January 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] overrun in second capture device after stopping first capture device
by Tobias Schneider 29 Jan '10

29 Jan '10

Dear ALSA Dev-Team, while testing my driver, I figured out that under some circumstances there occur overruns when recording two streams: first I startet a long capture at device 1: -bash-3.2# -bash-3.2# arecord -D hw:0,1 -d 180 -r 32000 -c 1 -t wav -f S16_LE arecord_049.wav Recording WAVE 'arecord_049.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono overrun!!! (at least 121.536 ms long) overrun!!! (at least 115.361 ms long) overrun!!! (at least 190.883 ms long) overrun!!! (at least 121.772 ms long) overrun!!! (at least 119.487 ms long) In another terminal, I started a second capture on the second device. Everytime this capture is done, I got an overrun in the first capture (see above) -bash-3.2# -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# It seems, that the duration of the overrun relates to the capture time of the second capture, as it increases with bigger capture times. But if capture times are bigger than 25s, it seems there is no overrun...just some logs to clarify: First Terminal: -bash-3.2# -bash-3.2# arecord -D hw:0,1 -d 180 -r 32000 -c 1 -t wav -f S16_LE arecord_049.wav Recording WAVE 'arecord_049.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono overrun!!! (at least 99.074 ms long) Second Terminal: -bash-3.2# -bash-3.2# arecord -D hw:0,0 -d 30 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 30 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 30 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# First Terminal: -bash-3.2# arecord -D hw:0,1 -d 180 -r 32000 -c 1 -t wav -f S16_LE arecord_049.wav Recording WAVE 'arecord_049.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono overrun!!! (at least 193.656 ms long) Second Terminal: -bash-3.2# arecord -D hw:0,0 -d 25 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 20 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# So I am wondering why such an overrun occurs, can someone please give me some hints what my be the reason? Thanks in Advance! Tobias

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 29 Jan '10

29 Jan '10

The following changes since commit 64a028a6de08545a2c94f302bc7694bf48aee5b5: Linus Torvalds (1): Merge branch 'for-linus' of git://git.kernel.org/.../viro/vfs-2.6 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Pull request against Linus' current master branch since fix/asoc doesn't seem to have been updated - should I be looking at a different branch? Anuj Aggarwal (2): ASoC: AIC23: Fixing writes to non-existing registers in resume function ASoC: AM3517: ASoC driver not getting compiled sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/omap/Makefile | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH 0/2] McBSP: OMAP3: Add sidetone feature
by Ilkka Koskinen 29 Jan '10

29 Jan '10

The first patch implements McBSP sidetone feature on OMAP3 while the second one provides ALSA interface for using it. The patch set is based on the patch Eduardo Valentin sent to alsa-devel mailing list in October 2009. Channel gain control and enabling the sidetone have been moved to ALSA interface as suggested in comments. Eero Nurkkala (1): McBSP: OMAP3: Add sidetone feature Ilkka Koskinen (1): ASoC: OMAP-McBSP: ASoC interface for McBSP sidetone arch/arm/mach-omap2/mcbsp.c | 2 + arch/arm/plat-omap/include/plat/mcbsp.h | 63 +++++ arch/arm/plat-omap/mcbsp.c | 402 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 187 ++++++++++++++ sound/soc/omap/omap-mcbsp.h | 2 + 5 files changed, 655 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH v2] ASoC: clean up wm8974 and wm8978 clock divider handling
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

