg | TWL4030_CNCL_OFFSET_START); /* * Wait for offset cancellation to complete. @@ -337,7 +337,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) usleep_range(1000, 2000); twl_set_regcache_bypass(TWL4030_MODULE_AUDIO_VOICE, true); twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, - TWL4030_REG_ANAMICL); + TWL4030_REG_ANAMICL); twl_set_regcache_bypass(TWL4030_MODULE_AUDIO_VOICE, false); } while ((i++ < 100) && ((byte & TWL4030_CNCL_OFFSET_START) == @@ -577,7 +577,7 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control = */ #define TWL4030_OUTPUT_PGA(pin_name, reg, mask) \ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ - struct snd_kcontrol *kcontrol, int event) \ + struct snd_kcontrol *kcontrol, int event) \ { \ struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); \ \ @@ -588,8 +588,7 @@ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ break; \ case SND_SOC_DAPM_POST_PMD: \ twl4030->pin_name##_enabled = 0; \ - twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ - 0, reg); \ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, 0, reg); \ break; \ } \ return 0; \ @@ -632,7 +631,7 @@ static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp) } static int handsfreelpga_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) + struct snd_kcontrol *kcontrol, int event) { switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -646,7 +645,7 @@ static int handsfreelpga_event(struct snd_soc_dapm_widget *w, } static int handsfreerpga_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) + struct snd_kcontrol *kcontrol, int event) { switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -660,14 +659,14 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w, } static int vibramux_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) + struct snd_kcontrol *kcontrol, int event) { twl4030_write(w->codec, TWL4030_REG_VIBRA_SET, 0xff); return 0; } static int apll_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) + struct snd_kcontrol *kcontrol, int event) { switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -681,7 +680,7 @@ static int apll_event(struct snd_soc_dapm_widget *w, } static int aif_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) + struct snd_kcontrol *kcontrol, int event) { u8 audio_if; @@ -693,12 +692,12 @@ static int aif_event(struct snd_soc_dapm_widget *w, twl4030_apll_enable(w->codec, 1); twl4030_write(w->codec, TWL4030_REG_AUDIO_IF, - audio_if | TWL4030_AIF_EN); + audio_if | TWL4030_AIF_EN); break; case SND_SOC_DAPM_POST_PMD: /* disable the DAI before we stop it's source PLL */ twl4030_write(w->codec, TWL4030_REG_AUDIO_IF, - audio_if & ~TWL4030_AIF_EN); + audio_if & ~TWL4030_AIF_EN); twl4030_apll_enable(w->codec, 0); break; } @@ -736,9 +735,8 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_pop |= TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); /* Actually write to the register */ - twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - hs_gain, - TWL4030_REG_HS_GAIN_SET); + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, hs_gain, + TWL4030_REG_HS_GAIN_SET); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); /* Wait ramp delay time + 1, so the VMID can settle */ @@ -751,9 +749,8 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) /* Wait ramp delay time + 1, so the VMID can settle */ twl4030_wait_ms(delay); /* Bypass the reg_cache to mute the headset */ - twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - hs_gain & (~0x0f), - TWL4030_REG_HS_GAIN_SET); + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, hs_gain & (~0x0f), + TWL4030_REG_HS_GAIN_SET); hs_pop &= ~TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); @@ -771,7 +768,7 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) } static int headsetlpga_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) + struct snd_kcontrol *kcontrol, int event) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); @@ -795,7 +792,7 @@ static int headsetlpga_event(struct snd_soc_dapm_widget *w, } static int headsetrpga_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) + struct snd_kcontrol *kcontrol, int event) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); @@ -819,7 +816,7 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w, } static int digimic_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) + struct snd_kcontrol *kcontrol, int event) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); struct twl4030_codec_data *pdata = twl4030->pdata; @@ -840,7 +837,7 @@ static int digimic_event(struct snd_soc_dapm_widget *w, * Custom volsw and volsw_2r get/put functions to handle these gain bits. */ static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; @@ -869,7 +866,7 @@ static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol, } static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; @@ -898,7 +895,7 @@ static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol, } static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; @@ -925,7 +922,7 @@ static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, } static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; @@ -1656,7 +1653,7 @@ static void twl4030_constraints(struct twl4030_priv *twl4030, /* In case of 4 channel mode, the RX1 L/R for playback and the TX2 L/R for * capture has to be enabled/disabled. */ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, - int enable) + int enable) { u8 reg, mask; @@ -1695,8 +1692,8 @@ static int twl4030_startup(struct snd_pcm_substream *substream, * constraint for the first stream for channels, the * second stream will 'inherit' this cosntraint */ snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, - 2, 2); + SNDRV_PCM_HW_PARAM_CHANNELS, + 2, 2); } twl4030->master_substream = substream; } @@ -1728,8 +1725,8 @@ static void twl4030_shutdown(struct snd_pcm_substream *substream, } static int twl4030_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); @@ -1845,8 +1842,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, return 0; } -static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) +static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, + unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); @@ -1871,8 +1868,7 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) +static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); @@ -1942,7 +1938,7 @@ static int twl4030_set_tristate(struct snd_soc_dai *dai, int tristate) /* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R * (VTXL, VTXR) for uplink has to be enabled/disabled. */ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, - int enable) + int enable) { u8 reg, mask; @@ -1962,7 +1958,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, } static int twl4030_voice_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); @@ -1994,7 +1990,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, } static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; @@ -2003,7 +1999,8 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, } static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); @@ -2013,8 +2010,8 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, twl4030_voice_enable(codec, substream->stream, 1); /* bit rate */ - old_mode = twl4030_read(codec, TWL4030_REG_CODEC_MODE) - & ~(TWL4030_CODECPDZ); + old_mode = twl4030_read(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; mode = old_mode; switch (params_rate(params)) { @@ -2048,7 +2045,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, } static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) + int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); @@ -2069,7 +2066,7 @@ static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, } static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) + unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); @@ -2242,7 +2239,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { static int twl4030_codec_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_twl4030, - twl4030_dai, ARRAY_SIZE(twl4030_dai)); + twl4030_dai, ARRAY_SIZE(twl4030_dai)); } static int twl4030_codec_remove(struct platform_device *pdev) -- cgit v1.2.3-70-g09d2 From a450aa6f507542d87ad237cb402b8b6e56329924 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 Jan 2014 15:27:55 +0200 Subject: ASoC: twl4030: Move the ctl cache update local to twl4030_write() function There's no other users of this functionality, the code can be moved inside of twl4030_write. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 36 ++++++++++++++++-------------------- 1 file changed, 16 insertions(+), 20 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 7a5b91e70f9..c3c15f89127 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -86,25 +86,6 @@ static void tw4030_init_ctl_cache(struct twl4030_priv *twl4030) } } -static void twl4030_update_ctl_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - - switch (reg) { - case TWL4030_REG_EAR_CTL: - case TWL4030_REG_PREDL_CTL: - case TWL4030_REG_PREDR_CTL: - case TWL4030_REG_PRECKL_CTL: - case TWL4030_REG_PRECKR_CTL: - case TWL4030_REG_HS_GAIN_SET: - twl4030->ctl_cache[reg - TWL4030_REG_EAR_CTL] = value; - break; - default: - break; - } -} - static unsigned int twl4030_read(struct snd_soc_codec *codec, unsigned int reg) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); @@ -174,7 +155,22 @@ static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec, static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - twl4030_update_ctl_cache(codec, reg, value); + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); + + /* Update the ctl cache */ + switch (reg) { + case TWL4030_REG_EAR_CTL: + case TWL4030_REG_PREDL_CTL: + case TWL4030_REG_PREDR_CTL: + case TWL4030_REG_PRECKL_CTL: + case TWL4030_REG_PRECKR_CTL: + case TWL4030_REG_HS_GAIN_SET: + twl4030->ctl_cache[reg - TWL4030_REG_EAR_CTL] = value; + break; + default: + break; + } + if (twl4030_can_write_to_chip(codec, reg)) return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); -- cgit v1.2.3-70-g09d2 From b703b504856b9a9df6bace81e251d185dd72e958 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 Jan 2014 15:27:56 +0200 Subject: ASoC: twl4030: Pass the twl4030_priv directly to twl4030_can_write_to_chip() To avoid another lookup for the twl4030_priv in there. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c3c15f89127..00665ada23e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -111,10 +111,9 @@ static unsigned int twl4030_read(struct snd_soc_codec *codec, unsigned int reg) return value; } -static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec, +static bool twl4030_can_write_to_chip(struct twl4030_priv *twl4030, unsigned int reg) { - struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); bool write_to_reg = false; /* Decide if the given register can be written */ @@ -171,7 +170,7 @@ static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg, break; } - if (twl4030_can_write_to_chip(codec, reg)) + if (twl4030_can_write_to_chip(twl4030, reg)) return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); return 0; -- cgit v1.2.3-70-g09d2 From 7ae2b55b0129ecb63d73129ddcba6dcda0d37332 Mon Sep 17 00:00:00 2001 From: Andreas Pretzsch Date: Tue, 7 Jan 2014 22:34:41 +0100 Subject: ASoC: ssm2602: add 16kHz sampling rate support SSM260x also supports 16kHz with external master clocks of 12.000MHz, 12.288MHz and 18.432MHz. Add matching coefficients, update constraints and announced rates. Signed-off-by: Andreas Pretzsch Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c6dd4856188..af76bbd1b24 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -194,7 +194,7 @@ static const struct snd_soc_dapm_route ssm2604_routes[] = { }; static const unsigned int ssm2602_rates_12288000[] = { - 8000, 32000, 48000, 96000, + 8000, 16000, 32000, 48000, 96000, }; static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { @@ -231,6 +231,11 @@ static const struct ssm2602_coeff ssm2602_coeff_table[] = { {18432000, 32000, SSM2602_COEFF_SRATE(0x6, 0x1, 0x0)}, {12000000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x1)}, + /* 16k */ + {12288000, 16000, SSM2602_COEFF_SRATE(0x5, 0x0, 0x0)}, + {18432000, 16000, SSM2602_COEFF_SRATE(0x5, 0x1, 0x0)}, + {12000000, 16000, SSM2602_COEFF_SRATE(0xa, 0x0, 0x1)}, + /* 8k */ {12288000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x0)}, {18432000, 8000, SSM2602_COEFF_SRATE(0x3, 0x1, 0x0)}, @@ -473,9 +478,10 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -- cgit v1.2.3-70-g09d2 From 2841be9afa6c9d37d41386af30cd8813acd06739 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 20 Dec 2013 14:11:28 +0100 Subject: ASoC: fsl-ssi: Fix probe error handling This patch fixes the error handling in the fsl-ssi probe function. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 816ae4b28a5..19891f2a5de 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -141,6 +141,7 @@ struct fsl_ssi_private { bool imx_ac97; bool use_dma; bool baudclk_locked; + bool irq_stats; u8 i2s_mode; spinlock_t baudclk_lock; struct clk *baudclk; @@ -1224,6 +1225,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = devm_request_irq(&pdev->dev, ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, ssi_private); + ssi_private->irq_stats = true; if (ret < 0) { dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); @@ -1274,11 +1276,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = imx_pcm_fiq_init(pdev, &ssi_private->fiq_params); if (ret) - goto error_dev; + goto error_pcm; } else { ret = imx_pcm_dma_init(pdev); if (ret) - goto error_dev; + goto error_pcm; } } @@ -1320,6 +1322,10 @@ done: return 0; error_dai: + if (ssi_private->ssi_on_imx && !ssi_private->use_dma) + imx_pcm_fiq_exit(pdev); + +error_pcm: snd_soc_unregister_component(&pdev->dev); error_dev: @@ -1333,7 +1339,8 @@ error_clk: } error_irqmap: - irq_dispose_mapping(ssi_private->irq); + if (ssi_private->irq_stats) + irq_dispose_mapping(ssi_private->irq); return ret; } @@ -1351,7 +1358,8 @@ static int fsl_ssi_remove(struct platform_device *pdev) clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); } - irq_dispose_mapping(ssi_private->irq); + if (ssi_private->irq_stats) + irq_dispose_mapping(ssi_private->irq); return 0; } -- cgit v1.2.3-70-g09d2 From 9368acc4383bd8cca671fdc49c5f7e241b6909b3 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 20 Dec 2013 14:11:29 +0100 Subject: ASoC: fsl-ssi: Move sysfs stats to debugfs Interrupt statistics of fsl_ssi are mainly for debugging purpose. Most of those interrupts show error states, e.g. under/overflow. The stats should be exposed via debugfs instead of sysfs. This patch moves the statistics file to debugfs. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 184 ++++++++++++++++++++++++++++++------------------ 1 file changed, 117 insertions(+), 67 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 19891f2a5de..e483e9d84f8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -35,6 +35,7 @@ #include #include #include +#include #include #include #include @@ -114,6 +115,14 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) +#define FSLSSI_SIER_DBG_RX_FLAGS (CCSR_SSI_SIER_RFF0_EN | \ + CCSR_SSI_SIER_RLS_EN | CCSR_SSI_SIER_RFS_EN | \ + CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_RFRC_EN) +#define FSLSSI_SIER_DBG_TX_FLAGS (CCSR_SSI_SIER_TFE0_EN | \ + CCSR_SSI_SIER_TLS_EN | CCSR_SSI_SIER_TFS_EN | \ + CCSR_SSI_SIER_TUE0_EN | CCSR_SSI_SIER_TFRC_EN) +#define FSLSSI_SISR_MASK (FSLSSI_SIER_DBG_RX_FLAGS | FSLSSI_SIER_DBG_TX_FLAGS) + /** * fsl_ssi_private: per-SSI private data * @@ -133,7 +142,6 @@ struct fsl_ssi_private { unsigned int irq; unsigned int fifo_depth; struct snd_soc_dai_driver cpu_dai_drv; - struct device_attribute dev_attr; struct platform_device *pdev; bool new_binding; @@ -175,6 +183,8 @@ struct fsl_ssi_private { unsigned int tfe1; unsigned int tfe0; } stats; + struct dentry *dbg_dir; + struct dentry *dbg_stats; char name[1]; }; @@ -203,7 +213,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) were interrupted for. We mask it with the Interrupt Enable register so that we only check for events that we're interested in. */ - sisr = read_ssi(&ssi->sisr) & SIER_FLAGS; + sisr = read_ssi(&ssi->sisr) & FSLSSI_SISR_MASK; if (sisr & CCSR_SSI_SISR_RFRC) { ssi_private->stats.rfrc++; @@ -323,6 +333,102 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) return ret; } +#if IS_ENABLED(CONFIG_DEBUG_FS) +/* Show the statistics of a flag only if its interrupt is enabled. The + * compiler will optimze this code to a no-op if the interrupt is not + * enabled. + */ +#define SIER_SHOW(flag, name) \ + do { \ + if (FSLSSI_SISR_MASK & CCSR_SSI_SIER_##flag) \ + seq_printf(s, #name "=%u\n", ssi_private->stats.name); \ + } while (0) + + +/** + * fsl_sysfs_ssi_show: display SSI statistics + * + * Display the statistics for the current SSI device. To avoid confusion, + * we only show those counts that are enabled. + */ +static ssize_t fsl_ssi_stats_show(struct seq_file *s, void *unused) +{ + struct fsl_ssi_private *ssi_private = s->private; + + SIER_SHOW(RFRC_EN, rfrc); + SIER_SHOW(TFRC_EN, tfrc); + SIER_SHOW(CMDAU_EN, cmdau); + SIER_SHOW(CMDDU_EN, cmddu); + SIER_SHOW(RXT_EN, rxt); + SIER_SHOW(RDR1_EN, rdr1); + SIER_SHOW(RDR0_EN, rdr0); + SIER_SHOW(TDE1_EN, tde1); + SIER_SHOW(TDE0_EN, tde0); + SIER_SHOW(ROE1_EN, roe1); + SIER_SHOW(ROE0_EN, roe0); + SIER_SHOW(TUE1_EN, tue1); + SIER_SHOW(TUE0_EN, tue0); + SIER_SHOW(TFS_EN, tfs); + SIER_SHOW(RFS_EN, rfs); + SIER_SHOW(TLS_EN, tls); + SIER_SHOW(RLS_EN, rls); + SIER_SHOW(RFF1_EN, rff1); + SIER_SHOW(RFF0_EN, rff0); + SIER_SHOW(TFE1_EN, tfe1); + SIER_SHOW(TFE0_EN, tfe0); + + return 