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

[alsa-devel] My understanding of ALSA playback mechanism
by Yu Hang 30 Apr '10

30 Apr '10

Hi, I'm a newbie of ALSA. I've wrote some audio applications under Windows OS with DirectSound and Waveform Audio API. And I also have some knowledge about Sound Architecture of Windows Vista/Windows 7. All my experience about audio architecture was from Windows. I'm currently working on a ALSA plugin for AEC function. What I have to do is to retrieve audio data being played (after mixing) as reference to cancel the echo captured by MIC. According to my previous experience about Sound Architecture of Win7/Vista, the audio data is sending from OS component to my own component: no matter for capture or playback. All I have to do is just simply modify the input data and put them into output buffer. If my component is under OS mixer, the received audio data is after mixing. When I read source code of ALSA lib, I found the mechanism between Windows and ALSA lib are quite different. Following was my understanding about ALSA lib mechanism during playback. There is a hardware buffer. ALSA driver takes data from the buffer and sends it to HW. The driver also update the play position on the fly. dmix receive the data from application and mix them into a shared memory, and then copy it into hardware buffer. Each of dmix instance keep a pointer to indicate current mixing position which might different from other instance. [play pos] [1st mixed data] | | +----------------------------+ hardware buffer +----------------------------+ | [2nd mixed data] The figure shows, the data mixed from 1st dmix instance lasts to [1st mixed data] pos, and data mixed from 2nd dmix instance lasts to [2nd mixed data]. The buffer between [1st mixed data] and [2nd mixed data] contains only data of 2nd dmix instance, but it does not matter, ALSA just guarantees the data of 1st dmix instance can be mixed up before the [play pos] reaches [1st mixed data] pos. Of course the hardware buffer is not unlimited (it's a ring buffer), so before dmix mix data to hardware buffer, it has to check if there is enough space. Is my understanding of ALSA playback mechanism correct?

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Re: [alsa-devel] [RFC] disabling ALSA period interrupts
by Raymond Yau 30 Apr '10

30 Apr '10

2010/4/30 pl bossart <bossart.nospam(a)gmail.com> > > do you mean that it only work on HDaudio and not intel8x0 ? > > > > How about other drivers ? > > It's driver/hardware specific, I guess everyone would have to enable > this in their drivers. I don't understand what 8x0 does...You're > welcome to test and do the same for 8x0 if you have the specs and know > how to disable interrupts. > > Any side effect of this disabling interrupt ? Do you mean that all applications must use pulse api instead of alsa-pulse plugin ? Did your test included those voip application which require low latency ? did your test only play one audio stream to the sound server ? Do those alsa application still work with "hw" device when pulseaudio is disabled ?

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[alsa-devel] [PATCHv2 0/2] ASoC: TWL4030: APLL/AIF ordering and DAPM updates
by Peter Ujfalusi 29 Apr '10

29 Apr '10

Hello, Correct the APLL/AIF powering sequence. Also introducing DAPM outputs and input for keeping APLL/AIF enabled in case when no valid DAPM route selected during audio activity. The second patch removes the legacy OUTL/R. As a note: The APLL/AIF patch replaces the same patch in Liam's series. The OUTL/R removal can be applied after the OUTL/R reference has been removed from machine drivers (as in Lima's series). --- Peter Ujfalusi (2): ASoC: TWL4030: AIF/APLL fix in DAPM domain ASoC: TWL4030: Remove OUTL/R outputs sound/soc/codecs/twl4030.c | 91 ++++++++++++++++++++++++++++++++++---------- 1 files changed, 71 insertions(+), 20 deletions(-)

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[alsa-devel] [PATCH] ASoC: TWL4030: AIF/APLL fix in DAPM domain
by Peter Ujfalusi 28 Apr '10

