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authorLinus Torvalds <torvalds@linux-foundation.org>2015-03-06 10:55:41 -0800
committerLinus Torvalds <torvalds@linux-foundation.org>2015-03-06 10:55:41 -0800
commit5f237425f352487a2e3fdef2f0557eedcd97d898 (patch)
tree763d96a3ec3f89896663b0f07b85adba14d329f2
parent39ed853a2447ce85cf29b3c0357998ff968beeb5 (diff)
parent4fda87df09bee2b1bf236aba408c3236d4f1fbca (diff)
Merge tag 'sound-4.0-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai: "Here are a few more ASoC changes that have been gathered since rc1, but it's still fairly calm over all. The only largish LOC is found in atmel driver, and it's just a removal of broken non-DT stuff. The rest are all small driver-specific fixes, nothing to worry much" * tag 'sound-4.0-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (26 commits) ALSA: hda - One more Dell macine needs DELL1_MIC_NO_PRESENCE quirk ALSA: opl3: small array underflow ALSA: line6: Clamp values correctly ALSA: msnd: add some missing curly braces ASoC: omap-pcm: Correct dma mask ASoC: simple-card: Add a NULL pointer check in asoc_simple_card_dai_link_of ASoC: sam9g20_wm8731: drop machine_is_xxx ALSA: dice: fix wrong offsets for Dice interface ALSA: oxfw: fix a condition and return code in start_stream() ASoC: OMAP: mcbsp: Fix CLKX and CLKR pinmux when used as inputs ASoC: rt5677: Correct the routing paths of that after IF1/2 DACx Mux ASoC: sta32x: fix register range in regmap. ASoC: rt5670: Set RT5670_IRQ_CTRL1 non volatile ASoC: Intel: reset the DSP while suspending ASoC: Intel: save and restore the CSR register ASoC: Intel: update MMX ID to 3 ASoC: max98357a: Add missing header files ASoC: cirrus: tlv320aic23 needs I2C ASoC: Samsung: add missing I2C/SPI dependencies ASoC: rt5670: Fix the speaker mono output issue ...
-rw-r--r--sound/drivers/opl3/opl3_midi.c2
-rw-r--r--sound/firewire/dice/dice-interface.h18
-rw-r--r--sound/firewire/dice/dice-proc.c4
-rw-r--r--sound/firewire/oxfw/oxfw-stream.c5
-rw-r--r--sound/isa/msnd/msnd_pinnacle_mixer.c3
-rw-r--r--sound/pci/hda/patch_realtek.c7
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c68
-rw-r--r--sound/soc/cirrus/Kconfig2
-rw-r--r--sound/soc/codecs/Kconfig2
-rw-r--r--sound/soc/codecs/max98357a.c12
-rw-r--r--sound/soc/codecs/rt5670.c7
-rw-r--r--sound/soc/codecs/rt5677.c32
-rw-r--r--sound/soc/codecs/sta32x.c6
-rw-r--r--sound/soc/fsl/fsl_ssi.c11
-rw-r--r--sound/soc/generic/simple-card.c5
-rw-r--r--sound/soc/intel/sst-atom-controls.h2
-rw-r--r--sound/soc/intel/sst/sst.c10
-rw-r--r--sound/soc/omap/omap-hdmi-audio.c3
-rw-r--r--sound/soc/omap/omap-mcbsp.c11
-rw-r--r--sound/soc/omap/omap-pcm.c2
-rw-r--r--sound/soc/samsung/Kconfig10
-rw-r--r--sound/soc/sh/rcar/core.c4
-rw-r--r--sound/usb/line6/playback.c6
23 files changed, 140 insertions, 92 deletions
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index f62780ed64a..7821b07415a 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -105,6 +105,8 @@ static void snd_opl3_calc_pitch(unsigned char *fnum, unsigned char *blocknum,
int pitchbend = chan->midi_pitchbend;
int segment;
+ if (pitchbend < -0x2000)
+ pitchbend = -0x2000;
if (pitchbend > 0x1FFF)
pitchbend = 0x1FFF;
diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h
index 27b044f84c8..de7602bd69b 100644
--- a/sound/firewire/dice/dice-interface.h
+++ b/sound/firewire/dice/dice-interface.h
@@ -299,23 +299,23 @@
#define RX_ISOCHRONOUS 0x008
/*
- * Index of first quadlet to be interpreted; read/write. If > 0, that many
- * quadlets at the beginning of each data block will be ignored, and all the
- * audio and MIDI quadlets will follow.
