From 1007da0604b1d2f064bfecece0f131d57237b03f Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 26 May 2011 09:57:33 -0600 Subject: ASoC: Fix dapm_is_shared_kcontrol so everything isn't shared Commit af46800 ("ASoC: Implement mux control sharing") introduced function dapm_is_shared_kcontrol. When this function returns true, the naming of DAPM controls is derived from the kcontrol_new. Otherwise, the name comes from the widget (and possibly a widget's naming prefix). A bug in the implementation of dapm_is_shared_kcontrol made it return 1 in all cases. Hence, that commit caused a change in control naming for all controls instead of just shared controls. Specifically, a control is always considered shared because it is always compared against itself. Solve this by never comparing against the widget containing the control being created. Equally, controls should never be shared between DAPM contexts; when the same codec is instantiated multiple times, the same kcontrol_new will be used. However, the control should no be shared between the multiple instances. I tested that with the Tegra WM8903 driver: * Shared is now mostly 0 as expected, and sometimes 1. * The expected controls are still generated after this change. However, I don't have any systems that have a widget/control naming prefix, so I can't test that aspect. Thanks for Jarkko Nikula for pointing out how to fix this. Reported-by: Liam Girdwood Tested-by: Jarkko Nikula Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 999bb08cdfb..776e6f41830 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -325,6 +325,7 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, } static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *kcontrolw, const struct snd_kcontrol_new *kcontrol_new, struct snd_kcontrol **kcontrol) { @@ -334,6 +335,8 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, *kcontrol = NULL; list_for_each_entry(w, &dapm->card->widgets, list) { + if (w == kcontrolw || w->dapm != kcontrolw->dapm) + continue; for (i = 0; i < w->num_kcontrols; i++) { if (&w->kcontrol_news[i] == kcontrol_new) { if (w->kcontrols) @@ -468,7 +471,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, return -EINVAL; } - shared = dapm_is_shared_kcontrol(dapm, &w->kcontrol_news[0], + shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[0], &kcontrol); if (kcontrol) { wlist = kcontrol->private_data; -- cgit v1.2.3-70-g09d2 From ea02c63d57d7ec099f66ddb2942b4022e865cd5f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 27 May 2011 21:56:16 +0800 Subject: ASoC: Fix wm_hubs input PGA ZC bits Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e55b298c14a..9e370d14ad8 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -215,23 +215,23 @@ static const struct snd_kcontrol_new analogue_snd_controls[] = { SOC_SINGLE_TLV("IN1L Volume", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN1L Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 1), -SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 0), +SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("IN1R Volume", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN1R Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 1), -SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 0), +SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("IN2L Volume", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN2L Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 1), -SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 0), +SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("IN2R Volume", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN2R Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 1), -SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 0), +SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("MIXINL IN2L Volume", WM8993_INPUT_MIXER3, 7, 1, 0, inmix_sw_tlv), -- cgit v1.2.3-70-g09d2 From 840d8e5e964dc51673d0f26e119b27d2898e8417 Mon Sep 17 00:00:00 2001 From: Joachim Eastwood Date: Wed, 1 Jun 2011 23:59:10 +0200 Subject: ASoC: atmel_ssc: Don't try to free ssc if request failed We should only call ssc_free() when ssc_request() succeeds or bad things will happen. Signed-off-by: Joachim Eastwood Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 7fbfa051f6e..eda955b1583 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -848,9 +848,10 @@ int atmel_ssc_set_audio(int ssc_id) if (IS_ERR(ssc)) pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n", PTR_ERR(ssc)); - else + else { ssc_pdev->dev.parent = &(ssc->pdev->dev); - ssc_free(ssc); + ssc_free(ssc); + } ret = platform_device_add(ssc_pdev); if (ret < 0) -- cgit v1.2.3-70-g09d2 From bb3d6bf1919a20c56b3257b4ec09e67a9226cfb2 Mon Sep 17 00:00:00 2001 From: Linus Torvalds Date: Sat, 4 Jun 2011 07:00:50 +0900 Subject: Revert "ASoC: Update cx20442 for TTY API change" This reverts commit ed0bd2333cffc3d856db9beb829543c1dfc00982. Since we reverted the TTY API change, we should revert the ASoC update to it too. Cc: Mark Brown Cc: Liam Girdwood Cc: Greg Kroah-Hartman Signed-off-by: Linus Torvalds --- sound/soc/codecs/cx20442.