28 Apr '10

This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. The apll and aif reference counting code has been lifted, and reused from Liam Girdwood's earlier patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Some of the ideas (appl, aif refcounting) has been taken from Liam's patch: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-April/027217.html But I'm reusing the existing DAPM_SUPPLY widgets to handle the APLL and AIF enable. I have tested all possible scenarios, and the codec behaved correctly. The voice path was not verified due to missing HW, but that part has not been changed, so I expect no regressions related to that. sound/soc/codecs/twl4030.c | 91 ++++++++++++++++++++++++++++++++++---------- 1 files changed, 71 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2e025a3..816a995 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -123,7 +123,10 @@ struct twl4030_priv { struct snd_soc_codec codec; unsigned int codec_powered; + + /* reference counts of AIF/APLL users */ unsigned int apll_enabled; + unsigned int aif_enabled; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; @@ -259,22 +262,49 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int status; - - if (enable == twl4030->apll_enabled) - return; + int status = -1; if (enable) + twl4030->apll_enabled++; + else + twl4030->apll_enabled--; + + if (enable && twl4030->apll_enabled == 1) /* Enable PLL */ status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL); - else + else if (!enable && twl4030->apll_enabled == 0) /* Disable PLL */ status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL); if (status >= 0) twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); +} - twl4030->apll_enabled = enable; +static void twl4030_aif_enable(struct snd_soc_codec *codec, int enable) +{ + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); + u8 audio_if; + + /* update the user count */ + if (enable) + twl4030->aif_enabled++; + else + twl4030->aif_enabled--; + + audio_if = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + if (enable && twl4030->aif_enabled == 1) { + /* Enable AIF */ + /* enable the PLL before we use it to clock the DAI */ + twl4030_apll_enable(codec, 1); + + twl4030_write(codec, TWL4030_REG_AUDIO_IF, + audio_if | TWL4030_AIF_EN); + } else if (!enable && twl4030->aif_enabled == 0) { + /* disable the DAI before we stop it's source PLL */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, + audio_if & ~TWL4030_AIF_EN); + twl4030_apll_enable(codec, 0); + } } static void twl4030_power_up(struct snd_soc_codec *codec) @@ -672,6 +702,20 @@ static int apll_event(struct snd_soc_dapm_widget *w, return 0; } +static int aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + twl4030_aif_enable(w->codec, 1); + break; + case SND_SOC_DAPM_POST_PMD: + twl4030_aif_enable(w->codec, 0); + break; + } + return 0; +} + static void headset_ramp(struct snd_soc_codec *codec, int ramp) { struct snd_soc_device *socdev = codec->socdev; @@ -1167,8 +1211,6 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DIGIMIC1"), /* Outputs */ - SND_SOC_DAPM_OUTPUT("OUTL"), - SND_SOC_DAPM_OUTPUT("OUTR"), SND_SOC_DAPM_OUTPUT("EARPIECE"), SND_SOC_DAPM_OUTPUT("PREDRIVEL"), SND_SOC_DAPM_OUTPUT("PREDRIVER"), @@ -1180,6 +1222,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFR"), SND_SOC_DAPM_OUTPUT("VIBRA"), + /* AIF and APLL clocks for running DAIs (including loopback) */ + SND_SOC_DAPM_OUTPUT("AIF DAC"), + SND_SOC_DAPM_INPUT("AIF ADC"), + SND_SOC_DAPM_OUTPUT("APLL"), + /* DACs */ SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback", SND_SOC_NOPM, 0, 0), @@ -1243,7 +1290,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF Enable", SND_SOC_NOPM, 0, 0, aif_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), /* Output MIXER controls */ /* Earpiece */ @@ -1373,10 +1421,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital Voice Playback Mixer", NULL, "DAC Voice"}, /* Supply for the digital part (APLL) */ - {"Digital R1 Playback Mixer", NULL, "APLL Enable"}, - {"Digital L1 Playback Mixer", NULL, "APLL Enable"}, - {"Digital R2 Playback Mixer", NULL, "APLL Enable"}, - {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, @@ -1450,8 +1494,14 @@ static const struct snd_soc_dapm_route intercon[] = { {"Vibra Mux", "AudioR2", "DAC Right2"}, /* outputs */ - {"OUTL", NULL, "Analog L2 Playback Mixer"}, - {"OUTR", NULL, "Analog R2 Playback Mixer"}, + /* Must be always connected (for AIF and APLL) */ + {"AIF DAC", NULL, "Digital L1 Playback Mixer"}, + {"AIF DAC", NULL, "Digital R1 Playback Mixer"}, + {"AIF DAC", NULL, "Digital L2 Playback Mixer"}, + {"AIF DAC", NULL, "Digital R2 Playback Mixer"}, + /* Must be always connected (for APLL) */ + {"APLL", NULL, "Digital Voice Playback Mixer"}, + /* Physical outputs */ {"EARPIECE", NULL, "Earpiece PGA"}, {"PREDRIVEL", NULL, "PredriveL PGA"}, {"PREDRIVER", NULL, "PredriveR PGA"}, @@ -1465,6 +1515,12 @@ static const struct snd_soc_dapm_route intercon[] = { {"VIBRA", NULL, "Vibra Route"}, /* Capture path */ + /* Must be always connected (for AIF and APLL) */ + {"ADC Virtual Left1", NULL, "AIF ADC"}, + {"ADC Virtual Right1", NULL, "AIF ADC"}, + {"ADC Virtual Left2", NULL, "AIF ADC"}, + {"ADC Virtual Right2", NULL, "AIF ADC"}, + /* Physical inputs */ {"Analog Left", "Main Mic Capture Switch", "MAINMIC"}, {"Analog Left", "Headset Mic Capture Switch", "HSMIC"}, {"Analog Left", "AUXL Capture Switch", "AUXL"}, @@ -1497,11 +1553,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, - {"ADC Virtual Left1", NULL, "APLL Enable"}, - {"ADC Virtual Right1", NULL, "APLL Enable"}, - {"ADC Virtual Left2", NULL, "APLL Enable"}, - {"ADC Virtual Right2", NULL, "APLL Enable"}, - {"ADC Virtual Left1", NULL, "AIF Enable"}, {"ADC Virtual Right1", NULL, "AIF Enable"}, {"ADC Virtual Left2", NULL, "AIF Enable"}, -- 1.7.0.4