0; +} + +static int fsl_ssi_stats_open(struct inode *inode, struct file *file) +{ + return single_open(file, fsl_ssi_stats_show, inode->i_private); +} + +static const struct file_operations fsl_ssi_stats_ops = { + .open = fsl_ssi_stats_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static int fsl_ssi_debugfs_create(struct fsl_ssi_private *ssi_private, + struct device *dev) +{ + ssi_private->dbg_dir = debugfs_create_dir(dev_name(dev), NULL); + if (!ssi_private->dbg_dir) + return -ENOMEM; + + ssi_private->dbg_stats = debugfs_create_file("stats", S_IRUGO, + ssi_private->dbg_dir, ssi_private, &fsl_ssi_stats_ops); + if (!ssi_private->dbg_stats) { + debugfs_remove(ssi_private->dbg_dir); + return -ENOMEM; + } + + return 0; +} + +static void fsl_ssi_debugfs_remove(struct fsl_ssi_private *ssi_private) +{ + debugfs_remove(ssi_private->dbg_stats); + debugfs_remove(ssi_private->dbg_dir); +} + +#else + +static int fsl_ssi_debugfs_create(struct fsl_ssi_private *ssi_private, + struct device *dev) +{ + return 0; +} + +static void fsl_ssi_debugfs_remove(struct fsl_ssi_private *ssi_private) +{ +} + +#endif /* IS_ENABLED(CONFIG_DEBUG_FS) */ + static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; @@ -991,56 +1097,6 @@ static struct snd_ac97_bus_ops fsl_ssi_ac97_ops = { .write = fsl_ssi_ac97_write, }; -/* Show the statistics of a flag only if its interrupt is enabled. The - * compiler will optimze this code to a no-op if the interrupt is not - * enabled. - */ -#define SIER_SHOW(flag, name) \ - do { \ - if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \ - length += sprintf(buf + length, #name "=%u\n", \ - ssi_private->stats.name); \ - } while (0) - - -/** - * fsl_sysfs_ssi_show: display SSI statistics - * - * Display the statistics for the current SSI device. To avoid confusion, - * we only show those counts that are enabled. - */ -static ssize_t fsl_sysfs_ssi_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct fsl_ssi_private *ssi_private = - container_of(attr, struct fsl_ssi_private, dev_attr); - ssize_t length = 0; - - SIER_SHOW(RFRC_EN, rfrc); - SIER_SHOW(TFRC_EN, tfrc); - SIER_SHOW(CMDAU_EN, cmdau); - SIER_SHOW(CMDDU_EN, cmddu); - SIER_SHOW(RXT_EN, rxt); - SIER_SHOW(RDR1_EN, rdr1); - SIER_SHOW(RDR0_EN, rdr0); - SIER_SHOW(TDE1_EN, tde1); - SIER_SHOW(TDE0_EN, tde0); - SIER_SHOW(ROE1_EN, roe1); - SIER_SHOW(ROE0_EN, roe0); - SIER_SHOW(TUE1_EN, tue1); - SIER_SHOW(TUE0_EN, tue0); - SIER_SHOW(TFS_EN, tfs); - SIER_SHOW(RFS_EN, rfs); - SIER_SHOW(TLS_EN, tls); - SIER_SHOW(RLS_EN, rls); - SIER_SHOW(RFF1_EN, rff1); - SIER_SHOW(RFF0_EN, rff0); - SIER_SHOW(TFE1_EN, tfe1); - SIER_SHOW(TFE0_EN, tfe0); - - return length; -} - /** * Make every character in a string lower-case */ @@ -1233,20 +1289,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) } } - /* Initialize the the device_attribute structure */ - dev_attr = &ssi_private->dev_attr; - sysfs_attr_init(&dev_attr->attr); - dev_attr->attr.name = "statistics"; - dev_attr->attr.mode = S_IRUGO; - dev_attr->show = fsl_sysfs_ssi_show; - - ret = device_create_file(&pdev->dev, dev_attr); - if (ret) { - dev_err(&pdev->dev, "could not create sysfs %s file\n", - ssi_private->dev_attr.attr.name); - goto error_clk; - } - /* Register with ASoC */ dev_set_drvdata(&pdev->dev, ssi_private); @@ -1257,6 +1299,10 @@ static int fsl_ssi_probe(struct platform_device *pdev) goto error_dev; } + ret = fsl_ssi_debugfs_create(ssi_private, &pdev->dev); + if (ret) + goto error_dbgfs; + if (ssi_private->ssi_on_imx) { if (!ssi_private->use_dma) { @@ -1326,6 +1372,9 @@ error_dai: imx_pcm_fiq_exit(pdev); error_pcm: + fsl_ssi_debugfs_remove(ssi_private); + +error_dbgfs: snd_soc_unregister_component(&pdev->dev); error_dev: @@ -1349,10 +1398,11 @@ static int fsl_ssi_remove(struct platform_device *pdev) { struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev); + fsl_ssi_debugfs_remove(ssi_private); + if (!ssi_private->new_binding) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); - device_remove_file(&pdev->dev, &ssi_private->dev_attr); if (ssi_private->ssi_on_imx) { if (!IS_ERR(ssi_private->baudclk)) clk_disable_unprepare(ssi_private->baudclk); -- cgit v1.2.3-70-g09d2 From c1953bfe1329eeb16991d430d574c4280697ad17 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 20 Dec 2013 14:11:30 +0100 Subject: ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching There is a small but significant difference between imx21-ssi and imx51-ssi and above. imx21-ssi does not allow online reconfiguration of some registers. They have to be configured at the beginning. imx51-ssi has to reconfigure the SSI unit while it is running. Otherwise it would not be possible to have two streams in parallel. The new SDMA unit in imx51 and above has to be configured before the first DMA request arrives. Therefor we need to setup TDMAE/RDMAE just before starting the stream. This patch introduces distinct imx51-ssi as compatible and adds of_device_id matching in the probe function. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e483e9d84f8..671be33aa9d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -123,6 +123,13 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) CCSR_SSI_SIER_TUE0_EN | CCSR_SSI_SIER_TFRC_EN) #define FSLSSI_SISR_MASK (FSLSSI_SIER_DBG_RX_FLAGS | FSLSSI_SIER_DBG_TX_FLAGS) + +enum fsl_ssi_type { + FSL_SSI_MCP8610, + FSL_SSI_MX21, + FSL_SSI_MX51, +}; + /** * fsl_ssi_private: per-SSI private data * @@ -189,6 +196,14 @@ struct fsl_ssi_private { char name[1]; }; +static const struct of_device_id fsl_ssi_ids[] = { + { .compatible = "fsl,mpc8610-ssi", .data = (void *) FSL_SSI_MCP8610}, + { .compatible = "fsl,imx51-ssi", .data = (void *) FSL_SSI_MX51}, + { .compatible = "fsl,imx21-ssi", .data = (void *) FSL_SSI_MX21}, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_ssi_ids); + /** * fsl_ssi_isr: SSI interrupt handler * @@ -1118,6 +1133,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) int ret = 0; struct device_attribute *dev_attr = NULL; struct device_node *np = pdev->dev.of_node; + const struct of_device_id *of_id; + enum fsl_ssi_type hw_type; const char *p, *sprop; const uint32_t *iprop; struct resource res; @@ -1132,6 +1149,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (!of_device_is_available(np)) return -ENODEV; + of_id = of_match_device(fsl_ssi_ids, &pdev->dev); + if (!of_id) + return -EINVAL; + hw_type = (enum fsl_ssi_type) of_id->data; + /* We only support the SSI in "I2S Slave" mode */ sprop = of_get_property(np, "fsl,mode", NULL); if (!sprop) { @@ -1211,7 +1233,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private->baudclk_locked = false; spin_lock_init(&ssi_private->baudclk_lock); - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { + if (hw_type == FSL_SSI_MX21 || hw_type == FSL_SSI_MX51 || + hw_type == FSL_SSI_MX35) { u32 dma_events[2]; ssi_private->ssi_on_imx = true; @@ -1414,13 +1437,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) return 0; } -static const struct of_device_id fsl_ssi_ids[] = { - { .compatible = "fsl,mpc8610-ssi", }, - { .compatible = "fsl,imx21-ssi", }, - {} -}; -MODULE_DEVICE_TABLE(of, fsl_ssi_ids); - static struct platform_driver fsl_ssi_driver = { .driver = { .name = "fsl-ssi-dai", -- cgit v1.2.3-70-g09d2 From 0888efd166fa99b733b0b68e70d2fb3c3c7684ec Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 20 Dec 2013 14:11:31 +0100 Subject: ASoC: fsl-ssi: Fix interrupt stats for imx irqs should only be requested/released with enabled DMA. Previously interrupt statistics where disabled for IMX Processors because of different writeable SISR bits. This patch introduces support for irqstats on imx processors again by creating a sisr writeback mask and introducing a imx35-ssi compatible. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 40 ++++++++++++++++++++++++++++++++-------- 1 file changed, 32 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 671be33aa9d..bc904696d82 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -127,6 +127,7 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) enum fsl_ssi_type { FSL_SSI_MCP8610, FSL_SSI_MX21, + FSL_SSI_MX35, FSL_SSI_MX51, }; @@ -151,6 +152,7 @@ struct fsl_ssi_private { struct snd_soc_dai_driver cpu_dai_drv; struct platform_device *pdev; + enum fsl_ssi_type hw_type; bool new_binding; bool ssi_on_imx; bool imx_ac97; @@ -199,6 +201,7 @@ struct fsl_ssi_private { static const struct of_device_id fsl_ssi_ids[] = { { .compatible = "fsl,mpc8610-ssi", .data = (void *) FSL_SSI_MCP8610}, { .compatible = "fsl,imx51-ssi", .data = (void *) FSL_SSI_MX51}, + { .compatible = "fsl,imx35-ssi", .data = (void *) FSL_SSI_MX35}, { .compatible = "fsl,imx21-ssi", .data = (void *) FSL_SSI_MX21}, {} }; @@ -222,7 +225,26 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) struct ccsr_ssi __iomem *ssi = ssi_private->ssi; irqreturn_t ret = IRQ_NONE; __be32 sisr; - __be32 sisr2 = 0; + __be32 sisr2; + __be32 sisr_write_mask = 0; + + switch (ssi_private->hw_type) { + case FSL_SSI_MX21: + sisr_write_mask = 0; + break; + + case FSL_SSI_MCP8610: + case FSL_SSI_MX35: + sisr_write_mask = CCSR_SSI_SISR_RFRC | CCSR_SSI_SISR_TFRC | + CCSR_SSI_SISR_ROE0 | CCSR_SSI_SISR_ROE1 | + CCSR_SSI_SISR_TUE0 | CCSR_SSI_SISR_TUE1; + break; + + case FSL_SSI_MX51: + sisr_write_mask = CCSR_SSI_SISR_ROE0 | CCSR_SSI_SISR_ROE1 | + CCSR_SSI_SISR_TUE0 | CCSR_SSI_SISR_TUE1; + break; + } /* We got an interrupt, so read the status register to see what we were interrupted for. We mask it with the Interrupt Enable register @@ -232,13 +254,11 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) if (sisr & CCSR_SSI_SISR_RFRC) { ssi_private->stats.rfrc++; - sisr2 |= CCSR_SSI_SISR_RFRC; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TFRC) { ssi_private->stats.tfrc++; - sisr2 |= CCSR_SSI_SISR_TFRC; ret = IRQ_HANDLED; } @@ -279,25 +299,21 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) if (sisr & CCSR_SSI_SISR_ROE1) { ssi_private->stats.roe1++; - sisr2 |= CCSR_SSI_SISR_ROE1; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_ROE0) { ssi_private->stats.roe0++; - sisr2 |= CCSR_SSI_SISR_ROE0; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TUE1) { ssi_private->stats.tue1++; - sisr2 |= CCSR_SSI_SISR_TUE1; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TUE0) { ssi_private->stats.tue0++; - sisr2 |= CCSR_SSI_SISR_TUE0; ret = IRQ_HANDLED; } @@ -341,6 +357,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) ret = IRQ_HANDLED; } + sisr2 = sisr & sisr_write_mask; /* Clear the bits that we set */ if (sisr2) write_ssi(sisr2, &ssi->sisr); @@ -1180,6 +1197,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private->use_dma = !of_property_read_bool(np, "fsl,fiq-stream-filter"); + ssi_private->hw_type = hw_type; if (ac97) { memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_ac97_dai, @@ -1299,7 +1317,13 @@ static int fsl_ssi_probe(struct platform_device *pdev) dma_events[0], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); imx_pcm_dma_params_init_data(&ssi_private->filter_data_rx, dma_events[1], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); - } else if (ssi_private->use_dma) { + } + + /* + * Enable interrupts only for MCP8610 and MX51. The other MXs have + * different writeable interrupt status registers. + */ + if (ssi_private->use_dma) { /* The 'name' should not have any slashes in it. */ ret = devm_request_irq(&pdev->dev, ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, -- cgit v1.2.3-70-g09d2 From bd3ca7d1b8ee0dcd502c8c15d1cf741bc165722f Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 20 Dec 2013 14:11:32 +0100 Subject: ASoC: fsl-ssi: Add offline_config flag imx50-ssi and later versions of this IP support online reconfiguration of all registers. The reference manual does not list any registers that can only be configured while the SSI unit is disabled. This patch introduces the flag for later use. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bc904696d82..d0b9fe31f49 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -159,6 +159,7 @@ struct fsl_ssi_private { bool use_dma; bool baudclk_locked; bool irq_stats; + bool offline_config; u8 i2s_mode; spinlock_t baudclk_lock; struct clk *baudclk; @@ -1251,6 +1252,32 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private->baudclk_locked = false; spin_lock_init(&ssi_private->baudclk_lock); + /* + * imx51 and later SoCs have a slightly different IP that allows the + * SSI configuration while the SSI unit is running. + * + * More important, it is necessary on those SoCs to configure the + * sperate TX/RX DMA bits just before starting the stream + * (fsl_ssi_trigger). The SDMA unit has to be configured before fsl_ssi + * sends any DMA requests to the SDMA unit, otherwise it is not defined + * how the SDMA unit handles the DMA request. + * + * SDMA units are present on devices starting at imx35 but the imx35 + * reference manual states that the DMA bits should not be changed + * while the SSI unit is running (SSIEN). So we support the necessary + * online configuration of fsl-ssi starting at imx51. + */ + switch (hw_type) { + case FSL_SSI_MCP8610: + case FSL_SSI_MX21: + case FSL_SSI_MX35: + ssi_private->offline_config = true; + break; + case FSL_SSI_MX51: + ssi_private->offline_config = false; + break; + } + if (hw_type == FSL_SSI_MX21 || hw_type == FSL_SSI_MX51 || hw_type == FSL_SSI_MX35) { u32 dma_events[2]; -- cgit v1.2.3-70-g09d2 From 4e6ec0d98c045cb2c0c6550c65c4afae208872e9 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 20 Dec 2013 14:11:33 +0100 Subject: ASoC: fsl-ssi: Add configuration helper functions This patch adds a struct 'fsl_ssi_rxtx_reg_val' which holds register values necessary to enable rx/tx. Based on those preset register values, the added configuration functions will cleanly enable/disable different parts of the SSI IP while supporting online/offline configuration. Different operating modes can be setup directly as different register values in fsl_ssi_reg_val. These functions and structs will help to cleanup and simplify the trigger function to support many different IP versions (online/offline configuration) and different operating modes. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 122 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 122 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d0b9fe31f49..a85268bb450 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -131,6 +131,18 @@ enum fsl_ssi_type { FSL_SSI_MX51, }; +struct fsl_ssi_reg_val { + u32 sier; + u32 srcr; + u32 stcr; + u32 scr; +}; + +struct fsl_ssi_rxtx_reg_val { + struct fsl_ssi_reg_val rx; + struct fsl_ssi_reg_val tx; +}; + /** * fsl_ssi_private: per-SSI private data * @@ -169,6 +181,8 @@ struct fsl_ssi_private { struct imx_dma_data filter_data_tx; struct imx_dma_data filter_data_rx; struct imx_pcm_fiq_params fiq_params; + /* Register values for rx/tx configuration */ + struct fsl_ssi_rxtx_reg_val rxtx_reg_val; struct { unsigned int rfrc; @@ -462,6 +476,114 @@ static void fsl_ssi_debugfs_remove(struct fsl_ssi_private *ssi_private) #endif /* IS_ENABLED(CONFIG_DEBUG_FS) */ +/* + * Enable/Disable all rx/tx config flags at once. + */ +static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private, + bool enable) +{ + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + struct fsl_ssi_rxtx_reg_val *vals = &ssi_private->rxtx_reg_val; + + if (enable) { + write_ssi_mask(&ssi->sier, 0, vals->rx.sier | vals->tx.sier); + write_ssi_mask(&ssi->srcr, 0, vals->rx.srcr | vals->tx.srcr); + write_ssi_mask(&ssi->stcr, 0, vals->rx.stcr | vals->tx.stcr); + } else { + write_ssi_mask(&ssi->srcr, vals->rx.srcr | vals->tx.srcr, 0); + write_ssi_mask(&ssi->stcr, vals->rx.stcr | vals->tx.stcr, 0); + write_ssi_mask(&ssi->sier, vals->rx.sier | vals->tx.sier, 0); + } +} + +/* + * Enable/Disable a ssi configuration. You have to pass either + * ssi_private->rxtx_reg_val.rx or tx as vals parameter. + */ +static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, + struct fsl_ssi_reg_val *vals) +{ + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + struct fsl_ssi_reg_val *avals; + u32 scr_val = read_ssi(&ssi->scr); + int nr_active_streams = !!(scr_val & CCSR_SSI_SCR_TE) + + !!