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index f8c663dcff0..d68ea532cc7 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -262,14 +262,14 @@ static int v253_hangup(struct tty_struct *tty) } /* Line discipline .receive_buf() */ -static unsigned int v253_receive(struct tty_struct *tty, - const unsigned char *cp, char *fp, int count) +static void v253_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) { struct snd_soc_codec *codec = tty->disc_data; struct cx20442_priv *cx20442; if (!codec) - return count; + return; cx20442 = snd_soc_codec_get_drvdata(codec); @@ -281,8 +281,6 @@ static unsigned int v253_receive(struct tty_struct *tty, codec->hw_write = (hw_write_t)tty->ops->write; codec->card->pop_time = 1; } - - return count; } /* Line discipline .write_wakeup() */ -- cgit v1.2.3-70-g09d2 From 1622ee1822e8adb391b55a09e3cd5144bd9fad47 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2011 17:13:57 +0100 Subject: ASoC: Only update SYSCLK_ENA when pausing WM8915 SYSCLK Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8915.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index a0b1a727828..28fbf072b9c 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -1839,7 +1839,7 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, int old; /* Disable SYSCLK while we reconfigure */ - old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1); + old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA; snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, WM8915_SYSCLK_ENA, 0); -- cgit v1.2.3-70-g09d2 From 6ac340623c5d2a945030814d900701439772ff57 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2011 18:20:50 +0100 Subject: ASoC: Add missing break in WM8915 FLL source selection Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8915.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index 28fbf072b9c..e2ab4fac281 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -2038,6 +2038,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source, break; case WM8915_FLL_MCLK2: reg = 1; + break; case WM8915_FLL_DACLRCLK1: reg = 2; break; -- cgit v1.2.3-70-g09d2 From fd137e2bba53b7207cbae6a1312e89ef3ae55624 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 Jun 2011 11:26:15 +0100 Subject: ASoC: Check for NULL register bank in snd_soc_get_cache_val() Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-cache.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 06b7b81a160..c005ceb70c9 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -466,6 +466,9 @@ static bool snd_soc_set_cache_val(void *base, unsigned int idx, static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, unsigned int word_size) { + if (!base) + return -1; + switch (word_size) { case 1: { const u8 *cache = base; -- cgit v1.2.3-70-g09d2 From 8ca695f273709a9d147826716a8dee3e0eb2407f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jun 2011 13:38:35 +0200 Subject: ASoC: AD1836: Fix setting the PCM format Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/ad1836.c | 14 +++++++------- sound/soc/codecs/ad1836.h | 6 ++++++ 2 files changed, 13 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index ab63d52e36e..754c496412b 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -145,22 +145,22 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - word_len = 3; + word_len = AD1836_WORD_LEN_16; break; case SNDRV_PCM_FORMAT_S20_3LE: - word_len = 1; + word_len = AD1836_WORD_LEN_20; break; case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S32_LE: - word_len = 0; + word_len = AD1836_WORD_LEN_24; break; } - snd_soc_update_bits(codec, AD1836_DAC_CTRL1, - AD1836_DAC_WORD_LEN_MASK, word_len); + snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK, + word_len << AD1836_DAC_WORD_LEN_OFFSET); - snd_soc_update_bits(codec, AD1836_ADC_CTRL2, - AD1836_ADC_WORD_LEN_MASK, word_len); + snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK, + word_len << AD1836_ADC_WORD_OFFSET); return 0; } diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 845596717fd..9d6a3f8f8aa 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -25,6 +25,7 @@ #define AD1836_DAC_SERFMT_PCK256 (0x4 << 5) #define AD1836_DAC_SERFMT_PCK128 (0x5 << 5) #define AD1836_DAC_WORD_LEN_MASK 0x18 +#define AD1836_DAC_WORD_LEN_OFFSET 3 #define AD1836_DAC_CTRL2 1 #define AD1836_DACL1_MUTE 0 @@ -51,6 +52,7 @@ #define AD1836_ADCL2_MUTE 2 #define AD1836_ADCR2_MUTE 3 #define AD1836_ADC_WORD_LEN_MASK 0x30 +#define AD1836_ADC_WORD_OFFSET 5 #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) @@ -60,4 +62,8 @@ #define AD1836_NUM_REGS 16 +#define AD1836_WORD_LEN_24 0x0 +#define AD1836_WORD_LEN_20 0x1 +#define AD1836_WORD_LEN_16 0x2 + #endif -- cgit v1.2.3-70-g09d2 From 064d58ee3afb8a865a72d24e069c7258ec38640e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 7 Jun 2011 10:24:46 +0200 Subject: ASoC: Blackfin: bf5xx-ad1836: Fix codec device name Fix the codec_name field of the dai_link to match the actual device name of the codec. Otherwise the card won't be instantiated. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/blackfin/bf5xx-ad1836.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index ea4951cf552..f79d1655e03 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -75,7 +75,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .cpu_dai_name = "bfin-tdm.0", .codec_dai_name = "ad1836-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad1836.0", + .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, }, { @@ -84,7 +84,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .cpu_dai_name = "bfin-tdm.1", .codec_dai_name = "ad1836-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad1836.0", + .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, }, }; -- cgit v1.2.3-70-g09d2 From 0f82bdf572fc6e42147151aa4d52542f7fc6d793 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Jun 2011 23:42:04 +0100 Subject: ASoC: Fix WM8962 headphone volume update for use of advanced caches Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- 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 f90ae427242..5e05eed96c3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1999,12 +1999,12 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTL_PGA_ENA) + if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA) return snd_soc_write(codec, WM8962_HPOUTL_VOLUME, reg_cache[WM8962_HPOUTL_VOLUME]); /* ...otherwise the right. The VU is stereo. */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTR_PGA_ENA) + if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA) return snd_soc_write(codec, WM8962_HPOUTR_VOLUME, reg_cache[WM8962_HPOUTR_VOLUME]); -- cgit v1.2.3-70-g09d2 From 3115ae174620eeab4b16f52c8d0a9a35d2717e3c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jun 2011 18:07:49 +0100 Subject: ASoC: WM8804 does not support sample rates below 32kHz Reported-by: Kieran O'Leary Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8804.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 6785688f880..9a5e67c5a6b 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -680,20 +680,25 @@ static struct snd_soc_dai_ops wm8804_dai_ops = { #define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) +#define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + static struct snd_soc_dai_driver wm8804_dai = { .name = "wm8804-spdif", .playback = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = WM8804_RATES, .formats = WM8804_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = WM8804_RATES, .formats = WM8804_FORMATS, }, .ops = &wm8804_dai_ops, -- cgit v1.2.3-70-g09d2 From 0cd114fff9ace7014c0d3ef8ab385fc5d3cf2d2f Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Wed, 8 Jun 2011 15:02:56 -0500 Subject: ASoC: fsl: fix initialization of DMA buffers The DMA (PCM) driver used by some Freescale PowerPC supports separate DAIs for playback and capture, so DMA buffers should be allocated only for the initialized streams. Instead of checking for the number of active channels, which apparently is not reliable, check to see if the actual stream object exists. Also provide a better name for the DMA interrupt. Signed-off-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 15dac0f20cd..6680c0b4d20 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, * should allocate a DMA buffer only for the streams that are valid. */ - if (dai->driver->playback.channels_min) { + if (pcm->streams[0].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); @@ -320,13 +320,13 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, } } - if (dai->driver->capture.channels_min) { + if (pcm->streams[1].