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[alsa-devel] [PATCH 1/5] ASoC: twl4030 - AIF/APLL clock fix for all DAPM routes.
by Liam Girdwood 28 Apr '10

28 Apr '10

This allows the AIF and APLL clocks to continue running in loopback mode and when invalid DAPM routes are selected at stream runtime. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/codecs/twl4030.c | 123 +++++++++++++++++++++++++++++-------------- 1 files changed, 83 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index dc26b71..1a3e7f6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -123,7 +123,10 @@ struct twl4030_priv { struct snd_soc_codec codec; unsigned int codec_powered; + + /* reference counts of AIF/APLL users */ unsigned int apll_enabled; + unsigned int aif_enabled; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; @@ -259,22 +262,50 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int status; - - if (enable == twl4030->apll_enabled) - return; + int status = -1; + /* update the user count */ if (enable) + twl4030->apll_enabled++; + else + twl4030->apll_enabled--; + + if (enable && twl4030->apll_enabled == 1) /* Enable PLL */ status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL); - else + else if (!enable && twl4030->apll_enabled == 0) /* Disable PLL */ status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL); if (status >= 0) twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); +} - twl4030->apll_enabled = enable; +static void twl4030_aif_enable(struct snd_soc_codec *codec, int enable) +{ + struct twl4030_priv *twl4030 = codec->private_data; + + /* update the user count */ + if (enable) + twl4030->aif_enabled++; + else + twl4030->aif_enabled--; + + if (enable && twl4030->aif_enabled == 1) { + /* Enable AIF */ + /* enable the PLL before we use it to clock the DAI */ + twl4030_apll_enable(codec, 1); + + twl4030_write(codec, TWL4030_REG_AUDIO_IF, + twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF) + | TWL4030_AIF_EN); + } else if (!enable && twl4030->aif_enabled == 0) { + /* disable the DAI before we stop it's source PLL */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, + twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF) + & ~TWL4030_AIF_EN); + twl4030_apll_enable(codec, 0); + } } static void twl4030_power_up(struct snd_soc_codec *codec) @@ -672,6 +703,20 @@ static int apll_event(struct snd_soc_dapm_widget *w, return 0; } +static int aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + twl4030_aif_enable(w->codec, 1); + break; + case SND_SOC_DAPM_POST_PMD: + twl4030_aif_enable(w->codec, 0); + break; + } + return 0; +} + static void headset_ramp(struct snd_soc_codec *codec, int ramp) { struct snd_soc_device *socdev = codec->socdev; @@ -1178,6 +1223,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFR"), SND_SOC_DAPM_OUTPUT("VIBRA"), + /* AIF and APLL clocks for running DAIs (including loopback) */ + SND_SOC_DAPM_OUTPUT("AIF DAC"), + SND_SOC_DAPM_INPUT("AIF ADC"), + SND_SOC_DAPM_OUTPUT("APLL"), + /* DACs */ SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback", SND_SOC_NOPM, 0, 0), @@ -1215,16 +1265,21 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_dbypassv_control), /* Digital mixers, power control for the physical DACs */ - SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer", - TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Digital L1 Playback Mixer", - TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Digital R2 Playback Mixer", - TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Digital L2 Playback Mixer", - TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Digital Voice Playback Mixer", - TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Digital R1 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0, aif_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("Digital L1 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0, aif_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("Digital R2 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0, aif_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("Digital L2 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0, aif_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("Digital Voice Playback Mixer", + TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0, apll_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), /* Analog mixers, power control for the physical PGAs */ SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", @@ -1238,11 +1293,6 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, - SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), - - SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0), - /* Output MIXER controls */ /* Earpiece */ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, @@ -1370,17 +1420,14 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital R2 Playback Mixer", NULL, "DAC Right2"}, {"Digital Voice Playback Mixer", NULL, "DAC Voice"}, - /* Supply for the digital part (APLL) */ - {"Digital R1 Playback Mixer", NULL, "APLL Enable"}, - {"Digital L1 Playback Mixer", NULL, "APLL Enable"}, - {"Digital R2 Playback Mixer", NULL, "APLL Enable"}, - {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, - {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, + /* Route for the digital part (APLL) */ + {"APLL", NULL, "Digital Voice Playback Mixer"}, - {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, - {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, - {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, - {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, + /* Route for digital part (AIF) */ + {"AIF DAC", NULL, "Digital L1 Playback Mixer"}, + {"AIF DAC", NULL, "Digital R1 Playback Mixer"}, + {"AIF DAC", NULL, "Digital L2 Playback Mixer"}, + {"AIF DAC", NULL, "Digital R2 Playback Mixer"}, {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, @@ -1493,15 +1540,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, - {"ADC Virtual Left1", NULL, "APLL Enable"}, - {"ADC Virtual Right1", NULL, "APLL Enable"}, - {"ADC Virtual Left2", NULL, "APLL Enable"}, - {"ADC Virtual Right2", NULL, "APLL Enable"}, - - {"ADC Virtual Left1", NULL, "AIF Enable"}, - {"ADC Virtual Right1", NULL, "AIF Enable"}, - {"ADC Virtual Left2", NULL, "AIF Enable"}, - {"ADC Virtual Right2", NULL, "AIF Enable"}, + /* Route for digital part AIF */ + {"ADC Virtual Left1", NULL, "AIF ADC"}, + {"ADC Virtual Right1", NULL, "AIF ADC"}, + {"ADC Virtual Left2", NULL, "AIF ADC"}, + {"ADC Virtual Right2", NULL, "AIF ADC"}, /* Analog bypass routes */ {"Right1 Analog Loopback", "Switch", "Analog Right"}, -- 1.6.3.3