(scr_val & CCSR_SSI_SCR_RE); + + /* Find the other direction values rx or tx which we do not want to + * modify */ + if (&ssi_private->rxtx_reg_val.rx == vals) + avals = &ssi_private->rxtx_reg_val.tx; + else + avals = &ssi_private->rxtx_reg_val.rx; + + /* If vals should be disabled, start with disabling the unit */ + if (!enable) { + u32 scr = vals->scr & (vals->scr ^ avals->scr); + write_ssi_mask(&ssi->scr, scr, 0); + } + + /* + * We are running on a SoC which does not support online SSI + * reconfiguration, so we have to enable all necessary flags at once + * even if we do not use them later (capture and playback configuration) + */ + if (ssi_private->offline_config) { + if ((enable && !nr_active_streams) || + (!enable && nr_active_streams == 1)) + fsl_ssi_rxtx_config(ssi_private, enable); + + goto config_done; + } + + /* + * Configure single direction units while the SSI unit is running + * (online configuration) + */ + if (enable) { + write_ssi_mask(&ssi->sier, 0, vals->sier); + write_ssi_mask(&ssi->srcr, 0, vals->srcr); + write_ssi_mask(&ssi->stcr, 0, vals->stcr); + } else { + u32 sier; + u32 srcr; + u32 stcr; + + /* + * Disabling the necessary flags for one of rx/tx while the + * other stream is active is a little bit more difficult. We + * have to disable only those flags that differ between both + * streams (rx XOR tx) and that are set in the stream that is + * disabled now. Otherwise we could alter flags of the other + * stream + */ + + /* These assignments are simply vals without bits set in avals*/ + sier = vals->sier & (vals->sier ^ avals->sier); + srcr = vals->srcr & (vals->srcr ^ avals->srcr); + stcr = vals->stcr & (vals->stcr ^ avals->stcr); + + write_ssi_mask(&ssi->srcr, srcr, 0); + write_ssi_mask(&ssi->stcr, stcr, 0); + write_ssi_mask(&ssi->sier, sier, 0); + } + +config_done: + /* Enabling of subunits is done after configuration */ + if (enable) + write_ssi_mask(&ssi->scr, 0, vals->scr); +} + + +static void fsl_ssi_rx_config(struct fsl_ssi_private *ssi_private, bool enable) +{ + fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.rx); +} + +static void fsl_ssi_tx_config(struct fsl_ssi_private *ssi_private, bool enable) +{ + fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.tx); +} + static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; -- cgit v1.2.3-70-g09d2 From 6de8387905a69568489284b4660737eebb0db8cf Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 20 Dec 2013 14:11:34 +0100 Subject: ASoC: fsl-ssi: Move RX/TX configuration to seperate functions This patch defines the appropriate register values for different oparation modes and IP versions. We have to handle DMA/FIQ, AC97, DEBUG-IRQs and offline/online configuration support. With this patch we cleanup some driver code that was not reference manual conform and try to cleanup the whole trigger function to seperate the actual register values from the enable/disable logic, which is now hidden in fsl_ssi_config helpers. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 89 +++++++++++++++++++++++++------------------------ 1 file changed, 46 insertions(+), 43 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a85268bb450..a96ab4e6065 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -108,13 +108,6 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE) #endif -/* SIER bitflag of interrupts to enable */ -#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ - CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ - CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ - CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) - #define FSLSSI_SIER_DBG_RX_FLAGS (CCSR_SSI_SIER_RFF0_EN | \ CCSR_SSI_SIER_RLS_EN | CCSR_SSI_SIER_RFS_EN | \ CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_RFRC_EN) @@ -584,6 +577,41 @@ static void fsl_ssi_tx_config(struct fsl_ssi_private *ssi_private, bool enable) fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.tx); } +/* + * Setup rx/tx register values used to enable/disable the streams. These will + * be used later in fsl_ssi_config to setup the streams without the need to + * check for all different SSI modes. + */ +static void fsl_ssi_setup_reg_vals(struct fsl_ssi_private *ssi_private) +{ + struct fsl_ssi_rxtx_reg_val *reg = &ssi_private->rxtx_reg_val; + + reg->rx.sier = CCSR_SSI_SIER_RFF0_EN; + reg->rx.srcr = CCSR_SSI_SRCR_RFEN0; + reg->rx.scr = 0; + reg->tx.sier = CCSR_SSI_SIER_TFE0_EN; + reg->tx.stcr = CCSR_SSI_STCR_TFEN0; + reg->tx.scr = 0; + + if (!ssi_private->imx_ac97) { + reg->rx.scr = CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE; + reg->rx.sier |= CCSR_SSI_SIER_RFF0_EN; + reg->tx.scr = CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE; + reg->tx.sier |= CCSR_SSI_SIER_TFE0_EN; + } + + if (ssi_private->use_dma) { + reg->rx.sier |= CCSR_SSI_SIER_RDMAE; + reg->tx.sier |= CCSR_SSI_SIER_TDMAE; + } else { + reg->rx.sier |= CCSR_SSI_SIER_RIE; + reg->tx.sier |= CCSR_SSI_SIER_TIE; + } + + reg->rx.sier |= FSLSSI_SIER_DBG_RX_FLAGS; + reg->tx.sier |= FSLSSI_SIER_DBG_TX_FLAGS; +} + static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; @@ -620,6 +648,8 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) u8 wm; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; + fsl_ssi_setup_reg_vals(ssi_private); + if (ssi_private->imx_ac97) ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; else @@ -643,13 +673,12 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) ssi_private->i2s_mode | (synchronous ? CCSR_SSI_SCR_SYN : 0)); - write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | - CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | - CCSR_SSI_STCR_TSCKP, &ssi->stcr); + write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFSI | + CCSR_SSI_STCR_TEFS | CCSR_SSI_STCR_TSCKP, &ssi->stcr); + + write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFSI | + CCSR_SSI_SRCR_REFS | CCSR_SSI_SRCR_RSCKP, &ssi->srcr); - write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | - CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | - CCSR_SSI_SRCR_RSCKP, &ssi->srcr); /* * The DC and PM bits are only used if the SSI is the clock master. */ @@ -1023,51 +1052,26 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - unsigned int sier_bits; unsigned long flags; - /* - * Enable only the interrupts and DMA requests - * that are needed for the channel. As the fiq - * is polling for this bits, we have to ensure - * that this are aligned with the preallocated - * buffers - */ - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (ssi_private->use_dma) - sier_bits = SIER_FLAGS; - else - sier_bits = CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TFE0_EN; - } else { - if (ssi_private->use_dma) - sier_bits = SIER_FLAGS; - else - sier_bits = CCSR_SSI_SIER_RIE | CCSR_SSI_SIER_RFF0_EN; - } - switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - write_ssi_mask(&ssi->scr, 0, - CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); + fsl_ssi_tx_config(ssi_private, true); else - write_ssi_mask(&ssi->scr, 0, - CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); + fsl_ssi_rx_config(ssi_private, true); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); + fsl_ssi_tx_config(ssi_private, false); else - write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); + fsl_ssi_rx_config(ssi_private, false); if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) & (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) { - write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); spin_lock_irqsave(&ssi_private->baudclk_lock, flags); ssi_private->baudclk_locked = false; spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); @@ -1078,7 +1082,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, return -EINVAL; } - write_ssi(sier_bits, &ssi->sier); return 0; } -- cgit v1.2.3-70-g09d2 From a5a7ee7c98bc2a7d0324de661778783ab2c29343 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Fri, 20 Dec 2013 14:11:35 +0100 Subject: ASoC: fsl-ssi: Drop ac97 specific trigger function The normal trigger function can now be used for AC97. Drop AC97 trigger function. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 61 +++++++------------------------------------------ 1 file changed, 8 insertions(+), 53 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a96ab4e6065..94dedcb0868 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1052,6 +1052,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; unsigned long flags; switch (cmd) { @@ -1082,6 +1083,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, return -EINVAL; } + if (ssi_private->imx_ac97) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + write_ssi(CCSR_SSI_SOR_TX_CLR, &ssi->sor); + else + write_ssi(CCSR_SSI_SOR_RX_CLR, &ssi->sor); + } return 0; } @@ -1129,58 +1136,6 @@ static const struct snd_soc_component_driver fsl_ssi_component = { .name = "fsl-ssi", }; -/** - * fsl_ssi_ac97_trigger: start and stop the AC97 receive/transmit. - * - * This function is called by ALSA to start, stop, pause, and resume the - * transfer of data. - */ -static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata( - rtd->cpu_dai); - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_TIE | - CCSR_SSI_SIER_TFE0_EN); - else - write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_RIE | - CCSR_SSI_SIER_RFF0_EN); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_TIE | - CCSR_SSI_SIER_TFE0_EN, 0); - else - write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_RIE | - CCSR_SSI_SIER_RFF0_EN, 0); - break; - - default: - return -EINVAL; - } - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - write_ssi(CCSR_SSI_SOR_TX_CLR, &ssi->sor); - else - write_ssi(CCSR_SSI_SOR_RX_CLR, &ssi->sor); - - return 0; -} - -static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = { - .startup = fsl_ssi_startup, - .trigger = fsl_ssi_ac97_trigger, -}; - static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .ac97_control = 1, .playback = { @@ -1197,7 +1152,7 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &fsl_ssi_ac97_dai_ops, + .ops = &fsl_ssi_dai_ops, }; -- cgit v1.2.3-70-g09d2 From d8a64d6ade6a27dec2b8b37e4d9630c40a373bba Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 8 Jan 2014 17:42:18 +0000 Subject: ASoC: wm_adsp: Factor out ADSP2 boot proceedure Move the ADSP2 boot proceedure into a work structure in preparation for running it asynchronously with the reset of the audio path bring up. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 181 ++++++++++++++++++++++++++------------------- sound/soc/codecs/wm_adsp.h | 2 + 2 files changed, 105 insertions(+), 78 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 8f720ded27c..2087ae2eb32 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1492,107 +1492,131 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return 0; } -int wm_adsp2_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +void wm_adsp2_boot_work(struct work_struct *work) { - struct snd_soc_codec *codec = w->codec; - struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); - struct wm_adsp *dsp = &dsps[w->shift]; - struct wm_adsp_alg_region *alg_region; - struct wm_coeff_ctl *ctl; - unsigned int val; + struct wm_adsp *dsp = container_of(work, + struct wm_adsp, + boot_work); int ret; + unsigned int val; - dsp->card = codec->card; + /* + * For simplicity set the DSP clock rate to be the + * SYSCLK rate rather than making it configurable. + */ + ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val); + if (ret != 0) { + adsp_err(dsp, "Failed to read SYSCLK state: %d\n", ret); + return; + } + val = (val & ARIZONA_SYSCLK_FREQ_MASK) + >> ARIZONA_SYSCLK_FREQ_SHIFT; - switch (event) { - case SND_SOC_DAPM_POST_PMU: - /* - * For simplicity set the DSP clock rate to be the - * SYSCLK rate rather than making it configurable. - */ - ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val); - if (ret != 0) { - adsp_err(dsp, "Failed to read SYSCLK state: %d\n", - ret); - return ret; - } - val = (val & ARIZONA_SYSCLK_FREQ_MASK) - >> ARIZONA_SYSCLK_FREQ_SHIFT; + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CLOCKING, + ADSP2_CLK_SEL_MASK, val); + if (ret != 0) { + adsp_err(dsp, "Failed to set clock rate: %d\n", ret); + return; + } - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CLOCKING, - ADSP2_CLK_SEL_MASK, val); + if (dsp->dvfs) { + ret = regmap_read(dsp->regmap, + dsp->base + ADSP2_CLOCKING, &val); if (ret != 0) { - adsp_err(dsp, "Failed to set clock rate: %d\n", - ret); - return ret; + dev_err(dsp->dev, "Failed to read clocking: %d\n", ret); + return; } - if (dsp->dvfs) { - ret = regmap_read(dsp->regmap, - dsp->base + ADSP2_CLOCKING, &val); + if ((val & ADSP2_CLK_SEL_MASK) >= 3) { + ret = regulator_enable(dsp->dvfs); if (ret != 0) { dev_err(dsp->dev, - "Failed to read clocking: %d\n", ret); - return ret; + "Failed to enable supply: %d\n", + ret); + return; } - if ((val & ADSP2_CLK_SEL_MASK) >= 3) { - ret = regulator_enable(dsp->dvfs); - if (ret != 0) { - dev_err(dsp->dev, - "Failed to enable supply: %d\n", - ret); - return ret; - } - - ret = regulator_set_voltage(dsp->dvfs, - 1800000, - 1800000); - if (ret != 0) { - dev_err(dsp->dev, - "Failed to raise supply: %d\n", - ret); - return ret; - } + ret = regulator_set_voltage(dsp->dvfs, + 1800000, + 1800000); + if (ret != 0) { + dev_err(dsp->dev, + "Failed to raise supply: %d\n", + ret); + return; } } + } - ret = wm_adsp2_ena(dsp); - if (ret != 0) - return ret; + ret = wm_adsp2_ena(dsp); + if (ret != 0) + return; - ret = wm_adsp_load(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_load(dsp); + if (ret != 0) + goto err; - ret = wm_adsp_setup_algs(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_setup_algs(dsp); + if (ret != 0) + goto err; - ret = wm_adsp_load_coeff(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_load_coeff(dsp); + if (ret != 0) + goto err; - /* Initialize caches for enabled and unset controls */ - ret = wm_coeff_init_control_caches(dsp); - if (ret != 0) - goto err; + /* Initialize caches for enabled and unset controls */ + ret = wm_coeff_init_control_caches(dsp); + if (ret != 0) + goto err; - /* Sync set controls */ - ret = wm_coeff_sync_controls(dsp); - if (ret != 0) - goto err; + /* Sync set controls */ + ret = wm_coeff_sync_controls(dsp); + if (ret != 0) + goto err; + + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_CORE_ENA, + ADSP2_CORE_ENA); + if (ret != 0) + goto err; + + dsp->running = true; + + return; - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA | ADSP2_START, - ADSP2_CORE_ENA | ADSP2_START); +err: + regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0); +} + +int wm_adsp2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); + struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; + int ret; + + dsp->card = codec->card; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + queue_work(system_unbound_wq, &dsp->boot_work); + flush_work(&dsp->boot_work); + + if (!dsp->running) + return -EIO; + + ret = regmap_update_bits(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_START, + ADSP2_START); if (ret != 0) goto err; - - dsp->running = true; break; case SND_SOC_DAPM_PRE_PMD: @@ -1663,6 +1687,7 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) INIT_LIST_HEAD(&adsp->alg_regions); INIT_LIST_HEAD(&adsp->ctl_list); + INIT_WORK(&adsp->boot_work, wm_adsp2_boot_work); if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index d018dea6254..b172c1df915 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -59,6 +59,8 @@ struct wm_adsp { struct regulator *dvfs; struct list_head ctl_list; + + struct work_struct boot_work; }; #define WM_ADSP1(wname, num) \ -- cgit v1.2.3-70-g09d2 From 12db5edd6986a8358b92eb3fa6f8d2ee4fe1173b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 8 Jan 2014 17:42:19 +0000 Subject: ASoC: wm_adsp: Start DSP booting earlier in the DAPM process Move the start of booting the DSP to earlier in the DAPM process, and move the final starting of the DSP to later in the DAPM process. This allows us to overlap some of the processing with other components of the system being brought up. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.h | 17 +++++++++-------- sound/soc/codecs/wm_adsp.c | 24 +++++++++++++++++++++--- sound/soc/codecs/wm_adsp.h | 10 ++++++++-- 3 files changed, 38 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 9e81b639269..256548a5230 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -166,20 +166,21 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MIXER_INPUT_ROUTES(name " Input 4") #define ARIZONA_DSP_ROUTES(name) \ - { name, NULL, name " Aux 1" }, \ - { name, NULL, name " Aux 2" }, \ - { name, NULL, name " Aux 3" }, \ - { name, NULL, name " Aux 4" }, \ - { name, NULL, name " Aux 5" }, \ - { name, NULL, name " Aux 6" }, \ + { name, NULL, name " Preloader"}, \ + { name " Preloader", NULL, name " Aux 1" }, \ + { name " Preloader", NULL, name " Aux 2" }, \ + { name " Preloader", NULL, name " Aux 3" }, \ + { name " Preloader", NULL, name " Aux 4" }, \ + { name " Preloader", NULL, name " Aux 5" }, \ + { name " Preloader", NULL, name " Aux 6" }, \ ARIZONA_MIXER_INPUT_ROUTES(name " Aux 1"), \ ARIZONA_MIXER_INPUT_ROUTES(name " Aux 2"), \ ARIZONA_MIXER_INPUT_ROUTES(name " Aux 3"), \ ARIZONA_MIXER_INPUT_ROUTES(name " Aux 4"), \ ARIZONA_MIXER_INPUT_ROUTES(name " Aux 5"), \ ARIZONA_MIXER_INPUT_ROUTES(name " Aux 6"), \ - ARIZONA_MIXER_ROUTES(name, name "L"), \ - ARIZONA_MIXER_ROUTES(name, name "R") + ARIZONA_MIXER_ROUTES(name " Preloader", name "L"), \ + ARIZONA_MIXER_ROUTES(name " Preloader", name "R") #define ARIZONA_RATE_ENUM_SIZE 4 extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 2087ae2eb32..a061183add6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1591,6 +1591,27 @@ err: ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0); } +int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); + struct wm_adsp *dsp = &dsps[w->shift]; + + dsp->card = codec->card; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + queue_work(system_unbound_wq, &dsp->boot_work); + break; + default: + break; + }; + + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_early_event); + int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1601,11 +1622,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct wm_coeff_ctl *ctl; int ret; - dsp->card = codec->card; - switch (event) { case SND_SOC_DAPM_POST_PMU: - queue_work(system_unbound_wq, &dsp->boot_work); flush_work(&dsp->boot_work); if (!dsp->running) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index b172c1df915..a4f6b64deb6 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -68,8 +68,12 @@ struct wm_adsp { wm_adsp1_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) #define WM_ADSP2(wname, num) \ - SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ - wm_adsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) +{ .id = snd_soc_dapm_dai_link, .name = wname " Preloader", \ + .reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_early_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU }, \ +{ .id = snd_soc_dapm_out_drv, .name = wname, \ + .reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \ + .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } extern const struct snd_kcontrol_new wm_adsp1_fw_controls[]; extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; @@ -78,6 +82,8 @@ int wm_adsp1_init(struct wm_adsp *adsp); int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs); int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- cgit v1.2.3-70-g09d2 From 053ad6a057d168f9f09006c84a4be73f35b21da9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Jan 2014 10:37:07 +0100 Subject: ASoC: bcm: Remove obsoleted Kconfig dependency CONFIG_SND_SOC_DMAENGINE_PCM was renamed to CONFIG_SND_DMAENGINE_PCM recently. And yet we don't have to select it since CONFIG_SND_GENERIC_DMAENGINE_PCM selects the dependency by itself, so just rip it off. Signed-off-by: Takashi Iwai Acked-by: Florian Meier Signed-off-by: Mark Brown --- sound/soc/bcm/Kconfig | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig index 3d82a29ce3a..6a834e109f1 100644 --- a/sound/soc/bcm/Kconfig +++ b/sound/soc/bcm/Kconfig @@ -1,7 +1,6 @@ config SND_BCM2835_SOC_I2S tristate "SoC Audio support for the Broadcom BCM2835 I2S module" depends on ARCH_BCM2835 || COMPILE_TEST - select SND_SOC_DMAENGINE_PCM select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help -- cgit v1.2.3-70-g09d2 From 18b1a902ad55610b161bfc8fb905c372bb8372df Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 9 Jan 2014 09:06:54 +0000 Subject: ASoC: wm_adsp: Mark wm_adsp2_boot_work as static Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index a061183add6..f6e317c7845 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1492,7 +1492,7 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return 0; } -void wm_adsp2_boot_work(struct work_struct *work) +static void wm_adsp2_boot_work(struct work_struct *work) { struct wm_adsp *dsp = container_of(work, struct wm_adsp, -- cgit v1.2.3-70-g09d2 From d7fa71042304fbc43cfc81d199b922759c67e013 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Thu, 9 Jan 2014 11:16:11 +0100 Subject: ASoC: fsl-ssi: Fix stats compile warning single_open requires a function with signature 'int (*)(struct seq_file *, void *)'. This patch fixes the warning by fixing the wrong return type of fsl_ssi_stats_show. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 94dedcb0868..f662dddf208 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -391,7 +391,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) * Display the statistics for the current SSI device. To avoid confusion, * we only show those counts that are enabled. */ -static ssize_t fsl_ssi_stats_show(struct seq_file *s, void *unused) +static int fsl_ssi_stats_show(struct seq_file *s, void *unused) { struct fsl_ssi_private *ssi_private = s->private; -- cgit v1.2.3-70-g09d2 From fa69b0f93e3e383dc50df9529db67c09a1db3787 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 18:37:22 +0000 Subject: ASoC: ad1836: Use params_width() rather than explicit memory format Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/ad1836.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 9a92b7962f4..af490bebd7f 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -168,15 +168,15 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, int word_len = 0; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: word_len = AD1836_WORD_LEN_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: word_len = AD1836_WORD_LEN_20; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 24: + case 32: word_len = AD1836_WORD_LEN_24; break; } -- cgit v1.2.3-70-g09d2 From d4dd1fdf9ee320ef7fde77922c853c74a8cd3c7d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 18:38:20 +0000 Subject: ASoC: ad193x: Use params_width() rather than memory format Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/ad193x.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index aea7e52cf71..d6cdb3bb163 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -249,15 +249,15 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: word_len = 3; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: word_len = 1; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 24: + case 32: word_len = 0; break; } -- cgit v1.2.3-70-g09d2 From 7c2aff6ab53a24d6a688fd7ae45fc14a97f48eda Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 18:49:58 +0000 Subject: ASoC: adau1373: Use params_width() rather than memory format Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/adau1373.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 59654b1e7f3..eb836ed5271 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1078,17 +1078,17 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, (div << 2) | ADAU1373_BCLKDIV_64); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: ctrl = ADAU1373_DAI_WLEN_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: ctrl = ADAU1373_DAI_WLEN_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: ctrl = ADAU1373_DAI_WLEN_24; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: ctrl = ADAU1373_DAI_WLEN_32; break; default: -- cgit v1.2.3-70-g09d2 From 9b58e7163407f75ec150dc2f91f561fcb681753e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 18:50:25 +0000 Subject: ASoC: adau1701: Use params_width() rather than memory format Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/adau1701.c | 34 ++++++++++++++++------------------ 1 file changed, 16 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index ebff1128be5..52e3d83e26e 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -299,20 +299,20 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) } static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, - snd_pcm_format_t format) + struct snd_pcm_hw_params *params) { struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); unsigned int mask = ADAU1701_SEROCTL_WORD_LEN_MASK; unsigned int val; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: val = ADAU1701_SEROCTL_WORD_LEN_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: val = ADAU1701_SEROCTL_WORD_LEN_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: val = ADAU1701_SEROCTL_WORD_LEN_24; break; default: @@ -320,14 +320,14 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, } if (adau1701->dai_fmt == SND_SOC_DAIFMT_RIGHT_J) { - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: val |= ADAU1701_SEROCTL_MSB_DEALY16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: val |= ADAU1701_SEROCTL_MSB_DEALY12; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: val |= ADAU1701_SEROCTL_MSB_DEALY8; break; } @@ -340,7 +340,7 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, } static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, - snd_pcm_format_t format) + struct snd_pcm_hw_params *params) { struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); unsigned int val; @@ -348,14 +348,14 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, if (adau1701->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) return 0; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: val = ADAU1701_SERICTL_RIGHTJ_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: val = ADAU1701_SERICTL_RIGHTJ_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: val = ADAU1701_SERICTL_RIGHTJ_24; break; default: @@ -374,7 +374,6 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); unsigned int clkdiv = adau1701->sysclk / params_rate(params); - snd_pcm_format_t format; unsigned int val; int ret; @@ -406,11 +405,10 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_SR_MASK, val); - format = params_format(params); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return adau1701_set_playback_pcm_format(codec, format); + return adau1701_set_playback_pcm_format(codec, params); else - return adau1701_set_capture_pcm_format(codec, format); + return adau1701_set_capture_pcm_format(codec, params); } static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai, -- cgit v1.2.3-70-g09d2 From cf7d8b274f152f289bf9ef821f656133cd3401e4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 18:50:40 +0000 Subject: ASoC: adav80x: Use params_width() rather than memory format Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/adav80x.c | 30 ++++++++++++++---------------- 1 file changed, 14 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 14a7c169d00..371a0e9e1af 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -453,22 +453,22 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec, } static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, - struct snd_soc_dai *dai, snd_pcm_format_t format) + struct snd_soc_dai *dai, struct snd_pcm_hw_params *params) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: val = ADAV80X_CAPTURE_WORD_LEN16; break; - case SNDRV_PCM_FORMAT_S18_3LE: + case 18: val = ADAV80X_CAPTRUE_WORD_LEN18; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: val = ADAV80X_CAPTURE_WORD_LEN20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: val = ADAV80X_CAPTURE_WORD_LEN24; break; default: @@ -482,7 +482,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, } static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, - struct snd_soc_dai *dai, snd_pcm_format_t format) + struct snd_soc_dai *dai, struct snd_pcm_hw_params *params) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; @@ -490,17 +490,17 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J) return 0; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16; break; - case SNDRV_PCM_FORMAT_S18_3LE: + case 18: val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24; break; default: @@ -524,12 +524,10 @@ static int adav80x_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adav80x_set_playback_pcm_format(codec, dai, - params_format(params)); + adav80x_set_playback_pcm_format(codec, dai, params); adav80x_set_dac_clock(codec, rate); } else { - adav80x_set_capture_pcm_format(codec, dai, - params_format(params)); + adav80x_set_capture_pcm_format(codec, dai, params); adav80x_set_adc_clock(codec, rate); } adav80x->rate = rate; -- cgit v1.2.3-70-g09d2 From c098284a4bfb81b86331aadd08b482233c21fa2c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 14:19:10 +0100 Subject: ASoC: intel: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/sst_platform.c | 10 ---------- sound/soc/intel/sst_platform.h | 4 ---- 2 files changed, 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst_platform.c index b6b5eb698d3..f465a818086 100644 --- a/sound/soc/intel/sst_platform.c +++ b/sound/soc/intel/sst_platform.c @@ -89,16 +89,6 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 | - SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 | - SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32), - .rates = (SNDRV_PCM_RATE_8000| - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000), - .rate_min = SST_MIN_RATE, - .rate_max = SST_MAX_RATE, - .channels_min = SST_MIN_CHANNEL, - .channels_max = SST_MAX_CHANNEL, .buffer_bytes_max = SST_MAX_BUFFER, .period_bytes_min = SST_MIN_PERIOD_BYTES, .period_bytes_max = SST_MAX_PERIOD_BYTES, diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst_platform.h index cacc9066ec5..bee64fb7d2e 100644 --- a/sound/soc/intel/sst_platform.h +++ b/sound/soc/intel/sst_platform.h @@ -33,10 +33,6 @@ #define SST_STEREO 2 #define SST_MAX_CAP 5 -#define SST_MIN_RATE 8000 -#define SST_MAX_RATE 48000 -#define SST_MIN_CHANNEL 1 -#define SST_MAX_CHANNEL 5 #define SST_MAX_BUFFER (800*1024) #define SST_MIN_BUFFER (800*1024) #define SST_MIN_PERIOD_BYTES 32 -- cgit v1.2.3-70-g09d2 From 3317208c8838479a1cfe1ef395ec895d160957f0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 14:19:11 +0100 Subject: ASoC: kirkwood: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 16 ---------------- 1 file changed, 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 4af1936cf0f..aac22fccdcd 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -21,16 +21,6 @@ #include #include "kirkwood.h" -#define KIRKWOOD_RATES \ - (SNDRV_PCM_RATE_8000_192000 | \ - SNDRV_PCM_RATE_CONTINUOUS | \ - SNDRV_PCM_RATE_KNOT) - -#define KIRKWOOD_FORMATS \ - (SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) - static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) { struct snd_soc_pcm_runtime *soc_runtime = subs->private_data; @@ -43,12 +33,6 @@ static struct snd_pcm_hardware kirkwood_dma_snd_hw = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE), - .formats = KIRKWOOD_FORMATS, - .rates = KIRKWOOD_RATES, - .rate_min = 8000, - .rate_max = 384000, - .channels_min = 1, - .channels_max = 8, .buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES, .period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES, .period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES, -- cgit v1.2.3-70-g09d2 From 115367713460fd375380f5dc663271f07c513b33 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 14:19:13 +0100 Subject: ASoC: nuc900: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-pcm.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index f588ee45b4f..f434ed79d1b 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -32,9 +32,6 @@ static const struct snd_pcm_hardware nuc900_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .channels_min = 1, - .channels_max = 2, .buffer_bytes_max = 4*1024, .period_bytes_min = 1*1024, .period_bytes_max = 4*1024, -- cgit v1.2.3-70-g09d2 From a7ddf151b0eb12a8840d9d127f1679bb1c89a1ff Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 14:19:14 +0100 Subject: ASoC: sh: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/sh/dma-sh7760.c | 17 ----------------- sound/soc/sh/fsi.c | 6 ------ sound/soc/sh/rcar/core.c | 6 ------ 3 files changed, 29 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 1a8b03e4b41..c85f8eb66c9 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -89,29 +89,12 @@ struct camelot_pcm { #define DMABRG_PREALLOC_BUFFER 32 * 1024 #define DMABRG_PREALLOC_BUFFER_MAX 32 * 1024 -/* support everything the SSI supports */ -#define DMABRG_RATES \ - SNDRV_PCM_RATE_8000_192000 - -#define DMABRG_FMTS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ - SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ - SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ - SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) - static struct snd_pcm_hardware camelot_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH), - .formats = DMABRG_FMTS, - .rates = DMABRG_RATES, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 8, /* max of the SSI */ .buffer_bytes_max = DMABRG_PERIOD_MAX, .period_bytes_min = DMABRG_PERIOD_MIN, .period_bytes_max = DMABRG_PERIOD_MAX / 2, diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b33ca7cd085..ef89fa8e4fc 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1777,12 +1777,6 @@ static struct snd_pcm_hardware fsi_pcm_hardware = { SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE, - .formats = FSI_FMTS, - .rates = FSI_RATES, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 2, .buffer_bytes_max = 64 * 1024, .period_bytes_min = 32, .period_bytes_max = 8192, diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index b3653d37f75..743de5e3b1e 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -628,12 +628,6 @@ static struct snd_pcm_hardware rsnd_pcm_hardware = { SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE, - .formats = RSND_FMTS, - .rates = RSND_RATES, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 2, - .channels_max = 2, .buffer_bytes_max = 64 * 1024, .period_bytes_min = 32, .period_bytes_max = 8192, -- cgit v1.2.3-70-g09d2 From df021a72c92e8b9fe9b5d3f11105125484e8751f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 14:19:15 +0100 Subject: ASoC: ux500: Don't set unused struct snd_pcm_hardware fields The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_pcm.c | 15 --------------- 1 file changed, 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 55a8634cc3d..51a66a87305 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -28,12 +28,6 @@ #include "ux500_msp_i2s.h" #include "ux500_pcm.h" -#define UX500_PLATFORM_MIN_RATE 8000 -#define UX500_PLATFORM_MAX_RATE 48000 - -#define UX500_PLATFORM_MIN_CHANNELS 1 -#define UX500_PLATFORM_MAX_CHANNELS 8 - #define UX500_PLATFORM_PERIODS_BYTES_MIN 128 #define UX500_PLATFORM_PERIODS_BYTES_MAX (64 * PAGE_SIZE) #define UX500_PLATFORM_PERIODS_MIN 2 @@ -45,15 +39,6 @@ static const struct snd_pcm_hardware ux500_pcm_hw = { SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_S16_BE | - SNDRV_PCM_FMTBIT_U16_BE, - .rates = SNDRV_PCM_RATE_KNOT, - .rate_min = UX500_PLATFORM_MIN_RATE, - .rate_max = UX500_PLATFORM_MAX_RATE, - .channels_min = UX500_PLATFORM_MIN_CHANNELS, - .channels_max = UX500_PLATFORM_MAX_CHANNELS, .