wm8974 and wm8978 codec drivers control DAC and ADC oversampling rates in their .set_clkdiv() methods, which is wrong, because these are simple boolean switches and not clock dividers. Move these bits to sound controls. Also remove manual configuration of the MCLK divider in wm8978, since it is configured automatically. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- v1 -> v2: 1. Added " Switch" to wm8974 and wm8978 DAC/ADC oversampling controls sound/soc/codecs/wm8974.c | 12 ++++-------- sound/soc/codecs/wm8974.h | 12 +----------- sound/soc/codecs/wm8978.c | 19 ++++--------------- sound/soc/codecs/wm8978.h | 3 --- sound/soc/sh/migor.c | 4 ---- 5 files changed, 9 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 8812751..ee637af 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -170,6 +170,10 @@ SOC_ENUM("Aux Mode", wm8974_auxmode), SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1), + +/* DAC / ADC oversampling */ +SOC_SINGLE("DAC 128x Oversampling Switch", WM8974_DAC, 8, 1, 0), +SOC_SINGLE("ADC 128x Oversampling Switch", WM8974_ADC, 8, 1, 0), }; /* Speaker Output Mixer */ @@ -381,14 +385,6 @@ static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai, reg = snd_soc_read(codec, WM8974_CLOCK) & 0x11f; snd_soc_write(codec, WM8974_CLOCK, reg | div); break; - case WM8974_ADCCLK: - reg = snd_soc_read(codec, WM8974_ADC) & 0x1f7; - snd_soc_write(codec, WM8974_ADC, reg | div); - break; - case WM8974_DACCLK: - reg = snd_soc_read(codec, WM8974_DAC) & 0x1f7; - snd_soc_write(codec, WM8974_DAC, reg | div); - break; case WM8974_BCLKDIV: reg = snd_soc_read(codec, WM8974_CLOCK) & 0x1e3; snd_soc_write(codec, WM8974_CLOCK, reg | div); diff --git a/sound/soc/codecs/wm8974.h b/sound/soc/codecs/wm8974.h index 98de956..896a7f0 100644 --- a/sound/soc/codecs/wm8974.h +++ b/sound/soc/codecs/wm8974.h @@ -57,17 +57,7 @@ /* Clock divider Id's */ #define WM8974_OPCLKDIV 0 #define WM8974_MCLKDIV 1 -#define WM8974_ADCCLK 2 -#define WM8974_DACCLK 3 -#define WM8974_BCLKDIV 4 - -/* DAC clock dividers */ -#define WM8974_DACCLK_F2 (1 << 3) -#define WM8974_DACCLK_F4 (0 << 3) - -/* ADC clock dividers */ -#define WM8974_ADCCLK_F2 (1 << 3) -#define WM8974_ADCCLK_F4 (0 << 3) +#define WM8974_BCLKDIV 2 /* PLL Out dividers */ #define WM8974_OPCLKDIV_1 (0 << 4) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 8dcebaa..ec2624b 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -210,6 +210,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { /* Speaker */ SOC_DOUBLE_R("Speaker Switch", WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), + + /* DAC / ADC oversampling */ + SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ @@ -513,21 +517,6 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, if (wm8978->f_mclk) ret = wm8978_configure_pll(codec); break; - case WM8978_MCLKDIV: - if (div & ~0xe0) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); - break; - case WM8978_ADCCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); - break; - case WM8978_DACCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); - break; case WM8978_BCLKDIV: if (div & ~0x1c) return -EINVAL; diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h index b58f0bf..56ec832 100644 --- a/sound/soc/codecs/wm8978.h +++ b/sound/soc/codecs/wm8978.h @@ -72,9 +72,6 @@ /* Clock divider Id's */ enum wm8978_clk_id { WM8978_OPCLKRATE, - WM8978_MCLKDIV, - WM8978_ADCCLK, - WM8978_DACCLK, WM8978_BCLKDIV, }; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 3ccd9b3..b823a5c 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -59,10 +59,6 @@ static int migor_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); if (ret < 0) return ret; -- 1.6.2.4

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[alsa-devel] AT91 AC97 Support broken in menuconfig.
by javier Martin 29 Jan '10

29 Jan '10

Hello, I was trying to get AC97 support for an AT91SAM9G45 in kernel 2.6.32 and I have noticed that it requires that AVR32 is selected, however the following commit states that driver for AVR32 was adapted for AT91 too: commit d9fbd9a2cd5ac1b286a7d7cdb1a180ce1edaee2f Merge: f25f60b... baea7b9... Author: Linus Torvalds <torvalds(a)linux-foundation.org> Date: Thu Sep 24 17:06:51 2009 -0700 Merge branch 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm * 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm: (103 commits) .... ARM: 5710/1: at91: add AC97 support to at91sam9rl and at91sam9rlek board ARM: 5709/1: at91: add AC97 support to at91sam9g45 series and at91sam9m10g45ek board ARM: 5621/1: at91/dmaengine: integration of at_hdmac driver in at91sam9g45 series ARM: 5620/1: at91/dmaengine: integration of at_hdmac driver in at91sam9rl ... I just wanted to know if this is just an error in Kconfig tree or the driver under "sound/atmel/" does not support at91sam9g54. Thanks. -- Javier Martin Vista Silicon S.L. CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

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[alsa-devel] [PATCH] ASoC: remove bogus SLEEP mode from wm8978 driver
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