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[alsa-devel] [GIT PULL] ASoC updates for 2.6.35
by Liam Girdwood 27 Apr '10

27 Apr '10

The following changes since commit f4d593280652025f797d432e90f49d49b9334689: Peter Ujfalusi (1): ASoC: tlv320dac33: Fix for early interrupt in FIFO Mode1 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.35 Jarkko Nikula (4): ASoC: tlv320aic3x: Remove unused version string ASoC: tlv320aic3x: Remove needless power off from aic3x_set_bias_level ASoC: tlv320aic3x: Change bias management semantics ASoC: tlv320aic3x: Add basic regulator support Peter Ujfalusi (4): ASoC: tlv320dac33: Skip calculations in FIFO Bypass mode ASoC: tlv320dac33: Change magic numbers used in Mode7 ASoC: tlv320dac33: Calculate the interface speed during bursts ASoC: tlv320dac33: FIFO caused delay reporting sound/soc/codecs/tlv320aic3x.c | 81 +++++++------- sound/soc/codecs/tlv320dac33.c | 243 ++++++++++++++++++++++++++++++++++++++-- 2 files changed, 272 insertions(+), 52 deletions(-)

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[alsa-devel] [PATCH 1/4] ASoC: tlv320aic3x: Remove unused version string
by Jarkko Nikula 27 Apr '10

27 Apr '10

Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 5be580e..af463d8 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -50,8 +50,6 @@ #include "tlv320aic3x.h" -#define AIC3X_VERSION "0.2" - /* codec private data */ struct aic3x_priv { struct snd_soc_codec codec; -- 1.7.0

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[alsa-devel] [PATCH 0/3] ad193x set_pll, fix register setting and typo
by Barry Song 27 Apr '10

27 Apr '10

Hi Mark/Liam, The following 3 patches add new features or fix bugs in ad193x drivers. 1/3. add set_pll entry to support different clock input 2/3. fix wrong register setting in ad193x_set_dai_fmt 3/3. fix typo, delete redundant space Thanks Barry