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, -- cgit v1.2.3-70-g09d2 From 16d7ea9167839d0b971ab29030886280595dd5fc Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 14:19:16 +0100 Subject: ASoC: Allow PCMs to restrict the supported formats Some DMA cores might add additional restrictions on which in memory audio formats can be supported by the compound sound card. If the PCM component specifies a set of formats it supports (by setting the formats field to non 0) take these into account when calculating the format set for the compound sound card. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 141a302e4e7..e7f16b54a97 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -158,7 +158,10 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, cpu_stream->channels_min); hw->channels_max = min(codec_stream->channels_max, cpu_stream->channels_max); - hw->formats = codec_stream->formats & cpu_stream->formats; + if (hw->formats) + hw->formats &= codec_stream->formats & cpu_stream->formats; + else + hw->formats = codec_stream->formats & cpu_stream->formats; hw->rates = codec_stream->rates & cpu_stream->rates; if (codec_stream->rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) -- cgit v1.2.3-70-g09d2 From e1cffe8c9f3a4f74b8b212c9fbe2873a8ee2f395 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Thu, 9 Jan 2014 22:27:31 +0800 Subject: ASoC: fsl-ssi: Add missing clk_disable_unprepare() on error in fsl_ssi_probe() Add the missing clk_disable_unprepare() before return from fsl_ssi_probe() in the request irq error handling case. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f662dddf208..6c2f040f49a 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1439,7 +1439,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ret < 0) { dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); - goto error_irqmap; + goto error_clk; } } -- cgit v1.2.3-70-g09d2 From 2b56b5f02029531007c8601b23f282b840715401 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 9 Jan 2014 18:42:48 +0800 Subject: ASoC: fsl_ssi: Set default slot number for common cases For those platforms using DAI master mode like I2S, it's better to pre-set a default slot number so that there's no need for these common cases to set the slot number from its machine driver any more. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6c2f040f49a..7864ec5cf5f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -711,6 +711,17 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) if (ssi_private->imx_ac97) fsl_ssi_setup_ac97(ssi_private); + /* + * Set a default slot number so that there is no need for those common + * cases like I2S mode to call the extra set_tdm_slot() any more. + */ + if (!ssi_private->imx_ac97) { + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + } + return 0; } -- cgit v1.2.3-70-g09d2 From 708ec0241c56b85176937e79314430f4f71e40c6 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Thu, 9 Jan 2014 17:19:08 +0800 Subject: ASoC: simple-card: fix a bug where cinfo will be NULL before using it If the dt is not used, the cinfo will be always NULL before using it. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0430be85f23..6c61b1758f7 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -213,8 +213,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } } } else { - cinfo->snd_card.dev = &pdev->dev; cinfo = pdev->dev.platform_data; + cinfo->snd_card.dev = &pdev->dev; } if (!cinfo) { -- cgit v1.2.3-70-g09d2 From 34787d0a258ebb3686676fb37a9e8717cbdd835a Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Thu, 9 Jan 2014 17:49:40 +0800 Subject: ASoC: simple-card: fix the cinfo error check If the dt is used and the cinfo is NULL, the -ENOMEM should be return. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6c61b1758f7..11030a63b81 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -211,15 +211,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dev_err(dev, "parse error %d\n", ret); return ret; } + } else { + return -ENOMEM; } } else { cinfo = pdev->dev.platform_data; - cinfo->snd_card.dev = &pdev->dev; - } + if (!cinfo) { + dev_err(dev, "no info for asoc-simple-card\n"); + return -EINVAL; + } - if (!cinfo) { - dev_err(dev, "no info for asoc-simple-card\n"); - return -EINVAL; + cinfo->snd_card.dev = &pdev->dev; } if (!cinfo->name || -- cgit v1.2.3-70-g09d2 From afb6d4ed3fd88bacf8b0abcbf053c79c604f509f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 9 Jan 2014 14:29:25 +0000 Subject: ASoC: wm5110: Add controls for headphone short circuit protection Add controls to enable/disable the headphone short circuit protection of the headphone outputs. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index eee627b9bd1..618fea33b3a 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -313,6 +313,13 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE), +SOC_SINGLE("HPOUT1 SC Protect Switch", ARIZONA_HP1_SHORT_CIRCUIT_CTRL, + ARIZONA_HP1_SC_ENA_SHIFT, 1, 0), +SOC_SINGLE("HPOUT2 SC Protect Switch", ARIZONA_HP2_SHORT_CIRCUIT_CTRL, + ARIZONA_HP2_SC_ENA_SHIFT, 1, 0), +SOC_SINGLE("HPOUT3 SC Protect Switch", ARIZONA_HP3_SHORT_CIRCUIT_CTRL, + ARIZONA_HP3_SC_ENA_SHIFT, 1, 0), + SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, -- cgit v1.2.3-70-g09d2 From 43d24e76b69826ce32292f47060ad78cdd0197fa Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 10 Jan 2014 17:54:06 +0800 Subject: ASoC: fsl_esai: Add ESAI CPU DAI driver This patch implements a device-tree-only CPU DAI driver for Freescale ESAI controller that supports: - 12 channels playback and 8 channels record. [ Some of the inner transmitters and receivers are sharing same group of pins. So the maxmium 12 output or 8 input channels are only valid if there is no pin conflict occurring to it. ] - Independent (asynchronous mode) or shared (synchronous mode) transmit and receive sections with separate or shared internal/external clocks and frame syncs, operating in Master or Slave mode. [ Current ALSA seems not to allow CPU DAI drivers to configure DAI format separately for PLAYBACK and CAPTURE. So this first version only supports the case that uses the same DAI format for both directions. ] - Various DAI formats: I2S, Left-Justified, Right-Justified, DSP-A and DSP-B. - Programmable word length (8, 16, 20 or 24bits) - Flexible selection between system clock or external oscillator as input clock source, programmable internal clock divider and frame sync generation. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,esai.txt | 50 ++ sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_esai.c | 815 +++++++++++++++++++++ sound/soc/fsl/fsl_esai.h | 354 +++++++++ 5 files changed, 1224 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.txt create mode 100644 sound/soc/fsl/fsl_esai.c create mode 100644 sound/soc/fsl/fsl_esai.h (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt new file mode 100644 index 00000000000..d7b99fa637b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -0,0 +1,50 @@ +Freescale Enhanced Serial Audio Interface (ESAI) Controller + +The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port +for serial communication with a variety of serial devices, including industry +standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and +other DSPs. It has up to six transmitters and four receivers. + +Required properties: + + - compatible : Compatible list, must contain "fsl,imx35-esai". + + - reg : Offset and length of the register set for the device. + + - interrupts : Contains the spdif interrupt. + + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. + + - dma-names : Two dmas have to be defined, "tx" and "rx". + + - clocks: Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "core" The core clock used to access registers + "extal" The esai baud clock for esai controller used to derive + HCK, SCK and FS. + "fsys" The system clock derived from ahb clock used to derive + HCK, SCK and FS. + + - fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. + This number is the maximum allowed value for TFCR[TFWM] or RFCR[RFWM]. + + - fsl,esai-synchronous: This is a boolean property. If present, indicating + that ESAI would work in the synchronous mode, which means all the settings + for Receiving would be duplicated from Transmition related registers. + +Example: + +esai: esai@02024000 { + compatible = "fsl,imx35-esai"; + reg = <0x02024000 0x4000>; + interrupts = <0 51 0x04>; + clocks = <&clks 208>, <&clks 118>, <&clks 208>; + clock-names = "core", "extal", "fsys"; + dmas = <&sdma 23 21 0>, <&sdma 24 21 0>; + dma-names = "rx", "tx"; + fsl,fifo-depth = <128>; + fsl,esai-synchronous; + status = "disabled"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index ac4fe4ea15a..f2f39dd13bc 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -8,6 +8,9 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate +config SND_SOC_FSL_ESAI + tristate + config SND_SOC_FSL_UTILS tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index aaccbee1700..b12ad4b9b4d 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -14,11 +14,13 @@ obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-spdif-objs := fsl_spdif.o +snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o +obj-$(CONFIG_SND_SOC_FSL_ESAI) += snd-soc-fsl-esai.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c new file mode 100644 index 00000000000..d0c72ed261e --- /dev/null +++ b/sound/soc/fsl/fsl_esai.c @@ -0,0 +1,815 @@ +/* + * Freescale ESAI ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "fsl_esai.h" +#include "imx-pcm.h" + +#define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000 +#define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/** + * fsl_esai: ESAI private data + * + * @dma_params_rx: DMA parameters for receive channel + * @dma_params_tx: DMA parameters for transmit channel + * @pdev: platform device pointer + * @regmap: regmap handler + * @coreclk: clock source to access register + * @extalclk: esai clock source to derive HCK, SCK and FS + * @fsysclk: system clock source to derive HCK, SCK and FS + * @fifo_depth: depth of tx/rx FIFO + * @slot_width: width of each DAI slot + * @hck_rate: clock rate of desired HCKx clock + * @sck_div: if using PSR/PM dividers for SCKx clock + * @slave_mode: if fully using DAI slave mode + * @synchronous: if using tx/rx synchronous mode + * @name: driver name + */ +struct fsl_esai { + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct platform_device *pdev; + struct regmap *regmap; + struct clk *coreclk; + struct clk *extalclk; + struct clk *fsysclk; + u32 fifo_depth; + u32 slot_width; + u32 hck_rate[2]; + bool sck_div[2]; + bool slave_mode; + bool synchronous; + char name[32]; +}; + +static irqreturn_t esai_isr(int irq, void *devid) +{ + struct fsl_esai *esai_priv = (struct fsl_esai *)devid; + struct platform_device *pdev = esai_priv->pdev; + u32 esr; + + regmap_read(esai_priv->regmap, REG_ESAI_ESR, &esr); + + if (esr & ESAI_ESR_TINIT_MASK) + dev_dbg(&pdev->dev, "isr: Transmition Initialized\n"); + + if (esr & ESAI_ESR_RFF_MASK) + dev_warn(&pdev->dev, "isr: Receiving overrun\n"); + + if (esr & ESAI_ESR_TFE_MASK) + dev_warn(&pdev->dev, "isr: Transmition underrun\n"); + + if (esr & ESAI_ESR_TLS_MASK) + dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n"); + + if (esr & ESAI_ESR_TDE_MASK) + dev_dbg(&pdev->dev, "isr: Transmition data exception\n"); + + if (esr & ESAI_ESR_TED_MASK) + dev_dbg(&pdev->dev, "isr: Transmitting even slots\n"); + + if (esr & ESAI_ESR_TD_MASK) + dev_dbg(&pdev->dev, "isr: Transmitting data\n"); + + if (esr & ESAI_ESR_RLS_MASK) + dev_dbg(&pdev->dev, "isr: Just received the last slot\n"); + + if (esr & ESAI_ESR_RDE_MASK) + dev_dbg(&pdev->dev, "isr: Receiving data exception\n"); + + if (esr & ESAI_ESR_RED_MASK) + dev_dbg(&pdev->dev, "isr: Receiving even slots\n"); + + if (esr & ESAI_ESR_RD_MASK) + dev_dbg(&pdev->dev, "isr: Receiving data\n"); + + return IRQ_HANDLED; +} + +/** + * This function is used to calculate the divisors of psr, pm, fp and it is + * supposed to be called in set_dai_sysclk() and set_bclk(). + * + * @ratio: desired overall ratio for the paticipating dividers + * @usefp: for HCK setting, there is no need to set fp divider + * @fp: bypass other dividers by setting fp directly if fp != 0 + * @tx: current setting is for playback or capture + */ +static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio, + bool usefp, u32 fp) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 psr, pm = 999, maxfp, prod, sub, savesub, i, j; + + maxfp = usefp ? 16 : 1; + + if (usefp && fp) + goto out_fp; + + if (ratio > 2 * 8 * 256 * maxfp || ratio < 2) { + dev_err(dai->dev, "the ratio is out of range (2 ~ %d)\n", + 2 * 8 * 256 * maxfp); + return -EINVAL; + } else if (ratio % 2) { + dev_err(dai->dev, "the raio must be even if using upper divider\n"); + return -EINVAL; + } + + ratio /= 2; + + psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8; + + /* Set the max fluctuation -- 0.1% of the max devisor */ + savesub = (psr ? 1 : 8) * 256 * maxfp / 1000; + + /* Find the best value for PM */ + for (i = 1; i <= 256; i++) { + for (j = 1; j <= maxfp; j++) { + /* PSR (1 or 8) * PM (1 ~ 256) * FP (1 ~ 16) */ + prod = (psr ? 1 : 8) * i * j; + + if (prod == ratio) + sub = 0; + else if (prod / ratio == 1) + sub = prod - ratio; + else if (ratio / prod == 1) + sub = ratio - prod; + else + continue; + + /* Calculate the fraction */ + sub = sub * 1000 / ratio; + if (sub < savesub) { + savesub = sub; + pm = i; + fp = j; + } + + /* We are lucky */ + if (savesub == 0) + goto out; + } + } + + if (pm == 999) { + dev_err(dai->dev, "failed to calculate proper divisors\n"); + return -EINVAL; + } + +out: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), + ESAI_xCCR_xPSR_MASK | ESAI_xCCR_xPM_MASK, + psr | ESAI_xCCR_xPM(pm)); + +out_fp: + /* Bypass fp if not being required */ + if (maxfp <= 1) + return 0; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), + ESAI_xCCR_xFP_MASK, ESAI_xCCR_xFP(fp)); + + return 0; +} + +/** + * This function mainly configures the clock frequency of MCLK (HCKT/HCKR) + * + * @Parameters: + * clk_id: The clock source of HCKT/HCKR + * (Input from outside; output from inside, FSYS or EXTAL) + * freq: The required clock rate of HCKT/HCKR + * dir: The clock direction of HCKT/HCKR + * + * Note: If the direction is input, we do not care about clk_id. + */ +static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + struct clk *clksrc = esai_priv->extalclk; + bool tx = clk_id <= ESAI_HCKT_EXTAL; + bool in = dir == SND_SOC_CLOCK_IN; + u32 ret, ratio, ecr = 0; + unsigned long clk_rate; + + /* sck_div can be only bypassed if ETO/ERO=0 and SNC_SOC_CLOCK_OUT */ + esai_priv->sck_div[tx] = true; + + /* Set the direction of HCKT/HCKR pins */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), + ESAI_xCCR_xHCKD, in ? 0 : ESAI_xCCR_xHCKD); + + if (in) + goto out; + + switch (clk_id) { + case ESAI_HCKT_FSYS: + case ESAI_HCKR_FSYS: + clksrc = esai_priv->fsysclk; + break; + case ESAI_HCKT_EXTAL: + ecr |= ESAI_ECR_ETI; + case ESAI_HCKR_EXTAL: + ecr |= ESAI_ECR_ERI; + break; + default: + return -EINVAL; + } + + if (IS_ERR(clksrc)) { + dev_err(dai->dev, "no assigned %s clock\n", + clk_id % 2 ? "extal" : "fsys"); + return PTR_ERR(clksrc); + } + clk_rate = clk_get_rate(clksrc); + + ratio = clk_rate / freq; + if (ratio * freq > clk_rate) + ret = ratio * freq - clk_rate; + else if (ratio * freq < clk_rate) + ret = clk_rate - ratio * freq; + else + ret = 0; + + /* Block if clock source can not be divided into the required rate */ + if (ret != 0 && clk_rate / ret < 1000) { + dev_err(dai->dev, "failed to derive required HCK%c rate\n", + tx ? 'T' : 'R'); + return -EINVAL; + } + + if (ratio == 1) { + /* Bypass all the dividers if not being needed */ + ecr |= tx ? ESAI_ECR_ETO : ESAI_ECR_ERO; + goto out; + } + + ret = fsl_esai_divisor_cal(dai, tx, ratio, false, 0); + if (ret) + return ret; + + esai_priv->sck_div[tx] = false; + +out: + esai_priv->hck_rate[tx] = freq; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR, + tx ? ESAI_ECR_ETI | ESAI_ECR_ETO : + ESAI_ECR_ERI | ESAI_ECR_ERO, ecr); + + return 0; +} + +/** + * This function configures the related dividers according to the bclk rate + */ +static int fsl_esai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 hck_rate = esai_priv->hck_rate[tx]; + u32 sub, ratio = hck_rate / freq; + + /* Don't apply for fully slave mode*/ + if (esai_priv->slave_mode) + return 0; + + if (ratio * freq > hck_rate) + sub = ratio * freq - hck_rate; + else if (ratio * freq < hck_rate) + sub = hck_rate - ratio * freq; + else + sub = 0; + + /* Block if clock source can not be divided into the required rate */ + if (sub != 0 && hck_rate / sub < 1000) { + dev_err(dai->dev, "failed to derive required SCK%c rate\n", + tx ? 'T' : 'R'); + return -EINVAL; + } + + if (esai_priv->sck_div[tx] && (ratio > 16 || ratio == 0)) { + dev_err(dai->dev, "the ratio is out of range (1 ~ 16)\n"); + return -EINVAL; + } + + return fsl_esai_divisor_cal(dai, tx, ratio, true, + esai_priv->sck_div[tx] ? 