- */
-#define RX_SEQ_START 0x00c
-
-/*
* The number of audio channels; read-only. There will be one quadlet per
* channel.
*/
-#define RX_NUMBER_AUDIO 0x010
+#define RX_NUMBER_AUDIO 0x00c
/*
* The number of MIDI ports, 0-8; read-only. If > 0, there will be one
* additional quadlet in each data block, following the audio quadlets.
*/
-#define RX_NUMBER_MIDI 0x014
+#define RX_NUMBER_MIDI 0x010
+
+/*
+ * Index of first quadlet to be interpreted; read/write. If > 0, that many
+ * quadlets at the beginning of each data block will be ignored, and all the
+ * audio and MIDI quadlets will follow.
+ */
+#define RX_SEQ_START 0x014
/*
* Names of all audio channels; read-only. Quadlets are byte-swapped. Names
diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c
index f5c1d1bced5..ecfe20fd4de 100644
--- a/sound/firewire/dice/dice-proc.c
+++ b/sound/firewire/dice/dice-proc.c
@@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry,
} tx;
struct {
u32 iso;
- u32 seq_start;
u32 number_audio;
u32 number_midi;
+ u32 seq_start;
char names[RX_NAMES_SIZE];
u32 ac3_caps;
u32 ac3_enable;
@@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry,
break;
snd_iprintf(buffer, "rx %u:\n", stream);
snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso);
- snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start);
snd_iprintf(buffer, " audio channels: %u\n",
buf.rx.number_audio);
snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi);
+ snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start);
if (quadlets >= 68) {
dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE);
snd_iprintf(buffer, " names: %s\n", buf.rx.names);
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 29ccb363716..e6757cd8572 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -171,9 +171,10 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream,
}
/* Wait first packet */
- err = amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT);
- if (err < 0)
+ if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) {
stop_stream(oxfw, stream);
+ err = -ETIMEDOUT;
+ }
end:
return err;
}
diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c
index 17e49a071af..b408540798c 100644
--- a/sound/isa/msnd/msnd_pinnacle_mixer.c
+++ b/sound/isa/msnd/msnd_pinnacle_mixer.c
@@ -306,11 +306,12 @@ int snd_msndmix_new(struct snd_card *card)
spin_lock_init(&chip->mixer_lock);
strcpy(card->mixername, "MSND Pinnacle Mixer");
- for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++)
+ for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) {
err = snd_ctl_add(card,
snd_ctl_new1(snd_msnd_controls + idx, chip));
if (err < 0)
return err;
+ }
return 0;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b2b24a8b3da..526398a4a44 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5209,6 +5209,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x17, 0x40000000},
{0x1d, 0x40700001},
{0x21, 0x02211040}),
+ SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC255_STANDARD_PINS,
+ {0x12, 0x90a60170},
+ {0x14, 0x90170140},
+ {0x17, 0x40000000},
+ {0x1d, 0x40700001},
+ {0x21, 0x02211050}),
SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
{0x12, 0x90a60130},
{0x13, 0x40000000},
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index f5ad214663f..8de836165cf 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -46,8 +46,6 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <asm/mach-types.h>
-
#include "../codecs/wm8731.h"
#include "atmel-pcm.h"
#include "atmel_ssc_dai.h"
@@ -171,9 +169,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
int ret;
if (!np) {
- if (!(machine_is_at91sam9g20ek() ||
- machine_is_at91sam9g20ek_2mmc()))
- return -ENODEV;
+ return -ENODEV;
}
ret = atmel_ssc_set_audio(0);
@@ -210,39 +206,37 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
/* Parse device node info */
- if (np) {
- ret = snd_soc_of_parse_card_name(card, "atmel,model");
- if (ret)
- goto err;
-
- ret = snd_soc_of_parse_audio_routing(card,
- "atmel,audio-routing");
- if (ret)
- goto err;
-
- /* Parse codec info */
- at91sam9g20ek_dai.codec_name = NULL;
- codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
- if (!codec_np) {
- dev_err(&pdev->dev, "codec info missing\n");
- return -EINVAL;
- }
- at91sam9g20ek_dai.codec_of_node = codec_np;
-
- /* Parse dai and platform info */
- at91sam9g20ek_dai.cpu_dai_name = NULL;
- at91sam9g20ek_dai.platform_name = NULL;
- cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
- if (!cpu_np) {
- dev_err(&pdev->dev, "dai and pcm info missing\n");
- return -EINVAL;
- }
- at91sam9g20ek_dai.cpu_of_node = cpu_np;
- at91sam9g20ek_dai.platform_of_node = cpu_np;
-
- of_node_put(codec_np);
- of_node_put(cpu_np);
+ ret = snd_soc_of_parse_card_name(card, "atmel,model");
+ if (ret)
+ goto err;
+
+ ret = snd_soc_of_parse_audio_routing(card,
+ "atmel,audio-routing");
+ if (ret)
+ goto err;
+
+ /* Parse codec info */
+ at91sam9g20ek_dai.codec_name = NULL;
+ codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+ if (!codec_np) {
+ dev_err(&pdev->dev, "codec info missing\n");
+ return -EINVAL;
+ }
+ at91sam9g20ek_dai.codec_of_node = codec_np;
+
+ /* Parse dai and platform info */
+ at91sam9g20ek_dai.cpu_dai_name = NULL;
+ at91sam9g20ek_dai.platform_name = NULL;
+ cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+ if (!cpu_np) {
+ dev_err(&pdev->dev, "dai and pcm info missing\n");
+ return -EINVAL;
}
+ at91sam9g20ek_dai.cpu_of_node = cpu_np;
+ at91sam9g20ek_dai.platform_of_node = cpu_np;
+
+ of_node_put(codec_np);
+ of_node_put(cpu_np);
ret = snd_soc_register_card(card);
if (ret) {
diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig
index 7b7fbcd49e5..c7cd60f009e 100644
--- a/sound/soc/cirrus/Kconfig
+++ b/sound/soc/cirrus/Kconfig
@@ -16,7 +16,7 @@ config SND_EP93XX_SOC_AC97
config SND_EP93XX_SOC_SNAPPERCL15
tristate "SoC Audio support for Bluewater Systems Snapper CL15 module"
- depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15
+ depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C
select SND_EP93XX_SOC_I2S
select SND_SOC_TLV320AIC23_I2C
help
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 064e6c18e10..ea9f0e31f9d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX98090 if I2C
select SND_SOC_MAX98095 if I2C
- select SND_SOC_MAX98357A
+ select SND_SOC_MAX98357A if GPIOLIB
select SND_SOC_MAX9850 if I2C
select SND_SOC_MAX9768 if I2C
select SND_SOC_MAX9877 if I2C
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index 1806333ea29..e9e6efbc21d 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -12,9 +12,19 @@
* max98357a.c -- MAX98357A ALSA SoC Codec driver
*/
-#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/err.h>
#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
+#include <linux/kernel.h>
+#include <linux/mod_devicetable.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
#define DRV_NAME "max98357a"
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index e1a4a45c57e..fd102613d20 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -225,7 +225,6 @@ static bool rt5670_volatile_register(struct device *dev, unsigned int reg)
case RT5670_ADC_EQ_CTRL1:
case RT5670_EQ_CTRL1:
case RT5670_ALC_CTRL_1:
- case RT5670_IRQ_CTRL1:
case RT5670_IRQ_CTRL2:
case RT5670_INT_IRQ_ST:
case RT5670_IL_CMD:
@@ -2703,6 +2702,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
regmap_write(rt5670->regmap, RT5670_RESET, 0);
+ regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val);
+ if (val >= 4)
+ regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980);
+ else
+ regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00);
+
ret = regmap_register_patch(rt5670->regmap, init_list,
ARRAY_SIZE(init_list));
if (ret != 0)
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 5d0bb8748dd..fb9c20eace3 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -3284,8 +3284,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IB45 Bypass Mux", "Bypass", "IB45 Mux" },
{ "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" },
- { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" },
- { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" },
+ { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+ { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
{ "IB6 Mux", "SLB DAC 6", "SLB DAC6" },
{ "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" },
{ "IB6 Mux", "IF4 DAC L", "IF4 DAC L" },
@@ -3293,8 +3293,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" },
{ "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" },
- { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" },
- { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" },
+ { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+ { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
{ "IB7 Mux", "SLB DAC 7", "SLB DAC7" },
{ "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" },
{ "IB7 Mux", "IF4 DAC R", "IF4 DAC R" },
@@ -3635,15 +3635,15 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "DAC1 FS", NULL, "DAC1 MIXL" },
{ "DAC1 FS", NULL, "DAC1 MIXR" },
- { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" },
- { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" },
+ { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2 Mux" },
+ { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2 Mux" },
{ "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" },
{ "DAC2 L Mux", "OB 2", "OutBound2" },
- { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" },
- { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" },
+ { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3 Mux" },
+ { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3 Mux" },
{ "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" },
@@ -3651,29 +3651,29 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "DAC2 R Mux", "Haptic Generator", "Haptic Generator" },
{ "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" },
- { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" },
- { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" },
+ { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4 Mux" },
+ { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4 Mux" },
{ "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" },
{ "DAC3 L Mux", "OB 4", "OutBound4" },
- { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" },
- { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" },
+ { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC5 Mux" },
+ { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC5 Mux" },