Tests showed, that bit 6 of the WM8978_POWER_MANAGEMENT_2 register of wm8978 affects codec clocks. Being useless for suspend / resume, it cannot be used in bias-level control either. Remove this bit handling. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- sound/soc/codecs/wm8978.c | 7 +------ 1 files changed, 1 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index b86ef47..8dcebaa 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -873,9 +873,7 @@ static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); /* Also switch PLL off */ snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, 0); - /* Put to sleep */ - snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x40); - + return 0; } @@ -887,9 +885,6 @@ static int wm8978_resume(struct platform_device *pdev) int i; u16 *cache = codec->reg_cache; - /* Wake up the codec */ - snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); - /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { if (i == WM8978_RESET) -- 1.6.2.4

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[alsa-devel] [PATCH] ASoC: clean up wm8974 and wm8978 clock divider handling
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

wm8974 and wm8978 codec drivers control DAC and ADC oversampling rates in their .set_clkdiv() methods, which is wrong, because these are simple boolean switches and not clock dividers. Move these bits to sound controls. Also remove manual configuration of the MCLK divider in wm8978, since it is configured automatically. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- sound/soc/codecs/wm8974.c | 12 ++++-------- sound/soc/codecs/wm8974.h | 12 +----------- sound/soc/codecs/wm8978.c | 19 ++++--------------- sound/soc/codecs/wm8978.h | 3 --- sound/soc/sh/migor.c | 4 ---- 5 files changed, 9 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 8812751..051d58d 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -170,6 +170,10 @@ SOC_ENUM("Aux Mode", wm8974_auxmode), SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1), + +/* DAC / ADC oversampling */ +SOC_SINGLE("DAC 128x Oversampling", WM8974_DAC, 8, 1, 0), +SOC_SINGLE("ADC 128x Oversampling", WM8974_ADC, 8, 1, 0), }; /* Speaker Output Mixer */ @@ -381,14 +385,6 @@ static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai, reg = snd_soc_read(codec, WM8974_CLOCK) & 0x11f; snd_soc_write(codec, WM8974_CLOCK, reg | div); break; - case WM8974_ADCCLK: - reg = snd_soc_read(codec, WM8974_ADC) & 0x1f7; - snd_soc_write(codec, WM8974_ADC, reg | div); - break; - case WM8974_DACCLK: - reg = snd_soc_read(codec, WM8974_DAC) & 0x1f7; - snd_soc_write(codec, WM8974_DAC, reg | div); - break; case WM8974_BCLKDIV: reg = snd_soc_read(codec, WM8974_CLOCK) & 0x1e3; snd_soc_write(codec, WM8974_CLOCK, reg | div); diff --git a/sound/soc/codecs/wm8974.h b/sound/soc/codecs/wm8974.h index 98de956..896a7f0 100644 --- a/sound/soc/codecs/wm8974.h +++ b/sound/soc/codecs/wm8974.h @@ -57,17 +57,7 @@ /* Clock divider Id's */ #define WM8974_OPCLKDIV 0 #define WM8974_MCLKDIV 1 -#define WM8974_ADCCLK 2 -#define WM8974_DACCLK 3 -#define WM8974_BCLKDIV 4 - -/* DAC clock dividers */ -#define WM8974_DACCLK_F2 (1 << 3) -#define WM8974_DACCLK_F4 (0 << 3) - -/* ADC clock dividers */ -#define WM8974_ADCCLK_F2 (1 << 3) -#define WM8974_ADCCLK_F4 (0 << 3) +#define WM8974_BCLKDIV 2 /* PLL Out dividers */ #define WM8974_OPCLKDIV_1 (0 << 4) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 8dcebaa..6d807b1 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -210,6 +210,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { /* Speaker */ SOC_DOUBLE_R("Speaker Switch", WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), + + /* DAC / ADC oversampling */ + SOC_SINGLE("DAC 128x Oversampling", WM8978_DAC_CONTROL, 8, 1, 0), + SOC_SINGLE("ADC 128x Oversampling", WM8978_ADC_CONTROL, 8, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ @@ -513,21 +517,6 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, if (wm8978->f_mclk) ret = wm8978_configure_pll(codec); break; - case WM8978_MCLKDIV: - if (div & ~0xe0) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); - break; - case WM8978_ADCCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); - break; - case WM8978_DACCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); - break; case WM8978_BCLKDIV: if (div & ~0x1c) return -EINVAL; diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h index b58f0bf..56ec832 100644 --- a/sound/soc/codecs/wm8978.h +++ b/sound/soc/codecs/wm8978.h @@ -72,9 +72,6 @@ /* Clock divider Id's */ enum wm8978_clk_id { WM8978_OPCLKRATE, - WM8978_MCLKDIV, - WM8978_ADCCLK, - WM8978_DACCLK, WM8978_BCLKDIV, }; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 3ccd9b3..b823a5c 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -59,10 +59,6 @@ static int migor_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); if (ret < 0) return ret; -- 1.6.2.4