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[alsa-devel] ASoC: OMAP/TI upstream route
by Liam Girdwood 27 Apr '10

27 Apr '10

Now that I'm back working full time on ASoC I'm going to manage the upstream branch for the OMAP platform and TI codecs. Mark will continue to manage the upstream for all other codecs, platforms and core. As usual, please CC both Mark and I on new ASoC patches. The OMAP upstream branch tracks Takashi's topic/asoc branch and is here :- git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git #for-2.6.35 Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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[alsa-devel] [PATCH 01/11] ASoC: S3C: I2Sv2: Move defines closer to driver
by Jassi Brar 27 Apr '10

27 Apr '10

The header for I2Sv2 linux/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h contains only controller specific definitions and nothing SoC specific. So, it could be moved to sound/soc/s3c24xx/ Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- .../plat-samsung/include/plat/regs-s3c2412-iis.h | 82 -------------------- sound/soc/s3c24xx/regs-i2s-v2.h | 82 ++++++++++++++++++++ sound/soc/s3c24xx/s3c-i2s-v2.c | 3 +- sound/soc/s3c24xx/s3c2412-i2s.c | 3 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 2 +- 5 files changed, 85 insertions(+), 87 deletions(-) delete mode 100644 arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h create mode 100644 sound/soc/s3c24xx/regs-i2s-v2.h diff --git a/arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h deleted file mode 100644 index abf2fbc..0000000 --- a/arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h +++ /dev/null @@ -1,82 +0,0 @@ -/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h - * - * Copyright 2007 Simtec Electronics <linux(a)simtec.co.uk> - * http://armlinux.simtec.co.uk/ - * - * 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. - * - * S3C2412 IIS register definition -*/ - -#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H -#define __ASM_ARCH_REGS_S3C2412_IIS_H - -#define S3C2412_IISCON (0x00) -#define S3C2412_IISMOD (0x04) -#define S3C2412_IISFIC (0x08) -#define S3C2412_IISPSR (0x0C) -#define S3C2412_IISTXD (0x10) -#define S3C2412_IISRXD (0x14) - -#define S3C2412_IISCON_LRINDEX (1 << 11) -#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) -#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) -#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) -#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) -#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) -#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) -#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) -#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) -#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) -#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) -#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) - -#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) -#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) -#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) -#define S3C64XX_IISMOD_BLC_MASK (3 << 13) - -#define S3C64XX_IISMOD_IMS_PCLK (0 << 10) -#define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) - -#define S3C2412_IISMOD_MASTER_INTERNAL (0 << 10) -#define S3C2412_IISMOD_MASTER_EXTERNAL (1 << 10) -#define S3C2412_IISMOD_SLAVE (2 << 10) -#define S3C2412_IISMOD_MASTER_MASK (3 << 10) -#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) -#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) -#define S3C2412_IISMOD_MODE_TXRX (2 << 8) -#define S3C2412_IISMOD_MODE_MASK (3 << 8) -#define S3C2412_IISMOD_LR_LLOW (0 << 7) -#define S3C2412_IISMOD_LR_RLOW (1 << 7) -#define S3C2412_IISMOD_SDF_IIS (0 << 5) -#define S3C2412_IISMOD_SDF_MSB (1 << 5) -#define S3C2412_IISMOD_SDF_LSB (2 << 5) -#define S3C2412_IISMOD_SDF_MASK (3 << 5) -#define S3C2412_IISMOD_RCLK_256FS (0 << 3) -#define S3C2412_IISMOD_RCLK_512FS (1 << 3) -#define S3C2412_IISMOD_RCLK_384FS (2 << 3) -#define S3C2412_IISMOD_RCLK_768FS (3 << 3) -#define S3C2412_IISMOD_RCLK_MASK (3 << 3) -#define S3C2412_IISMOD_BCLK_32FS (0 << 1) -#define S3C2412_IISMOD_BCLK_48FS (1 << 1) -#define S3C2412_IISMOD_BCLK_16FS (2 << 1) -#define S3C2412_IISMOD_BCLK_24FS (3 << 1) -#define S3C2412_IISMOD_BCLK_MASK (3 << 1) -#define S3C2412_IISMOD_8BIT (1 << 0) - -#define S3C64XX_IISMOD_CDCLKCON (1 << 12) - -#define S3C2412_IISPSR_PSREN (1 << 15) - -#define S3C2412_IISFIC_TXFLUSH (1 << 15) -#define S3C2412_IISFIC_RXFLUSH (1 << 