{ "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" },
{ "DAC3 R Mux", "OB 5", "OutBound5" },
- { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" },
- { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" },
+ { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+ { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
{ "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" },
{ "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" },
{ "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" },
{ "DAC4 L Mux", "OB 6", "OutBound6" },
- { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" },
- { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" },
+ { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+ { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
{ "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" },
{ "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" },
{ "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" },
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 3a1343fa109..007a0e3bc27 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -106,13 +106,11 @@ static const struct reg_default sta32x_regs[] = {
};
static const struct regmap_range sta32x_write_regs_range[] = {
- regmap_reg_range(STA32X_CONFA, STA32X_AUTO2),
- regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2),
+ regmap_reg_range(STA32X_CONFA, STA32X_FDRC2),
};
static const struct regmap_range sta32x_read_regs_range[] = {
- regmap_reg_range(STA32X_CONFA, STA32X_AUTO2),
- regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2),
+ regmap_reg_range(STA32X_CONFA, STA32X_FDRC2),
};
static const struct regmap_range sta32x_volatile_regs_range[] = {
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 2595611e8a6..b9fabbf69db 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -603,10 +603,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
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);
if (baudclk_is_used)
@@ -614,6 +610,13 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
else
clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
+ /*
+ * Hardware limitation: The bclk rate must be
+ * never greater than 1/5 IPG clock rate
+ */
+ if (clkrate * 5 > clk_get_rate(ssi_private->clk))
+ continue;
+
clkrate /= factor;
afreq = clkrate / (i + 1);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index f7c6734bd5d..fb550b5869d 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
strlen(dai_link->cpu_dai_name) +
strlen(dai_link->codec_dai_name) + 2,
GFP_KERNEL);
+ if (!name) {
+ ret = -ENOMEM;
+ goto dai_link_of_err;
+ }
+
sprintf(name, "%s-%s", dai_link->cpu_dai_name,
dai_link->codec_dai_name);
dai_link->name = dai_link->stream_name = name;
diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
index dfebfdd5eb2..daecc58f28a 100644
--- a/sound/soc/intel/sst-atom-controls.h
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -150,7 +150,7 @@ enum sst_cmd_type {
enum sst_task {
SST_TASK_SBA = 1,
- SST_TASK_MMX,
+ SST_TASK_MMX = 3,
};
enum sst_type {
diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c
index 8a8d56a146e..11c578651c1 100644
--- a/sound/soc/intel/sst/sst.c
+++ b/sound/soc/intel/sst/sst.c
@@ -350,7 +350,9 @@ static inline void sst_save_shim64(struct intel_sst_drv *ctx,
spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
- shim_regs->imrx = sst_shim_read64(shim, SST_IMRX),
+ shim_regs->imrx = sst_shim_read64(shim, SST_IMRX);
+ shim_regs->csr = sst_shim_read64(shim, SST_CSR);
+
spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
}
@@ -367,6 +369,7 @@ static inline void sst_restore_shim64(struct intel_sst_drv *ctx,
*/
spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
sst_shim_write64(shim, SST_IMRX, shim_regs->imrx),
+ sst_shim_write64(shim, SST_CSR, shim_regs->csr),
spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
}
@@ -379,6 +382,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx)
* initially active. So change the state to active before
* enabling the pm
*/
+
+ if (!acpi_disabled)
+ pm_runtime_set_active(ctx->dev);
+
pm_runtime_enable(ctx->dev);
if (acpi_disabled)
@@ -409,6 +416,7 @@ static int intel_sst_runtime_suspend(struct device *dev)
synchronize_irq(ctx->irq_num);
flush_workqueue(ctx->post_msg_wq);
+ ctx->ops->reset(ctx);
/* save the shim registers because PMC doesn't save state */
sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64);
diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index ccfb41c22e5..f7eb42aa3f3 100644
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
return ret;
card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return -ENOMEM;
+
card->name = devm_kasprintf(dev, GFP_KERNEL,
"HDMI %s", dev_name(ad->dssdev));
card->owner = THIS_MODULE;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index c7eb9dd67f6..fd99d89de6a 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
case OMAP_MCBSP_SYSCLK_CLKX_EXT:
regs->srgr2 |= CLKSM;
+ regs->pcr0 |= SCLKME;
+ /*
+ * If McBSP is master but yet the CLKX/CLKR pin drives the SRG,
+ * disable output on those pins. This enables to inject the
+ * reference clock through CLKX/CLKR. For this to work
+ * set_dai_sysclk() _needs_ to be called after set_dai_fmt().