0 : ratio); +} + +static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, + ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, + ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask)); + + esai_priv->slot_width = slot_width; + + return 0; +} + +static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 xcr = 0, xccr = 0, mask; + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* Data on rising edge of bclk, frame low, 1clk before data */ + xcr |= ESAI_xCR_xFSR; + xccr |= ESAI_xCCR_xFSP | ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + /* Data on rising edge of bclk, frame high, right aligned */ + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + xcr |= ESAI_xCR_xFSL | ESAI_xCR_xFSR; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + xcr |= ESAI_xCR_xFSL; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + xccr ^= ESAI_xCCR_xFSP; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP; + break; + default: + return -EINVAL; + } + + esai_priv->slave_mode = false; + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + esai_priv->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBS_CFM: + xccr |= ESAI_xCCR_xCKD; + break; + case SND_SOC_DAIFMT_CBM_CFS: + xccr |= ESAI_xCCR_xFSD; + break; + case SND_SOC_DAIFMT_CBS_CFS: + xccr |= ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; + break; + default: + return -EINVAL; + } + + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); + + mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); + + return 0; +} + +static int fsl_esai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + /* + * Some platforms might use the same bit to gate all three or two of + * clocks, so keep all clocks open/close at the same time for safety + */ + clk_prepare_enable(esai_priv->coreclk); + if (!IS_ERR(esai_priv->extalclk)) + clk_prepare_enable(esai_priv->extalclk); + if (!IS_ERR(esai_priv->fsysclk)) + clk_prepare_enable(esai_priv->fsysclk); + + if (!dai->active) { + /* Reset Port C */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, + ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, + ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO)); + + /* Set synchronous mode */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_SAICR, + ESAI_SAICR_SYNC, esai_priv->synchronous ? + ESAI_SAICR_SYNC : 0); + + /* Set a default slot number -- 2 */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + } + + return 0; +} + +static int fsl_esai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 width = snd_pcm_format_width(params_format(params)); + u32 channels = params_channels(params); + u32 bclk, mask, val, ret; + + bclk = params_rate(params) * esai_priv->slot_width * 2; + + ret = fsl_esai_set_bclk(dai, tx, bclk); + if (ret) + return ret; + + /* Use Normal mode to support monaural audio */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ? + ESAI_xCR_xMOD_NETWORK : 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR); + + mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK | + (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK); + val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) | + (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); + + mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0); + val = ESAI_xCR_xSWS(esai_priv->slot_width, width) | (tx ? ESAI_xCR_PADC : 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + + return 0; +} + +static void fsl_esai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + if (!IS_ERR(esai_priv->fsysclk)) + clk_disable_unprepare(esai_priv->fsysclk); + if (!IS_ERR(esai_priv->extalclk)) + clk_disable_unprepare(esai_priv->extalclk); + clk_disable_unprepare(esai_priv->coreclk); +} + +static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u8 i, channels = substream->runtime->channels; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFEN_MASK, ESAI_xFCR_xFEN); + + /* Write initial words reqiured by ESAI as normal procedure */ + for (i = 0; tx && i < channels; i++) + regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, + tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels)); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); + + /* Disable and reset FIFO */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops fsl_esai_dai_ops = { + .startup = fsl_esai_startup, + .shutdown = fsl_esai_shutdown, + .trigger = fsl_esai_trigger, + .hw_params = fsl_esai_hw_params, + .set_sysclk = fsl_esai_set_dai_sysclk, + .set_fmt = fsl_esai_set_dai_fmt, + .set_tdm_slot = fsl_esai_set_dai_tdm_slot, +}; + +static int fsl_esai_dai_probe(struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &esai_priv->dma_params_tx, + &esai_priv->dma_params_rx); + + return 0; +} + +static struct snd_soc_dai_driver fsl_esai_dai = { + .probe = fsl_esai_dai_probe, + .playback = { + .channels_min = 1, + .channels_max = 12, + .rates = FSL_ESAI_RATES, + .formats = FSL_ESAI_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = FSL_ESAI_RATES, + .formats = FSL_ESAI_FORMATS, + }, + .ops = &fsl_esai_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_esai_component = { + .name = "fsl-esai", +}; + +static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ERDR: + case REG_ESAI_ECR: + case REG_ESAI_ESR: + case REG_ESAI_TFCR: + case REG_ESAI_TFSR: + case REG_ESAI_RFCR: + case REG_ESAI_RFSR: + case REG_ESAI_RX0: + case REG_ESAI_RX1: + case REG_ESAI_RX2: + case REG_ESAI_RX3: + case REG_ESAI_SAISR: + case REG_ESAI_SAICR: + case REG_ESAI_TCR: + case REG_ESAI_TCCR: + case REG_ESAI_RCR: + case REG_ESAI_RCCR: + case REG_ESAI_TSMA: + case REG_ESAI_TSMB: + case REG_ESAI_RSMA: + case REG_ESAI_RSMB: + case REG_ESAI_PRRC: + case REG_ESAI_PCRC: + return true; + default: + return false; + } +} + +static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ETDR: + case REG_ESAI_ECR: + case REG_ESAI_TFCR: + case REG_ESAI_RFCR: + case REG_ESAI_TX0: + case REG_ESAI_TX1: + case REG_ESAI_TX2: + case REG_ESAI_TX3: + case REG_ESAI_TX4: + case REG_ESAI_TX5: + case REG_ESAI_TSR: + case REG_ESAI_SAICR: + case REG_ESAI_TCR: + case REG_ESAI_TCCR: + case REG_ESAI_RCR: + case REG_ESAI_RCCR: + case REG_ESAI_TSMA: + case REG_ESAI_TSMB: + case REG_ESAI_RSMA: + case REG_ESAI_RSMB: + case REG_ESAI_PRRC: + case REG_ESAI_PCRC: + return true; + default: + return false; + } +} + +static const struct regmap_config fsl_esai_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = REG_ESAI_PCRC, + .readable_reg = fsl_esai_readable_reg, + .writeable_reg = fsl_esai_writeable_reg, +}; + +static int fsl_esai_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_esai *esai_priv; + struct resource *res; + const uint32_t *iprop; + void __iomem *regs; + int irq, ret; + + esai_priv = devm_kzalloc(&pdev->dev, sizeof(*esai_priv), GFP_KERNEL); + if (!esai_priv) + return -ENOMEM; + + esai_priv->pdev = pdev; + strcpy(esai_priv->name, np->name); + + /* Get the addresses and IRQ */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + esai_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "core", regs, &fsl_esai_regmap_config); + if (IS_ERR(esai_priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap: %ld\n", + PTR_ERR(esai_priv->regmap)); + return PTR_ERR(esai_priv->regmap); + } + + esai_priv->coreclk = devm_clk_get(&pdev->dev, "core"); + if (IS_ERR(esai_priv->coreclk)) { + dev_err(&pdev->dev, "failed to get core clock: %ld\n", + PTR_ERR(esai_priv->coreclk)); + return PTR_ERR(esai_priv->coreclk); + } + + esai_priv->extalclk = devm_clk_get(&pdev->dev, "extal"); + if (IS_ERR(esai_priv->extalclk)) + dev_warn(&pdev->dev, "failed to get extal clock: %ld\n", + PTR_ERR(esai_priv->extalclk)); + + esai_priv->fsysclk = devm_clk_get(&pdev->dev, "fsys"); + if (IS_ERR(esai_priv->fsysclk)) + dev_warn(&pdev->dev, "failed to get fsys clock: %ld\n", + PTR_ERR(esai_priv->fsysclk)); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0, + esai_priv->name, esai_priv); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + return ret; + } + + /* Set a default slot size */ + esai_priv->slot_width = 32; + + /* Set a default master/slave state */ + esai_priv->slave_mode = true; + + /* Determine the FIFO depth */ + iprop = of_get_property(np, "fsl,fifo-depth", NULL); + if (iprop) + esai_priv->fifo_depth = be32_to_cpup(iprop); + else + esai_priv->fifo_depth = 64; + + esai_priv->dma_params_tx.maxburst = 16; + esai_priv->dma_params_rx.maxburst = 16; + esai_priv->dma_params_tx.addr = res->start + REG_ESAI_ETDR; + esai_priv->dma_params_rx.addr = res->start + REG_ESAI_ERDR; + + esai_priv->synchronous = + of_property_read_bool(np, "fsl,esai-synchronous"); + + /* Implement full symmetry for synchronous mode */ + if (esai_priv->synchronous) { + fsl_esai_dai.symmetric_rates = 1; + fsl_esai_dai.symmetric_channels = 1; + fsl_esai_dai.symmetric_samplebits = 1; + } + + dev_set_drvdata(&pdev->dev, esai_priv); + + /* Reset ESAI unit */ + ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ERST); + if (ret) { + dev_err(&pdev->dev, "failed to reset ESAI: %d\n", ret); + return ret; + } + + /* + * We need to enable ESAI so as to access some of its registers. + * Otherwise, we would fail to dump regmap from user space. + */ + ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ESAIEN); + if (ret) { + dev_err(&pdev->dev, "failed to enable ESAI: %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, + &fsl_esai_dai, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); + return ret; + } + + ret = imx_pcm_dma_init(pdev); + if (ret) + dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); + + return ret; +} + +static const struct of_device_id fsl_esai_dt_ids[] = { + { .compatible = "fsl,imx35-esai", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); + +static struct platform_driver fsl_esai_driver = { + .probe = fsl_esai_probe, + .driver = { + .name = "fsl-esai-dai", + .owner = THIS_MODULE, + .of_match_table = fsl_esai_dt_ids, + }, +}; + +module_platform_driver(fsl_esai_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale ESAI CPU DAI driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:fsl-esai-dai"); diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h new file mode 100644 index 00000000000..9c9f957fcae --- /dev/null +++ b/sound/soc/fsl/fsl_esai.h @@ -0,0 +1,354 @@ +/* + * fsl_esai.h - ALSA ESAI interface for the Freescale i.MX SoC + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_ESAI_DAI_H +#define _FSL_ESAI_DAI_H + +/* ESAI Register Map */ +#define REG_ESAI_ETDR 0x00 +#define REG_ESAI_ERDR 0x04 +#define REG_ESAI_ECR 0x08 +#define REG_ESAI_ESR 0x0C +#define REG_ESAI_TFCR 0x10 +#define REG_ESAI_TFSR 0x14 +#define REG_ESAI_RFCR 0x18 +#define REG_ESAI_RFSR 0x1C +#define REG_ESAI_xFCR(tx) (tx ? REG_ESAI_TFCR : REG_ESAI_RFCR) +#define REG_ESAI_xFSR(tx) (tx ? REG_ESAI_TFSR : REG_ESAI_RFSR) +#define REG_ESAI_TX0 0x80 +#define REG_ESAI_TX1 0x84 +#define REG_ESAI_TX2 0x88 +#define REG_ESAI_TX3 0x8C +#define REG_ESAI_TX4 0x90 +#define REG_ESAI_TX5 0x94 +#define REG_ESAI_TSR 0x98 +#define REG_ESAI_RX0 0xA0 +#define REG_ESAI_RX1 0xA4 +#define REG_ESAI_RX2 0xA8 +#define REG_ESAI_RX3 0xAC +#define REG_ESAI_SAISR 0xCC +#define REG_ESAI_SAICR 0xD0 +#define REG_ESAI_TCR 0xD4 +#define REG_ESAI_TCCR 0xD8 +#define REG_ESAI_RCR 0xDC +#define REG_ESAI_RCCR 0xE0 +#define REG_ESAI_xCR(tx) (tx ? REG_ESAI_TCR : REG_ESAI_RCR) +#define REG_ESAI_xCCR(tx) (tx ? REG_ESAI_TCCR : REG_ESAI_RCCR) +#define REG_ESAI_TSMA 0xE4 +#define REG_ESAI_TSMB 0xE8 +#define REG_ESAI_RSMA 0xEC +#define REG_ESAI_RSMB 0xF0 +#define REG_ESAI_xSMA(tx) (tx ? REG_ESAI_TSMA : REG_ESAI_RSMA) +#define REG_ESAI_xSMB(tx) (tx ? REG_ESAI_TSMB : REG_ESAI_RSMB) +#define REG_ESAI_PRRC 0xF8 +#define REG_ESAI_PCRC 0xFC + +/* ESAI Control Register -- REG_ESAI_ECR 0x8 */ +#define ESAI_ECR_ETI_SHIFT 19 +#define ESAI_ECR_ETI_MASK (1 << ESAI_ECR_ETI_SHIFT) +#define ESAI_ECR_ETI (1 << ESAI_ECR_ETI_SHIFT) +#define ESAI_ECR_ETO_SHIFT 18 +#define ESAI_ECR_ETO_MASK (1 << ESAI_ECR_ETO_SHIFT) +#define ESAI_ECR_ETO (1 << ESAI_ECR_ETO_SHIFT) +#define ESAI_ECR_ERI_SHIFT 17 +#define ESAI_ECR_ERI_MASK (1 << ESAI_ECR_ERI_SHIFT) +#define ESAI_ECR_ERI (1 << ESAI_ECR_ERI_SHIFT) +#define ESAI_ECR_ERO_SHIFT 16 +#define ESAI_ECR_ERO_MASK (1 << ESAI_ECR_ERO_SHIFT) +#define ESAI_ECR_ERO (1 << ESAI_ECR_ERO_SHIFT) +#define ESAI_ECR_ERST_SHIFT 1 +#define ESAI_ECR_ERST_MASK (1 << ESAI_ECR_ERST_SHIFT) +#define ESAI_ECR_ERST (1 << ESAI_ECR_ERST_SHIFT) +#define ESAI_ECR_ESAIEN_SHIFT 0 +#define ESAI_ECR_ESAIEN_MASK (1 << ESAI_ECR_ESAIEN_SHIFT) +#define ESAI_ECR_ESAIEN (1 << ESAI_ECR_ESAIEN_SHIFT) + +/* ESAI Status Register -- REG_ESAI_ESR 0xC */ +#define ESAI_ESR_TINIT_SHIFT 10 +#define ESAI_ESR_TINIT_MASK (1 << ESAI_ESR_TINIT_SHIFT) +#define ESAI_ESR_TINIT (1 << ESAI_ESR_TINIT_SHIFT) +#define ESAI_ESR_RFF_SHIFT 9 +#define ESAI_ESR_RFF_MASK (1 << ESAI_ESR_RFF_SHIFT) +#define ESAI_ESR_RFF (1 << ESAI_ESR_RFF_SHIFT) +#define ESAI_ESR_TFE_SHIFT 8 +#define ESAI_ESR_TFE_MASK (1 << ESAI_ESR_TFE_SHIFT) +#define ESAI_ESR_TFE (1 << ESAI_ESR_TFE_SHIFT) +#define ESAI_ESR_TLS_SHIFT 7 +#define ESAI_ESR_TLS_MASK (1 << ESAI_ESR_TLS_SHIFT) +#define ESAI_ESR_TLS (1 << ESAI_ESR_TLS_SHIFT) +#define ESAI_ESR_TDE_SHIFT 6 +#define ESAI_ESR_TDE_MASK (1 << ESAI_ESR_TDE_SHIFT) +#define ESAI_ESR_TDE (1 << ESAI_ESR_TDE_SHIFT) +#define ESAI_ESR_TED_SHIFT 5 +#define ESAI_ESR_TED_MASK (1 << ESAI_ESR_TED_SHIFT) +#define ESAI_ESR_TED (1 << ESAI_ESR_TED_SHIFT) +#define ESAI_ESR_TD_SHIFT 4 +#define ESAI_ESR_TD_MASK (1 << ESAI_ESR_TD_SHIFT) +#define ESAI_ESR_TD (1 << ESAI_ESR_TD_SHIFT) +#define ESAI_ESR_RLS_SHIFT 3 +#define ESAI_ESR_RLS_MASK (1 << ESAI_ESR_RLS_SHIFT) +#define ESAI_ESR_RLS (1 << ESAI_ESR_RLS_SHIFT) +#define ESAI_ESR_RDE_SHIFT 2 +#define ESAI_ESR_RDE_MASK (1 << ESAI_ESR_RDE_SHIFT) +#define ESAI_ESR_RDE (1 << ESAI_ESR_RDE_SHIFT) +#define ESAI_ESR_RED_SHIFT 1 +#define ESAI_ESR_RED_MASK (1 << ESAI_ESR_RED_SHIFT) +#define ESAI_ESR_RED (1 << ESAI_ESR_RED_SHIFT) +#define ESAI_ESR_RD_SHIFT 0 +#define ESAI_ESR_RD_MASK (1 << ESAI_ESR_RD_SHIFT) +#define ESAI_ESR_RD (1 << ESAI_ESR_RD_SHIFT) + +/* + * Transmit FIFO Configuration Register -- REG_ESAI_TFCR 0x10 + * Receive FIFO Configuration Register -- REG_ESAI_RFCR 0x18 + */ +#define ESAI_xFCR_TIEN_SHIFT 19 +#define ESAI_xFCR_TIEN_MASK (1 << ESAI_xFCR_TIEN_SHIFT) +#define ESAI_xFCR_TIEN (1 << ESAI_xFCR_TIEN_SHIFT) +#define ESAI_xFCR_REXT_SHIFT 19 +#define ESAI_xFCR_REXT_MASK (1 << ESAI_xFCR_REXT_SHIFT) +#define ESAI_xFCR_REXT (1 << ESAI_xFCR_REXT_SHIFT) +#define ESAI_xFCR_xWA_SHIFT 16 +#define ESAI_xFCR_xWA_WIDTH 3 +#define ESAI_xFCR_xWA_MASK (((1 << ESAI_xFCR_xWA_WIDTH) - 1) << ESAI_xFCR_xWA_SHIFT) +#define ESAI_xFCR_xWA(v) (((8 - ((v) >> 2)) << ESAI_xFCR_xWA_SHIFT) & ESAI_xFCR_xWA_MASK) +#define ESAI_xFCR_xFWM_SHIFT 8 +#define ESAI_xFCR_xFWM_WIDTH 8 +#define ESAI_xFCR_xFWM_MASK (((1 << ESAI_xFCR_xFWM_WIDTH) - 1) << ESAI_xFCR_xFWM_SHIFT) +#define ESAI_xFCR_xFWM(v) ((((v) - 1) << ESAI_xFCR_xFWM_SHIFT) & ESAI_xFCR_xFWM_MASK) +#define ESAI_xFCR_xE_SHIFT 2 +#define ESAI_xFCR_TE_WIDTH 6 +#define ESAI_xFCR_RE_WIDTH 4 +#define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) +#define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) +#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK) +#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK) +#define ESAI_xFCR_xFR_SHIFT 1 +#define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT) +#define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT) +#define ESAI_xFCR_xFEN_SHIFT 0 +#define ESAI_xFCR_xFEN_MASK (1 << ESAI_xFCR_xFEN_SHIFT) +#define ESAI_xFCR_xFEN (1 << ESAI_xFCR_xFEN_SHIFT) + +/* + * Transmit FIFO Status Register -- REG_ESAI_TFSR 0x14 + * Receive FIFO Status Register --REG_ESAI_RFSR 0x1C + */ +#define ESAI_xFSR_NTFO_SHIFT 12 +#define ESAI_xFSR_NRFI_SHIFT 12 +#define ESAI_xFSR_NTFI_SHIFT 8 +#define ESAI_xFSR_NRFO_SHIFT 8 +#define ESAI_xFSR_NTFx_WIDTH 3 +#define ESAI_xFSR_NRFx_WIDTH 2 +#define ESAI_xFSR_NTFO_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFO_SHIFT) +#define ESAI_xFSR_NTFI_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFI_SHIFT) +#define ESAI_xFSR_NRFO_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFO_SHIFT) +#define ESAI_xFSR_NRFI_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFI_SHIFT) +#define ESAI_xFSR_xFCNT_SHIFT 0 +#define ESAI_xFSR_xFCNT_WIDTH 8 +#define ESAI_xFSR_xFCNT_MASK (((1 << ESAI_xFSR_xFCNT_WIDTH) - 1) << ESAI_xFSR_xFCNT_SHIFT) + +/* ESAI Transmit Slot Register -- REG_ESAI_TSR 0x98 */ +#define ESAI_TSR_SHIFT 0 +#define ESAI_TSR_WIDTH 24 +#define ESAI_TSR_MASK (((1 << ESAI_TSR_WIDTH) - 1) << ESAI_TSR_SHIFT) + +/* Serial Audio Interface Status Register -- REG_ESAI_SAISR 0xCC */ +#define ESAI_SAISR_TODFE_SHIFT 17 +#define ESAI_SAISR_TODFE_MASK (1 << ESAI_SAISR_TODFE_SHIFT) +#define ESAI_SAISR_TODFE (1 << ESAI_SAISR_TODFE_SHIFT) +#define ESAI_SAISR_TEDE_SHIFT 16 +#define ESAI_SAISR_TEDE_MASK (1 << ESAI_SAISR_TEDE_SHIFT) +#define ESAI_SAISR_TEDE (1 << ESAI_SAISR_TEDE_SHIFT) +#define ESAI_SAISR_TDE_SHIFT 15 +#define ESAI_SAISR_TDE_MASK (1 << ESAI_SAISR_TDE_SHIFT) +#define ESAI_SAISR_TDE (1 << ESAI_SAISR_TDE_SHIFT) +#define ESAI_SAISR_TUE_SHIFT 14 +#define ESAI_SAISR_TUE_MASK (1 << ESAI_SAISR_TUE_SHIFT) +#define ESAI_SAISR_TUE (1 << ESAI_SAISR_TUE_SHIFT) +#define ESAI_SAISR_TFS_SHIFT 13 +#define ESAI_SAISR_TFS_MASK (1 << ESAI_SAISR_TFS_SHIFT) +#define ESAI_SAISR_TFS (1 << ESAI_SAISR_TFS_SHIFT) +#define ESAI_SAISR_RODF_SHIFT 10 +#define ESAI_SAISR_RODF_MASK (1 << ESAI_SAISR_RODF_SHIFT) +#define ESAI_SAISR_RODF (1 << ESAI_SAISR_RODF_SHIFT) +#define ESAI_SAISR_REDF_SHIFT 9 +#define ESAI_SAISR_REDF_MASK (1 << ESAI_SAISR_REDF_SHIFT) +#define