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[alsa-devel] [PATCH 2b/4 v2] ASoC: add support for the sh7722 Migo-R board
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

Add support for audio on sh7722-based Migo-R boards, using SIU and wm8978 codec, recording via external microphone and playback via headphones are implemented. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- v2 -> v3: 1. adjusted to the new wm8978 clocking scheme 2. added .hw_free() to switch off PLL diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 8072a6d..a86696b 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -55,4 +61,12 @@ config SND_FSI_DA7210 This option enables generic sound support for the FSI - DA7210 unit +config SND_SIU_MIGOR + tristate "SIU sound support on Migo-R" + depends on SH_MIGOR + select SND_SOC_SH4_SIU + select SND_SOC_WM8978 + help + This option enables sound support for the SH7722 Migo-R board + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 1d0ec0a..8a5a192 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -16,7 +16,9 @@ obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-fsi-ak4642-objs := fsi-ak4642.o snd-soc-fsi-da7210-objs := fsi-da7210.o +snd-soc-migor-objs := migor.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o +obj-$(CONFIG_SND_SIU_MIGOR) += snd-soc-migor.o diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c new file mode 100644 index 0000000..3ccd9b3 --- /dev/null +++ b/sound/soc/sh/migor.c @@ -0,0 +1,222 @@ +/* + * ALSA SoC driver for Migo-R + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> + * + * 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. + */ + +#include <linux/device.h> +#include <linux/firmware.h> +#include <linux/module.h> + +#include <asm/clock.h> + +#include <cpu/sh7722.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8978.h" +#include "siu.h" + +/* Default 8000Hz sampling frequency */ +static unsigned long codec_freq = 8000 * 512; + +static unsigned int use_count; + +/* External clock, sourced from the codec at the SIUMCKB pin */ +static unsigned long siumckb_recalc(struct clk *clk) +{ + return codec_freq; +} + +static struct clk_ops siumckb_clk_ops = { + .recalc = siumckb_recalc, +}; + +static struct clk siumckb_clk = { + .name = "siumckb_clk", + .id = -1, + .ops = &siumckb_clk_ops, + .rate = 0, /* initialised at run-time */ +}; + +static int migor_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->dai->codec_dai; + int ret; + unsigned int rate = params_rate(params); + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 13000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(rtd->dai->cpu_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + codec_freq = rate * 512; + /* + * This propagates the parent frequency change to children and + * recalculates the frequency table + */ + clk_set_rate(&siumckb_clk, codec_freq); + dev_dbg(codec_dai->dev, "%s: configure %luHz\n", __func__, codec_freq); + + ret = snd_soc_dai_set_sysclk(rtd->dai->cpu_dai, SIU_CLKB_EXT, + codec_freq / 2, SND_SOC_CLOCK_IN); + + if (!ret) + use_count++; + + return ret; +} + +static int migor_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + if (use_count) { + use_count--; + + if (!use_count) + snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 0, + SND_SOC_CLOCK_IN); + } else { + dev_dbg(codec_dai->dev, "Unbalanced hw_free!\n"); + } + + return 0; +} + +static struct snd_soc_ops migor_dai_ops = { + .hw_params = migor_hw_params, + .hw_free = migor_hw_free, +}; + +static const struct snd_soc_dapm_widget migor_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Onboard Microphone", NULL), + SND_SOC_DAPM_MIC("External Microphone", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Headphone output connected to LHP/RHP, enable OUT4 for VMID */ + { "Headphone", NULL, "OUT4 VMID" }, + { "OUT4 VMID", NULL, "LHP" }, + { "OUT4 VMID", NULL, "RHP" }, + + /* On-board microphone */ + { "RMICN", NULL, "Mic Bias" }, + { "RMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Onboard Microphone" }, + + /* External microphone */ + { "LMICN", NULL, "Mic Bias" }, + { "LMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "External Microphone" }, +}; + +static int migor_dai_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, migor_dapm_widgets, + ARRAY_SIZE(migor_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +/* migor digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link migor_dai = { + .name = "wm8978", + .stream_name = "WM8978", + .cpu_dai = &siu_i2s_dai, + .codec_dai = &wm8978_dai, + .ops = &migor_dai_ops, + .init = migor_dai_init, +}; + +/* migor audio machine driver */ +static struct snd_soc_card snd_soc_migor = { + .name = "Migo-R", + .platform = &siu_platform, + .dai_link = &migor_dai, + .num_links = 1, +}; + +/* migor audio subsystem */ +static struct snd_soc_device migor_snd_devdata = { + .card = &snd_soc_migor, + .codec_dev = &soc_codec_dev_wm8978, +}; + +static struct platform_device *migor_snd_device; + +static int __init migor_init(void) +{ + int ret; + + ret = clk_register(&siumckb_clk); + if (ret < 0) + return ret; + + /* Port number used on this machine: port B */ + migor_snd_device = platform_device_alloc("soc-audio", 1); + if (!migor_snd_device) { + ret = -ENOMEM; + goto epdevalloc; + } + + platform_set_drvdata(migor_snd_device, &migor_snd_devdata); + + migor_snd_devdata.dev = &migor_snd_device->dev; + + ret = platform_device_add(migor_snd_device); + if (ret) + goto epdevadd; + + return 0; + +epdevadd: + platform_device_put(migor_snd_device); +epdevalloc: + clk_unregister(&siumckb_clk); + return ret; +} + +static void __exit migor_exit(void) +{ + clk_unregister(&siumckb_clk); + platform_device_unregister(migor_snd_device); +} + +module_init(migor_init); +module_exit(migor_exit); + +MODULE_AUTHOR("Guennadi Liakhovetski <g.liakhovetski(a)gmx.de>"); +MODULE_DESCRIPTION("ALSA SoC Migor"); +MODULE_LICENSE("GPL v2");