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { - snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); dev_err(card->dev, "can't alloc capture dma buffer\n"); + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); return ret; } } @@ -449,7 +449,8 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) dma_private->ld_buf_phys = ld_buf_phys; dma_private->dma_buf_phys = substream->dma_buffer.addr; - ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private); + ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "fsldma-audio", + dma_private); if (ret) { dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n", dma_private->irq, ret); -- cgit v1.2.3-70-g09d2 From 4b80b8c2eee5282dab57f094fd3893c0c09f750c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 9 Jun 2011 13:22:36 +0200 Subject: ASoC: snd_soc_new_{mixer,mux,pga} make sure to use right DAPM context Currently it is possible that snd_soc_new_{mixer,mux,pga} is called with a DAPM context not matching the widgets context. This can lead to a wrong prefix_len calculation, which will result in undefined behaviour. To avoid this always use the DAPM context from the widget itself. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/soc-dapm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 776e6f41830..32ab7fc4579 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -350,9 +350,9 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, } /* create new dapm mixer control */ -static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_mixer(struct snd_soc_dapm_widget *w) { + struct snd_soc_dapm_context *dapm = w->dapm; int i, ret = 0; size_t name_len, prefix_len; struct snd_soc_dapm_path *path; @@ -450,9 +450,9 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, } /* create new dapm mux control */ -static int dapm_new_mux(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_mux(struct snd_soc_dapm_widget *w) { + struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; struct snd_card *card = dapm->card->snd_card; @@ -535,8 +535,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, } /* create new dapm volume control */ -static int dapm_new_pga(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_pga(struct snd_soc_dapm_widget *w) { if (w->num_kcontrols) dev_err(w->dapm->dev, @@ -1826,13 +1825,13 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: w->power_check = dapm_generic_check_power; - dapm_new_mixer(dapm, w); + dapm_new_mixer(w); break; case snd_soc_dapm_mux: case snd_soc_dapm_virt_mux: case snd_soc_dapm_value_mux: w->power_check = dapm_generic_check_power; - dapm_new_mux(dapm, w); + dapm_new_mux(w); break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: @@ -1845,7 +1844,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: w->power_check = dapm_generic_check_power; - dapm_new_pga(dapm, w); + dapm_new_pga(w); break; case snd_soc_dapm_input: case snd_soc_dapm_output: -- cgit v1.2.3-70-g09d2 From 33195500edf260e8c8809ab9dfc67f50e0ce031f Mon Sep 17 00:00:00 2001 From: Sangbeom Kim Date: Fri, 10 Jun 2011 10:36:54 +0900 Subject: ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register If DMA active status should be checked, I2SCON register should be referenced. In this patch, Fix the incorrect referencing of I2SCON register. Reported-by : Lakkyung Jung Signed-off-by: Sangbeom Kim Acked-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/samsung/i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ffa09b3b2ca..992a732b521 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -191,7 +191,7 @@ static inline bool tx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SMOD); + active = readl(i2s->addr + I2SCON); if (is_secondary(i2s)) active &= CON_TXSDMA_ACTIVE; @@ -223,7 +223,7 @@ static inline bool rx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SMOD) & CON_RXDMA_ACTIVE; + active = readl(i2s->addr + I2SCON) & CON_RXDMA_ACTIVE; return active ? true : false; } -- cgit v1.2.3-70-g09d2 From e9c039052be59753e6bcc7c8b59763899dc1161c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Jun 2011 19:05:58 +0100 Subject: ASoC: Remove unused and about to be broken SND_SOC_CUSTOM I/O bus This will be removed in -next so let's drop it from mainline as soon as we can in order to minimise surprises. Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +-- sound/soc/soc-cache.c | 3 --- 2 files changed, 1 insertion(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index f1de3e0c75b..3a4bd3a3c68 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -248,8 +248,7 @@ typedef int (*hw_write_t)(void *,const char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; enum snd_soc_control_type { - SND_SOC_CUSTOM = 1, - SND_SOC_I2C, + SND_SOC_I2C = 1, SND_SOC_SPI, }; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index c005ceb70c9..039b9532b27 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -409,9 +409,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; switch (control) { - case SND_SOC_CUSTOM: - break; - case SND_SOC_I2C: #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) codec->hw_write = (hw_write_t)i2c_master_send; -- cgit v1.2.3-70-g09d2 From f6d96e0da1ee3cfe67b719570fba3bb2ea057131 Mon Sep 17 00:00:00 2001 From: "Arnaud Patard (Rtp)" Date: Wed, 22 Jun 2011 22:21:48 +0200 Subject: ASoC: imx: Remove unused Kconfig SND_MXC_SOC_SSI entry SND_MXC_SOC_SSI looks to be unused, so kill it. Signed-off-by: Arnaud Patard Acked-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index d8f130d39dd..bb699bb55a5 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -11,9 +11,6 @@ menuconfig SND_IMX_SOC if SND_IMX_SOC -config SND_MXC_SOC_SSI - tristate - config SND_MXC_SOC_FIQ tristate @@ -24,7 +21,6 @@ config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL select SND_SOC_WM8350 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Enable support for audio on the i.MX31ADS with the WM1133-EV1 @@ -34,7 +30,6 @@ config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 select SND_SOC_TVL320AIC32X4 - select SND_MXC_SOC_SSI select SND_MXC_SOC_MX2 help Say Y if you want to add support for SoC audio on Visstrim SM10 @@ -44,7 +39,6 @@ config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" depends on MACH_PCM043 || MACH_PCA100 select SND_SOC_WM9712 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Say Y if you want to add support for SoC audio on Phytec phyCORE @@ -57,7 +51,6 @@ config SND_SOC_EUKREA_TLV320 || MACH_EUKREA_MBIMXSD35_BASEBOARD \ || MACH_EUKREA_MBIMXSD51_BASEBOARD select SND_SOC_TLV320AIC23 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Enable I2S based access to the TLV320AIC23B codec attached -- cgit v1.2.3-70-g09d2 From 96dcabb99b9f63f2b65f2b0bfe5d4eb48f11b177 Mon Sep 17 00:00:00 2001 From: "Arnaud Patard (Rtp)" Date: Wed, 22 Jun 2011 22:21:49 +0200 Subject: ASoC: imx: add missing module informations - add some modules aliases - add module license to avoid tainted kernel when loading the imx-pcm-audio driver Signed-off-by: Arnaud Patard Acked-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-dma-mx2.c | 2 ++ sound/soc/imx/imx-ssi.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index aab7765f401..4173b3d87f9 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -337,3 +337,5 @@ static void __exit snd_imx_pcm_exit(void) platform_driver_unregister(&imx_pcm_driver); } module_exit(snd_imx_pcm_exit); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:imx-pcm-audio"); diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 5b13feca753..61fceb09cdb 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -774,4 +774,4 @@ module_exit(imx_ssi_exit); MODULE_AUTHOR("Sascha Hauer, "); MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface"); MODULE_LICENSE("GPL"); - +MODULE_ALIAS("platform:imx-ssi"); -- cgit v1.2.3-70-g09d2 From 53dea36c70c1857149a8c447224e3936eb8b5339 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 22 Jun 2011 20:48:25 +0200 Subject: ASoC: pxa-ssp: Correct check for stream presence Don't rely on the codec's channels_min information to decide wheter or not allocate a substream's DMA buffer. Rather check if the substream itself was allocated previously. Signed-off-by: Daniel Mack Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/pxa/pxa2xx-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 2ce0b2d891d..fab20a54e86 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -95,14 +95,14 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) -- cgit v1.2.3-70-g09d2 From 16866741bda5d16f3d30d1656ce941faf5dad34c Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Thu, 23 Jun 2011 23:54:40 +0200 Subject: ALSA: Remove unneeded version.h includes from sound/ In the sound/ directory there are two files (flagged by 'make versioncheck'); sound/pci/asihpi/asihpi.c and sound/soc/codecs/wm8991.c that include linux/version.h although they don't need it. This patch removes the unneeded includes. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 1 - sound/soc/codecs/wm8991.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 2ca6f4f85b4..e3569bdd3b6 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -27,7 +27,6 @@ #include "hpioctl.h" #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 3c2ee1bb73c..6af23d06870 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -13,7 +13,6 @@ #include #include -#include #include #include #include -- cgit v1.2.3-70-g09d2