7) -#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) -#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) - - - -#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ - diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h new file mode 100644 index 0000000..abf2fbc --- /dev/null +++ b/sound/soc/s3c24xx/regs-i2s-v2.h @@ -0,0 +1,82 @@ +/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h + * + * Copyright 2007 Simtec Electronics <linux(a)simtec.co.uk> + * http://armlinux.simtec.co.uk/ + * + * 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. + * + * S3C2412 IIS register definition +*/ + +#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H +#define __ASM_ARCH_REGS_S3C2412_IIS_H + +#define S3C2412_IISCON (0x00) +#define S3C2412_IISMOD (0x04) +#define S3C2412_IISFIC (0x08) +#define S3C2412_IISPSR (0x0C) +#define S3C2412_IISTXD (0x10) +#define S3C2412_IISRXD (0x14) + +#define S3C2412_IISCON_LRINDEX (1 << 11) +#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) +#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) +#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) +#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) +#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) +#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) +#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) +#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) +#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) +#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) +#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) + +#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) +#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) +#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) +#define S3C64XX_IISMOD_BLC_MASK (3 << 13) + +#define S3C64XX_IISMOD_IMS_PCLK (0 << 10) +#define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) + +#define S3C2412_IISMOD_MASTER_INTERNAL (0 << 10) +#define S3C2412_IISMOD_MASTER_EXTERNAL (1 << 10) +#define S3C2412_IISMOD_SLAVE (2 << 10) +#define S3C2412_IISMOD_MASTER_MASK (3 << 10) +#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) +#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) +#define S3C2412_IISMOD_MODE_TXRX (2 << 8) +#define S3C2412_IISMOD_MODE_MASK (3 << 8) +#define S3C2412_IISMOD_LR_LLOW (0 << 7) +#define S3C2412_IISMOD_LR_RLOW (1 << 7) +#define S3C2412_IISMOD_SDF_IIS (0 << 5) +#define S3C2412_IISMOD_SDF_MSB (1 << 5) +#define S3C2412_IISMOD_SDF_LSB (2 << 5) +#define S3C2412_IISMOD_SDF_MASK (3 << 5) +#define S3C2412_IISMOD_RCLK_256FS (0 << 3) +#define S3C2412_IISMOD_RCLK_512FS (1 << 3) +#define S3C2412_IISMOD_RCLK_384FS (2 << 3) +#define S3C2412_IISMOD_RCLK_768FS (3 << 3) +#define S3C2412_IISMOD_RCLK_MASK (3 << 3) +#define S3C2412_IISMOD_BCLK_32FS (0 << 1) +#define S3C2412_IISMOD_BCLK_48FS (1 << 1) +#define S3C2412_IISMOD_BCLK_16FS (2 << 1) +#define S3C2412_IISMOD_BCLK_24FS (3 << 1) +#define S3C2412_IISMOD_BCLK_MASK (3 << 1) +#define S3C2412_IISMOD_8BIT (1 << 0) + +#define S3C64XX_IISMOD_CDCLKCON (1 << 12) + +#define S3C2412_IISPSR_PSREN (1 << 15) + +#define S3C2412_IISFIC_TXFLUSH (1 << 15) +#define S3C2412_IISFIC_RXFLUSH (1 << 7) +#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) +#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) + + + +#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ + diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 865f931..cb3652e 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -24,10 +24,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <plat/regs-s3c2412-iis.h> - #include <mach/dma.h> +#include "regs-i2s-v2.h" #include "s3c-i2s-v2.h" #include "s3c-dma.h" diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index f3148f9..7134465 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -32,12 +32,11 @@ #include <sound/soc.h> #include <mach/hardware.h> -#include <plat/regs-s3c2412-iis.h> - #include <mach/regs-gpio.h> #include <mach/dma.h> #include "s3c-dma.h" +#include "regs-i2s-v2.h" #include "s3c2412-i2s.h" #define S3C2412_I2S_DEBUG 0 diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index ab1fa15..b7e951f 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -18,7 +18,6 @@ #include <sound/soc.h> -#include <plat/regs-s3c2412-iis.h> #include <mach/gpio-bank-d.h> #include <mach/gpio-bank-e.h> #include <plat/gpio-cfg.h> @@ -27,6 +26,7 @@ #include <mach/dma.h> #include "s3c-dma.h" +#include "regs-i2s-v2.h" #include "s3c64xx-i2s.h" /* The value should be set to maximum of the total number -- 1.6.2.5

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