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[alsa-devel] [PATCH 1/1] ASoC: AM3517: ASoC driver not getting compiled
by Anuj Aggarwal 29 Jan '10

29 Jan '10

Commit 761c9d45 (ASoC: Fix build of OMAP sound drivers) changes CONFIG_MACH_OMAP3517EVM -> CONFIG_SND_OMAP_SOC_OMAP3517EVM in the Makefile. Whereas the config option defined in Kconfig is SND_OMAP_SOC_AM3517EVM. Because of this, ASoC driver for AM3517 was not getting compiled. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Makefile | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3db8a6c..19283e5 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o +obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- 1.6.2.4

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[alsa-devel] [PATCH] ASoC: AC97: SMDK-WM9713: Convert notes from cset to sset
by jassisinghbrar＠gmail.com 29 Jan '10

29 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> It's more robust when references are provided in control names rather than numid. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/smdk_wm9713.c | 23 +++++++---------------- 1 files changed, 7 insertions(+), 16 deletions(-) diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 6fa2c9d..24fd39f 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -29,24 +29,15 @@ static struct snd_soc_card smdk; /* Playback (HeadPhone):- - Headphone Playback Switch - On - $ amixer cset numid=4 1 - - Right Headphone Out Mux - Headphone - $ amixer cset numid=92 2 - Left Headphone Out Mux - Headphone - $ amixer cset numid=93 2 - - Right HP Mixer PCM Playback Switch - On - $ amixer cset numid=75 1 - Left HP Mixer PCM Playback Switch - On - $ amixer cset numid=81 1 + $ amixer sset 'Headphone' unmute + $ amixer sset 'Right Headphone Out Mux' 'Headphone' + $ amixer sset 'Left Headphone Out Mux' 'Headphone' + $ amixer sset 'Right HP Mixer PCM' unmute + $ amixer sset 'Left HP Mixer PCM' unmute Capture (LineIn):- - Right Capture Source - Line - $ amixer cset numid=86 2 - Left Capture Source - Line - $ amixer cset numid=87 2 + $ amixer sset 'Right Capture Source' 'Line' + $ amixer sset 'Left Capture Source' 'Line' */ static struct snd_soc_dai_link smdk_dai = { -- 1.6.2.5

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