ESAI_SAISR_REDF (1 << ESAI_SAISR_REDF_SHIFT) +#define ESAI_SAISR_RDF_SHIFT 8 +#define ESAI_SAISR_RDF_MASK (1 << ESAI_SAISR_RDF_SHIFT) +#define ESAI_SAISR_RDF (1 << ESAI_SAISR_RDF_SHIFT) +#define ESAI_SAISR_ROE_SHIFT 7 +#define ESAI_SAISR_ROE_MASK (1 << ESAI_SAISR_ROE_SHIFT) +#define ESAI_SAISR_ROE (1 << ESAI_SAISR_ROE_SHIFT) +#define ESAI_SAISR_RFS_SHIFT 6 +#define ESAI_SAISR_RFS_MASK (1 << ESAI_SAISR_RFS_SHIFT) +#define ESAI_SAISR_RFS (1 << ESAI_SAISR_RFS_SHIFT) +#define ESAI_SAISR_IF2_SHIFT 2 +#define ESAI_SAISR_IF2_MASK (1 << ESAI_SAISR_IF2_SHIFT) +#define ESAI_SAISR_IF2 (1 << ESAI_SAISR_IF2_SHIFT) +#define ESAI_SAISR_IF1_SHIFT 1 +#define ESAI_SAISR_IF1_MASK (1 << ESAI_SAISR_IF1_SHIFT) +#define ESAI_SAISR_IF1 (1 << ESAI_SAISR_IF1_SHIFT) +#define ESAI_SAISR_IF0_SHIFT 0 +#define ESAI_SAISR_IF0_MASK (1 << ESAI_SAISR_IF0_SHIFT) +#define ESAI_SAISR_IF0 (1 << ESAI_SAISR_IF0_SHIFT) + +/* Serial Audio Interface Control Register -- REG_ESAI_SAICR 0xD0 */ +#define ESAI_SAICR_ALC_SHIFT 8 +#define ESAI_SAICR_ALC_MASK (1 << ESAI_SAICR_ALC_SHIFT) +#define ESAI_SAICR_ALC (1 << ESAI_SAICR_ALC_SHIFT) +#define ESAI_SAICR_TEBE_SHIFT 7 +#define ESAI_SAICR_TEBE_MASK (1 << ESAI_SAICR_TEBE_SHIFT) +#define ESAI_SAICR_TEBE (1 << ESAI_SAICR_TEBE_SHIFT) +#define ESAI_SAICR_SYNC_SHIFT 6 +#define ESAI_SAICR_SYNC_MASK (1 << ESAI_SAICR_SYNC_SHIFT) +#define ESAI_SAICR_SYNC (1 << ESAI_SAICR_SYNC_SHIFT) +#define ESAI_SAICR_OF2_SHIFT 2 +#define ESAI_SAICR_OF2_MASK (1 << ESAI_SAICR_OF2_SHIFT) +#define ESAI_SAICR_OF2 (1 << ESAI_SAICR_OF2_SHIFT) +#define ESAI_SAICR_OF1_SHIFT 1 +#define ESAI_SAICR_OF1_MASK (1 << ESAI_SAICR_OF1_SHIFT) +#define ESAI_SAICR_OF1 (1 << ESAI_SAICR_OF1_SHIFT) +#define ESAI_SAICR_OF0_SHIFT 0 +#define ESAI_SAICR_OF0_MASK (1 << ESAI_SAICR_OF0_SHIFT) +#define ESAI_SAICR_OF0 (1 << ESAI_SAICR_OF0_SHIFT) + +/* + * Transmit Control Register -- REG_ESAI_TCR 0xD4 + * Receive Control Register -- REG_ESAI_RCR 0xDC + */ +#define ESAI_xCR_xLIE_SHIFT 23 +#define ESAI_xCR_xLIE_MASK (1 << ESAI_xCR_xLIE_SHIFT) +#define ESAI_xCR_xLIE (1 << ESAI_xCR_xLIE_SHIFT) +#define ESAI_xCR_xIE_SHIFT 22 +#define ESAI_xCR_xIE_MASK (1 << ESAI_xCR_xIE_SHIFT) +#define ESAI_xCR_xIE (1 << ESAI_xCR_xIE_SHIFT) +#define ESAI_xCR_xEDIE_SHIFT 21 +#define ESAI_xCR_xEDIE_MASK (1 << ESAI_xCR_xEDIE_SHIFT) +#define ESAI_xCR_xEDIE (1 << ESAI_xCR_xEDIE_SHIFT) +#define ESAI_xCR_xEIE_SHIFT 20 +#define ESAI_xCR_xEIE_MASK (1 << ESAI_xCR_xEIE_SHIFT) +#define ESAI_xCR_xEIE (1 << ESAI_xCR_xEIE_SHIFT) +#define ESAI_xCR_xPR_SHIFT 19 +#define ESAI_xCR_xPR_MASK (1 << ESAI_xCR_xPR_SHIFT) +#define ESAI_xCR_xPR (1 << ESAI_xCR_xPR_SHIFT) +#define ESAI_xCR_PADC_SHIFT 17 +#define ESAI_xCR_PADC_MASK (1 << ESAI_xCR_PADC_SHIFT) +#define ESAI_xCR_PADC (1 << ESAI_xCR_PADC_SHIFT) +#define ESAI_xCR_xFSR_SHIFT 16 +#define ESAI_xCR_xFSR_MASK (1 << ESAI_xCR_xFSR_SHIFT) +#define ESAI_xCR_xFSR (1 << ESAI_xCR_xFSR_SHIFT) +#define ESAI_xCR_xFSL_SHIFT 15 +#define ESAI_xCR_xFSL_MASK (1 << ESAI_xCR_xFSL_SHIFT) +#define ESAI_xCR_xFSL (1 << ESAI_xCR_xFSL_SHIFT) +#define ESAI_xCR_xSWS_SHIFT 10 +#define ESAI_xCR_xSWS_WIDTH 5 +#define ESAI_xCR_xSWS_MASK (((1 << ESAI_xCR_xSWS_WIDTH) - 1) << ESAI_xCR_xSWS_SHIFT) +#define ESAI_xCR_xSWS(s, w) ((w < 24 ? (s - w + ((w - 8) >> 2)) : (s < 32 ? 0x1e : 0x1f)) << ESAI_xCR_xSWS_SHIFT) +#define ESAI_xCR_xMOD_SHIFT 8 +#define ESAI_xCR_xMOD_WIDTH 2 +#define ESAI_xCR_xMOD_MASK (((1 << ESAI_xCR_xMOD_WIDTH) - 1) << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_ONDEMAND (0x1 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_NETWORK (0x1 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_AC97 (0x3 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xWA_SHIFT 7 +#define ESAI_xCR_xWA_MASK (1 << ESAI_xCR_xWA_SHIFT) +#define ESAI_xCR_xWA (1 << ESAI_xCR_xWA_SHIFT) +#define ESAI_xCR_xSHFD_SHIFT 6 +#define ESAI_xCR_xSHFD_MASK (1 << ESAI_xCR_xSHFD_SHIFT) +#define ESAI_xCR_xSHFD (1 << ESAI_xCR_xSHFD_SHIFT) +#define ESAI_xCR_xE_SHIFT 0 +#define ESAI_xCR_TE_WIDTH 6 +#define ESAI_xCR_RE_WIDTH 4 +#define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) +#define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) +#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK) +#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK) + +/* + * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8 + * Receive Clock Control Register -- REG_ESAI_RCCR 0xE0 + */ +#define ESAI_xCCR_xHCKD_SHIFT 23 +#define ESAI_xCCR_xHCKD_MASK (1 << ESAI_xCCR_xHCKD_SHIFT) +#define ESAI_xCCR_xHCKD (1 << ESAI_xCCR_xHCKD_SHIFT) +#define ESAI_xCCR_xFSD_SHIFT 22 +#define ESAI_xCCR_xFSD_MASK (1 << ESAI_xCCR_xFSD_SHIFT) +#define ESAI_xCCR_xFSD (1 << ESAI_xCCR_xFSD_SHIFT) +#define ESAI_xCCR_xCKD_SHIFT 21 +#define ESAI_xCCR_xCKD_MASK (1 << ESAI_xCCR_xCKD_SHIFT) +#define ESAI_xCCR_xCKD (1 << ESAI_xCCR_xCKD_SHIFT) +#define ESAI_xCCR_xHCKP_SHIFT 20 +#define ESAI_xCCR_xHCKP_MASK (1 << ESAI_xCCR_xHCKP_SHIFT) +#define ESAI_xCCR_xHCKP (1 << ESAI_xCCR_xHCKP_SHIFT) +#define ESAI_xCCR_xFSP_SHIFT 19 +#define ESAI_xCCR_xFSP_MASK (1 << ESAI_xCCR_xFSP_SHIFT) +#define ESAI_xCCR_xFSP (1 << ESAI_xCCR_xFSP_SHIFT) +#define ESAI_xCCR_xCKP_SHIFT 18 +#define ESAI_xCCR_xCKP_MASK (1 << ESAI_xCCR_xCKP_SHIFT) +#define ESAI_xCCR_xCKP (1 << ESAI_xCCR_xCKP_SHIFT) +#define ESAI_xCCR_xFP_SHIFT 14 +#define ESAI_xCCR_xFP_WIDTH 4 +#define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) +#define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) +#define ESAI_xCCR_xDC_SHIFT 9 +#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) +#define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) +#define ESAI_xCCR_xPSR_SHIFT 8 +#define ESAI_xCCR_xPSR_MASK (1 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPSR_BYPASS (1 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPSR_DIV8 (0 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPM_SHIFT 0 +#define ESAI_xCCR_xPM_WIDTH 8 +#define ESAI_xCCR_xPM_MASK (((1 << ESAI_xCCR_xPM_WIDTH) - 1) << ESAI_xCCR_xPM_SHIFT) +#define ESAI_xCCR_xPM(v) ((((v) - 1) << ESAI_xCCR_xPM_SHIFT) & ESAI_xCCR_xPM_MASK) + +/* Transmit Slot Mask Register A/B -- REG_ESAI_TSMA/B 0xE4 ~ 0xF0 */ +#define ESAI_xSMA_xS_SHIFT 0 +#define ESAI_xSMA_xS_WIDTH 16 +#define ESAI_xSMA_xS_MASK (((1 << ESAI_xSMA_xS_WIDTH) - 1) << ESAI_xSMA_xS_SHIFT) +#define ESAI_xSMA_xS(v) ((v) & ESAI_xSMA_xS_MASK) +#define ESAI_xSMB_xS_SHIFT 0 +#define ESAI_xSMB_xS_WIDTH 16 +#define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT) +#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK) + +/* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */ +#define ESAI_PRRC_PDC_SHIFT 0 +#define ESAI_PRRC_PDC_WIDTH 12 +#define ESAI_PRRC_PDC_MASK (((1 << ESAI_PRRC_PDC_WIDTH) - 1) << ESAI_PRRC_PDC_SHIFT) +#define ESAI_PRRC_PDC(v) ((v) & ESAI_PRRC_PDC_MASK) + +/* Port C Control Register -- REG_ESAI_PCRC 0xFC */ +#define ESAI_PCRC_PC_SHIFT 0 +#define ESAI_PCRC_PC_WIDTH 12 +#define ESAI_PCRC_PC_MASK (((1 << ESAI_PCRC_PC_WIDTH) - 1) << ESAI_PCRC_PC_SHIFT) +#define ESAI_PCRC_PC(v) ((v) & ESAI_PCRC_PC_MASK) + +#define ESAI_GPIO 0xfff + +/* ESAI clock source */ +#define ESAI_HCKT_FSYS 0 +#define ESAI_HCKT_EXTAL 1 +#define ESAI_HCKR_FSYS 2 +#define ESAI_HCKR_EXTAL 3 + +/* ESAI clock divider */ +#define ESAI_TX_DIV_PSR 0 +#define ESAI_TX_DIV_PM 1 +#define ESAI_TX_DIV_FP 2 +#define ESAI_RX_DIV_PSR 3 +#define ESAI_RX_DIV_PM 4 +#define ESAI_RX_DIV_FP 5 +#endif /* _FSL_ESAI_DAI_H */ -- cgit v1.2.3-70-g09d2 From 4a608b3af38c6a98d1a3269703292137156407f8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 19:28:07 +0000 Subject: ASoC: alc5623: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 256c364193a..d3036283482 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -714,17 +714,17 @@ static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, iface &= ~ALC5623_DAI_I2S_DL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: iface |= ALC5623_DAI_I2S_DL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= ALC5623_DAI_I2S_DL_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= ALC5623_DAI_I2S_DL_24; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= ALC5623_DAI_I2S_DL_32; break; default: -- cgit v1.2.3-70-g09d2 From 2dad2283c5c1aaef5467e4cda67110ee236d7726 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 19:28:27 +0000 Subject: ASoC: alc5632: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 19e9f222d09..fb001c56cf8 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -869,14 +869,14 @@ static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, iface &= ~ALC5632_DAI_I2S_DL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: iface |= ALC5632_DAI_I2S_DL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= ALC5632_DAI_I2S_DL_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= ALC5632_DAI_I2S_DL_24; break; default: -- cgit v1.2.3-70-g09d2 From 1b6b0dfac283635eebf92b3bbb62ae5be898cea0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 19:48:20 +0000 Subject: ASoC: cs42l51: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 1e0fa3b5f79..6e9ea8379a9 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -423,21 +423,17 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream, intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_LJ24); break; case SND_SOC_DAIFMT_RIGHT_J: - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - case SNDRV_PCM_FORMAT_S16_BE: + switch (params_width(params)) { + case 16: fmt = CS42L51_DAC_DIF_RJ16; break; - case SNDRV_PCM_FORMAT_S18_3LE: - case SNDRV_PCM_FORMAT_S18_3BE: + case 18: fmt = CS42L51_DAC_DIF_RJ18; break; - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S20_3BE: + case 20: fmt = CS42L51_DAC_DIF_RJ20; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S24_BE: + case 24: fmt = CS42L51_DAC_DIF_RJ24; break; default: -- cgit v1.2.3-70-g09d2 From 0194c42a8f7e2b992558eb0bfd2274f850340782 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 19:49:37 +0000 Subject: ASoC: da7210: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 9c123145650..85b307c24b9 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -778,17 +778,17 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: dai_cfg1 |= DA7210_DAI_WORD_S16_LE; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: dai_cfg1 |= DA7210_DAI_WORD_S20_3LE; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: dai_cfg1 |= DA7210_DAI_WORD_S24_LE; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: dai_cfg1 |= DA7210_DAI_WORD_S32_LE; break; default: -- cgit v1.2.3-70-g09d2 From e7610743d4f4d54c2de32ae8f28fbd50922463d3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 19:49:51 +0000 Subject: ASoC: da7213: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 4a6f1daf911..0c77e7ad742 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1067,17 +1067,17 @@ static int da7213_hw_params(struct snd_pcm_substream *substream, u8 fs; /* Set DAI format */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: dai_ctrl |= DA7213_DAI_WORD_LENGTH_S16_LE; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: dai_ctrl |= DA7213_DAI_WORD_LENGTH_S20_LE; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: dai_ctrl |= DA7213_DAI_WORD_LENGTH_S24_LE; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: dai_ctrl |= DA7213_DAI_WORD_LENGTH_S32_LE; break; default: -- cgit v1.2.3-70-g09d2 From abf82ae6a6a875ff04e882aaf4dade40d5b0a794 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 19:50:07 +0000 Subject: ASoC: da732x: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index dc0284dc9e6..f295b656991 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -973,17 +973,17 @@ static int da732x_hw_params(struct snd_pcm_substream *substream, reg_aif = dai->driver->base; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: aif |= DA732X_AIF_WORD_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aif |= DA732X_AIF_WORD_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aif |= DA732X_AIF_WORD_24; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aif |= DA732X_AIF_WORD_32; break; default: -- cgit v1.2.3-70-g09d2 From 2822a9d01cc8132c82c8fad81df084f45af6a8e2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 19:50:21 +0000 Subject: ASoC: da9055: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/da9055.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index fc9802d1281..52b79a487ac 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1058,17 +1058,17 @@ static int da9055_hw_params(struct snd_pcm_substream *substream, u8 aif_ctrl, fs; u32 sysclk; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: aif_ctrl = DA9055_AIF_WORD_S16_LE; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aif_ctrl = DA9055_AIF_WORD_S20_3LE; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aif_ctrl = DA9055_AIF_WORD_S24_LE; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aif_ctrl = DA9055_AIF_WORD_S32_LE; break; default: -- cgit v1.2.3-70-g09d2 From 359e2ae8974550c65d0d85711c9f86aa6ed215d8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 19:50:38 +0000 Subject: ASoC: isabelle: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/isabelle.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 53b455b8c07..5839048ec46 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -951,11 +951,11 @@ static int isabelle_hw_params(struct snd_pcm_substream *substream, ISABELLE_FS_RATE_MASK, fs_val); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S20_3LE: + switch (params_width(params)) { + case 20: aif |= ISABELLE_AIF_LENGTH_20; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aif |= ISABELLE_AIF_LENGTH_32; break; default: -- cgit v1.2.3-70-g09d2 From 793f77036d5ca91d0dcfff16c7ae05d9116ce34a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 20:39:22 +0000 Subject: ASoC: max98088: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 53d7dab4e05..ee660e2d3df 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1233,12 +1233,12 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, M98088_DAI_WS, 0); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, M98088_DAI_WS, M98088_DAI_WS); break; -- cgit v1.2.3-70-g09d2 From 7821afc4865e976c55403bdb13d798a133efc815 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 20:39:30 +0000 Subject: ASoC: max98090: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 0569a4c3ae0..51f9b3d16b4 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1840,8 +1840,8 @@ static int max98090_dai_hw_params(struct snd_pcm_substream *substream, max98090->lrclk = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: snd_soc_update_bits(codec, M98090_REG_INTERFACE_FORMAT, M98090_WS_MASK, 0); break; -- cgit v1.2.3-70-g09d2 From 580ce08d5c1a96aeb0e3434bb5144defb6a334a2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 20:39:37 +0000 Subject: ASoC: max98095: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 67244315c72..3ba1170ebb5 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1213,12 +1213,12 @@ static int max98095_dai1_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: snd_soc_update_bits(codec, M98095_02A_DAI1_FORMAT, M98095_DAI_WS, 0); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: snd_soc_update_bits(codec, M98095_02A_DAI1_FORMAT, M98095_DAI_WS, M98095_DAI_WS); break; -- cgit v1.2.3-70-g09d2 From 0058e459600c87b03aad1842474b68a7cf6211ca Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jan 2014 20:39:44 +0000 Subject: ASoC: max9850: Use params_width() rather than memory format Signed-off-by: Mark Brown --- sound/soc/codecs/max9850.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index c5dd61785f8..82757ebf030 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -149,14 +149,14 @@ static int max9850_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, MAX9850_LRCLK_MSB, (lrclk_div >> 8) & 0x7f); snd_soc_write(codec, MAX9850_LRCLK_LSB, lrclk_div & 0xff); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: da = 0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: da = 0x2; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: da = 0x3; break; default: -- cgit v1.2.3-70-g09d2 From ba194a4de5c81ee200b6af88743b26f7b69aa659 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 13 Jan 2014 17:08:08 +0800 Subject: ASoC: simple-card: use snd_soc_card_set/get_drvdata Remove asoc_simple_get_card_info macro and use snd_soc_card_set_drvdata and snd_soc_card_get_drvdata instead. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 11030a63b81..5528dd6a4e4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -14,9 +14,6 @@ #include #include -#define asoc_simple_get_card_info(p) \ - container_of(p->dai_link, struct asoc_simple_card_info, snd_link) - static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -41,7 +38,8 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd); + struct asoc_simple_card_info *info = + snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; unsigned int daifmt = info->daifmt; @@ -256,6 +254,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; + snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); + return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); } -- cgit v1.2.3-70-g09d2 From 817873f4b155b22a24c48d6a38ee32007e2d856e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 10:24:40 +0100 Subject: ASoC: pcm: Properly initialize hw->rate_max If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll end up with the rate_max field of the runtime hardware set to 0. (Note that it is still possible for the components to constrain the supported sample rates using other methods, e.g. setting a list constraint) If rate_max is 0 this means that the sound card doesn't support any rates at all, which is not the desired result. So initialize rate_max to UINT_MAX. For symmetry reasons also set rate_min to 0. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1a617fde46e..2b8949647e3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -170,6 +170,9 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) hw->rates |= codec_stream->rates; + hw->rate_min = 0; + hw->rate_max = UINT_MAX; + snd_pcm_limit_hw_rates(runtime); hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); -- cgit v1.2.3-70-g09d2 From 24710c97960ac343c613786d250a1e0063555faa Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 10:24:41 +0100 Subject: ASoC: fsl: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates SNDRV_PCM_RATE_CONTINUOUS means that all rates (possibly limited to a certain interval) are supported. There is no need to manually set other rate bits. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 3 +-- sound/soc/fsl/mpc5200_psc_i2s.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 35e277379b8..dd5e6a76d29 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -79,8 +79,7 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) * ALSA that we support all rates and let the codec driver decide what rates * are really supported. */ -#define FSLSSI_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ - SNDRV_PCM_RATE_CONTINUOUS) +#define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS /** * FSLSSI_I2S_FORMATS: audio formats supported by the SSI diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index f4efaadb80a..5d07e8a74a2 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -26,8 +26,7 @@ * ALSA that we support all rates and let the codec driver decide what rates * are really supported. */ -#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ - SNDRV_PCM_RATE_CONTINUOUS) +#define PSC_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS /** * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode -- cgit v1.2.3-70-g09d2 From bf103eb4af73596edbab5faab67e29ea1e87c769 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 10:24:42 +0100 Subject: ASoC: s6000: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit SNDRV_PCM_RATE_CONTINUOUS means that all rates (possibly limited to a certain interval) are supported. There is no need to manually set other rate bits. Signed-off-by: Lars-Peter Clausen Acked-by: Daniel Glöckner Signed-off-by: Mark Brown --- sound/soc/s6000/s6000-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 73bb99f0109..7eba7979b9a 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -405,8 +405,7 @@ static int s6000_i2s_dai_probe(struct snd_soc_dai *dai) return 0; } -#define S6000_I2S_RATES (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_192000) +#define S6000_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS #define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops s6000_i2s_dai_ops = { -- cgit v1.2.3-70-g09d2 From 55dcdb5051930dee75e9e2c0da90bc82ee3dcd77 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 10:24:44 +0100 Subject: ASoC: pcm: Use snd_pcm_rate_mask_intersect() helper Instead of open-coding the intersecting of two rate masks (and getting slightly wrong for some of the corner cases) use the new snd_pcm_rate_mask_intersect() helper function. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2b8949647e3..4bbda0a4ee0 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -162,13 +162,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, hw->formats &= codec_stream->formats & cpu_stream->formats; else hw->formats = codec_stream->formats & cpu_stream->formats; - hw->rates = codec_stream->rates & cpu_stream->rates; - if (codec_stream->rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - hw->rates |= cpu_stream->rates; - if (cpu_stream->rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - hw->rates |= codec_stream->rates; + hw->rates = snd_pcm_rate_mask_intersect(codec_stream->rates, + cpu_stream->rates); hw->rate_min = 0; hw->rate_max = UINT_MAX; -- cgit v1.2.3-70-g09d2 From ca919fe4b972b9428ab42bead11b04a4ebf0f632 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 14 Jan 2014 12:35:32 +0800 Subject: ASoC: simple-card: fix one bug to writing to the platform data It's a bug that writing to the platform data directly, for it should be constant. So just copy it before writing. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 40 +++++++++++++++++++++------------------- 1 file changed, 21 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5528dd6a4e4..53395f54849 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -9,9 +9,10 @@ * published by the Free Software Foundation. */ #include +#include #include #include -#include +#include #include static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, @@ -190,36 +191,37 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; + int ret; cinfo = NULL; of_cpu = NULL; of_codec = NULL; of_platform = NULL; + + cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); + if (!cinfo) + return -ENOMEM; + if (np && of_device_is_available(np)) { - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (cinfo) { - int ret; - cinfo->snd_card.dev = &pdev->dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, - &of_cpu, - &of_codec, - &of_platform); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - return ret; - } - } else { - return -ENOMEM; + cinfo->snd_card.dev = dev; + + ret = asoc_simple_card_parse_of(np, cinfo, dev, + &of_cpu, + &of_codec, + &of_platform); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "parse error %d\n", ret); + return ret; } } else { - cinfo = pdev->dev.platform_data; - if (!cinfo) { + if (!dev->platform_data) { dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; + memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); + cinfo->snd_card.dev = dev; } if (!cinfo->name || -- cgit v1.2.3-70-g09d2 From 93b943edfc5e439f7b843535e0bb0f7d2371f67f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 14:02:18 +0100 Subject: ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime Currently we have two different snd_soc_platform_driver structs in the generic dmaengine PCM driver. One for dmaengine drivers that support residue reporting and one for those which do not. When registering the PCM component we check whether the NO_RESIDUE flag is set or not and use the corresponding snd_soc_platform_driver. This patch modifies the driver to only have one snd_soc_platform_driver struct where the pointer() callback checks the NO_RESIDUE flag at runtime. This allows us to set the NO_RESIDUE flag after the PCM component has been registered. This becomes necessary when querying whether the dmaengine driver supports residue reporting from the dmaengine driver itself since the DMA channel might only be requested after the PCM component has been registered. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 39 ++++++++++++++--------------------- 1 file changed, 15 insertions(+), 24 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 2a6c569d991..4e2bed89a4a 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -248,6 +248,18 @@ err_free: return ret; } +static snd_pcm_uframes_t dmaengine_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) + return snd_dmaengine_pcm_pointer_no_residue(substream); + else + return snd_dmaengine_pcm_pointer(substream); +} + static const struct snd_pcm_ops dmaengine_pcm_ops = { .open = dmaengine_pcm_open, .close = snd_dmaengine_pcm_close, @@ -255,7 +267,7 @@ static const struct snd_pcm_ops dmaengine_pcm_ops = { .hw_params = dmaengine_pcm_hw_params, .hw_free = snd_pcm_lib_free_pages, .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, + .pointer = dmaengine_pcm_pointer, }; static const struct snd_soc_platform_driver dmaengine_pcm_platform = { @@ -265,23 +277,6 @@ static const struct snd_soc_platform_driver dmaengine_pcm_platform = { .probe_order = SND_SOC_COMP_ORDER_LATE, }; -static const struct snd_pcm_ops dmaengine_no_residue_pcm_ops = { - .open = dmaengine_pcm_open, - .close = snd_dmaengine_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = dmaengine_pcm_hw_params, - .hw_free = snd_pcm_lib_free_pages, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer_no_residue, -}; - -static const struct snd_soc_platform_driver dmaengine_no_residue_pcm_platform = { - .ops = &dmaengine_no_residue_pcm_ops, - .pcm_new = dmaengine_pcm_new, - .pcm_free = dmaengine_pcm_free, - .probe_order = SND_SOC_COMP_ORDER_LATE, -}; - static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_PLAYBACK] = "tx", [SNDRV_PCM_STREAM_CAPTURE] = "rx", @@ -374,12 +369,8 @@ int snd_dmaengine_pcm_register(struct device *dev, if (ret) goto err_free_dma; - if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) - ret = snd_soc_add_platform(dev, &pcm->platform, - &dmaengine_no_residue_pcm_platform); - else - ret = snd_soc_add_platform(dev, &pcm->platform, - &dmaengine_pcm_platform); + ret = snd_soc_add_platform(dev, &pcm->platform, + &dmaengine_pcm_platform); if (ret) goto err_free_dma; -- cgit v1.2.3-70-g09d2 From 478028e088d6a94666d8a776be2cd2291faf3bbd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 14:02:19 +0100 Subject: ASoC: generic-dmaengine-pcm: Check DMA residue granularity The dmaengine framework now exposes the granularity with which it is able to report the transfer residue for a certain DMA channel. Check the granularity in the generic dmaengine PCM driver and a) Set the SNDRV_PCM_INFO_BATCH if the granularity is per period or worse. b) Fallback to the (race condition prone) period counting if the driver does not support any residue reporting. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 4e2bed89a4a..560a7787d8a 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -144,6 +144,8 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea if (ret == 0) { if (dma_caps.cmd_pause) hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; + if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) + hw.info |= SNDRV_PCM_INFO_BATCH; } return snd_soc_set_runtime_hwparams(substream, &hw); @@ -187,6 +189,21 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( dma_data->filter_data); } +static bool dmaengine_pcm_can_report_residue(struct dma_chan *chan) +{ + struct dma_slave_caps dma_caps; + int ret; + + ret = dma_get_slave_caps(chan, &dma_caps); + if (ret != 0) + return true; + + if (dma_caps.residue_granularity == DMA_RESIDUE_GRANULARITY_DESCRIPTOR) + return false; + + return true; +} + static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); @@ -239,6 +256,16 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) max_buffer_size); if (ret) goto err_free; + + /* + * This will only return false if we know for sure that at least + * one channel does not support residue reporting. If the DMA + * driver does not implement the slave_caps API we rely having + * the NO_RESIDUE flag set manually in case residue reporting is + * not supported. + */ + if (!dmaengine_pcm_can_report_residue(pcm->chan[i])) + pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE; } return 0; -- cgit v1.2.3-70-g09d2 From 153e66f5136bc5b33db253ad2db011177196626e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 14:02:20 +0100 Subject: ASoC: axi-{spdif,i2s}: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag The pl330 driver properly reports that it does not have residue reporting support, which means the PCM dmanegine driver is able to figure this out on its own. So there is no need to set the flag manually. Removing the flag has the advantage that once the pl330 driver gains support for residue reporting it will automatically be used by the generic dmaengine PCM driver. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/adi/axi-i2s.c | 3 +-- sound/soc/adi/axi-spdif.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 7f91a86dd73..6058c1fd507 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -236,8 +236,7 @@ static int axi_i2s_probe(struct platform_device *pdev) if (ret) goto err_clk_disable; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) goto err_clk_disable; diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index 8db7a992069..198e3a4640f 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -229,8 +229,7 @@ static int axi_spdif_probe(struct platform_device *pdev) if (ret) goto err_clk_disable; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) goto err_clk_disable; -- cgit v1.2.3-70-g09d2 From d70e861a3154833467023123e218e9b1ba558809 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 14:02:21 +0100 Subject: ASoC: samsung: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag The Samsung dmaengine ASoC driver is used with two different dmaengine drivers. The pl80x, which properly supports residue reporting and the pl330, which reports that it does not support residue reporting. So there is no need to manually set the NO_RESIDUE flag. This has the advantage that a proper (race condition free) PCM pointer() implementation is used when the pl80x driver is used. Also once the pl330 driver supports residue reporting the ASoC PCM driver will automatically start using it. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/dmaengine.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index 3be479d51b9..750ce5808d9 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -68,7 +68,6 @@ int samsung_asoc_dma_platform_register(struct device *dev) { return snd_dmaengine_pcm_register(dev, &samsung_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME | - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_COMPAT); } EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); -- cgit v1.2.3-70-g09d2 From f7d3c17096f6cbca8f0113d5a092ffcc72c7bf41 Mon Sep 17 00:00:00 2001 From: Arun Shamanna Lakshmi Date: Tue, 14 Jan 2014 15:31:54 -0800 Subject: ASoC: dapm: Change prototype of soc_widget_read soc_widget_read API returns the register data and it is possible that a register can contain 0xffffffff. Thus, change the prototype of soc_widget_read to return only the error code and pass the reg data through pointer argument. Signed-off-by: Arun Shamanna Lakshmi Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 51b4c192f41..2a44fe9122a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -371,12 +371,16 @@ static void dapm_reset(struct snd_soc_card *card) } } -static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) +static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg, + unsigned int *value) { - if (w->codec) - return snd_soc_read(w->codec, reg); - else if (w->platform) - return snd_soc_platform_read(w->platform, reg); + if (w->codec) { + *value = snd_soc_read(w->codec, reg); + return 0; + } else if (w->platform) { + *value = snd_soc_platform_read(w->platform, reg); + return 0; + } dev_err(w->dapm->dev, "ASoC: no valid widget read method\n"); return -1; @@ -430,13 +434,12 @@ static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, return ret; } else { soc_widget_lock(w); - ret = soc_widget_read(w, reg); + ret = soc_widget_read(w, reg, &old); if (ret < 0) { soc_widget_unlock(w); return ret; } - old = ret; new = (old & ~mask) | (value & mask); change = old != new; if (change) { @@ -513,7 +516,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, unsigned int invert = mc->invert; if (reg != SND_SOC_NOPM) { - val = soc_widget_read(w, reg); + soc_widget_read(w, reg, &val); val = (val >> shift) & mask; if (invert) val = max - val; @@ -529,7 +532,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, w->kcontrol_news[i].private_value; int val, item; - val = soc_widget_read(w, e->reg); + soc_widget_read(w, e->reg, &val); item = (val >> e->shift_l) & e->mask; if (item < e->max && !strcmp(p->name, e->texts[item])) @@ -558,7 +561,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, w->kcontrol_news[i].private_value; int val, item; - val = soc_widget_read(w, e->reg); + soc_widget_read(w, e->reg, &val); val = (val >> e->shift_l) & e->mask; for (item = 0; item < e->max; item++) { if (val == e->values[item]) @@ -2782,7 +2785,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) /* Read the initial power state from the device */ if (w->reg >= 0) { - val = soc_widget_read(w, w->reg) >> w->shift; + soc_widget_read(w, w->reg, &val); + val = val >> w->shift; val &= w->mask; if (val == w->on_val) w->power = 1; -- cgit v1.2.3-70-g09d2 From 1104a9c822f0e9f5e57a236f20a142166dd8f91e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 15 Jan 2014 19:04:19 +0000 Subject: ASoC: core: Return -ENOTSUPP from set_sysclk() if no operation provided Make it easier for generic code to work with set_sysclk() by distinguishing between the operation not being supported and an error as is done for other operations like set_dai_fmt() Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 03c779ebd72..0ebf1dac330 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3484,7 +3484,7 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0, freq, dir); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); @@ -3505,7 +3505,7 @@ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, return codec->driver->set_sysclk(codec, clk_id, source, freq, dir); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); -- cgit v1.2.3-70-g09d2