+ */
+ regs->pcr0 &= ~CLKXM;
+ break;
case OMAP_MCBSP_SYSCLK_CLKR_EXT:
regs->pcr0 |= SCLKME;
+ /* Disable ouput on CLKR pin in master mode */
+ regs->pcr0 &= ~CLKRM;
break;
default:
err = -ENODEV;
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index f4b05bc23e4..1343ecbf0bd 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -201,7 +201,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
struct snd_pcm *pcm = rtd->pcm;
int ret;
- ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64));
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
if (ret)
return ret;
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 3cebf6ca03d..0632a36852c 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -174,7 +174,7 @@ config SND_SOC_SMDK_WM8994_PCM
config SND_SOC_SPEYSIDE
tristate "Audio support for Wolfson Speyside"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER
select SND_SAMSUNG_I2S
select SND_SOC_WM8996
select SND_SOC_WM9081
@@ -189,7 +189,7 @@ config SND_SOC_TOBERMORY
config SND_SOC_BELLS
tristate "Audio support for Wolfson Bells"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER
select SND_SAMSUNG_I2S
select SND_SOC_WM5102
select SND_SOC_WM5110
@@ -206,7 +206,7 @@ config SND_SOC_LOWLAND
config SND_SOC_LITTLEMILL
tristate "Audio support for Wolfson Littlemill"
- depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C
select SND_SAMSUNG_I2S
select MFD_WM8994
select SND_SOC_WM8994
@@ -223,7 +223,7 @@ config SND_SOC_SNOW
config SND_SOC_ODROIDX2
tristate "Audio support for Odroid-X2 and Odroid-U3"
- depends on SND_SOC_SAMSUNG
+ depends on SND_SOC_SAMSUNG && I2C
select SND_SOC_MAX98090
select SND_SAMSUNG_I2S
help
@@ -231,6 +231,6 @@ config SND_SOC_ODROIDX2
config SND_SOC_ARNDALE_RT5631_ALC5631
tristate "Audio support for RT5631(ALC5631) on Arndale Board"
- depends on SND_SOC_SAMSUNG
+ depends on SND_SOC_SAMSUNG && I2C
select SND_SAMSUNG_I2S
select SND_SOC_RT5631
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 1b53605f715..110577c5231 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1252,6 +1252,8 @@ static int rsnd_probe(struct platform_device *pdev)
goto exit_snd_probe;
}
+ dev_set_drvdata(dev, priv);
+
/*
* asoc register
*/
@@ -1268,8 +1270,6 @@ static int rsnd_probe(struct platform_device *pdev)
goto exit_snd_soc;
}
- dev_set_drvdata(dev, priv);
-
pm_runtime_enable(dev);
dev_info(dev, "probed\n");
diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c
index 05dee690f48..97ed593f601 100644
--- a/sound/usb/line6/playback.c
+++ b/sound/usb/line6/playback.c
@@ -39,7 +39,7 @@ static void change_volume(struct urb *urb_out, int volume[],
for (; p < buf_end; ++p) {
short pv = le16_to_cpu(*p);
int val = (pv * volume[chn & 1]) >> 8;
- pv = clamp(val, 0x7fff, -0x8000);
+ pv = clamp(val, -0x8000, 0x7fff);
*p = cpu_to_le16(pv);
++chn;
}
@@ -54,7 +54,7 @@ static void change_volume(struct urb *urb_out, int volume[],
val = p[0] + (p[1] << 8) + ((signed char)p[2] << 16);
val = (val * volume[chn & 1]) >> 8;
- val = clamp(val, 0x7fffff, -0x800000);
+ val = clamp(val, -0x800000, 0x7fffff);
p[0] = val;
p[1] = val >> 8;
p[2] = val >> 16;
@@ -126,7 +126,7 @@ static void add_monitor_signal(struct urb *urb_out, unsigned char *signal,
short pov = le16_to_cpu(*po);
short piv = le16_to_cpu(*pi);
int val = pov + ((piv * volume) >> 8);
- pov = clamp(val, 0x7fff, -0x8000);
+ pov = clamp(val, -0x8000, 0x7fff);
*po = cpu_to_le16(pov);
}
}