October 2011 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH 1/1] intel8x0: Improve performance in virtual environment
by Jaroslav Kysela 20 Oct '11

20 Oct '11

Date 20.10.2011 17:02, Konstantin Ozerkov wrote: > This patch intended to improve performance in virtualized environments > like Parallels Desktop or VirtualBox/QEMU (virtual ICH/AC97 audio). > > I/O access is very time-expensive operation in virtual world: VCPU > can be rescheduled and in the worst case we got more than 10ms delay on > each I/O access. > > In the original code normal loop exit rule > (old_civ == current_civ && old_picb == current_picb) was never satisfied, > because old_picb was never the same as current_picb due delay inspired by > reading current_civ. As a result loop ended by timeout and we got 10x more > I/O operations. > > To prevent glitch on buffer swapping the rule (old_civ == current_civ) is > enough. I don't think so. This code was added because we got some strange situations (values) with the real intel8x0 hardware. A better way might be to identify the virtualisation (maybe using special PCI IDs) and skip these extra checks in the driver for these devices. NAK from me. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH] ASoC: rt5631: Remove unused codec field from struct rt5631_priv
by Axel Lin 20 Oct '11

20 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/rt5631.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 86e69f4..27a078c 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -31,7 +31,6 @@ #include "rt5631.h" struct rt5631_priv { - struct snd_soc_codec *codec; int codec_version; int master; int sysclk; @@ -1632,7 +1631,6 @@ static int rt5631_probe(struct snd_soc_codec *codec) } codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; - rt5631->codec = codec; return 0; } -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: ak4671: Use snd_soc_update_bits for read-modify-write
by Axel Lin 20 Oct '11

20 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4671.c | 18 ++++++------------ 1 files changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 41e3d55..de9ff66 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -168,18 +168,15 @@ static int ak4671_out2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - u8 reg; switch (event) { case SND_SOC_DAPM_POST_PMU: - reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); - reg |= AK4671_MUTEN; - snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT, + AK4671_MUTEN, AK4671_MUTEN); break; case SND_SOC_DAPM_PRE_PMD: - reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); - reg &= ~AK4671_MUTEN; - snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT, + AK4671_MUTEN, 0); break; } @@ -575,15 +572,12 @@ static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int ak4671_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u8 reg; - switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT); - snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, - reg | AK4671_PMVCM); + snd_soc_update_bits(codec, AK4671_AD_DA_POWER_MANAGEMENT, + AK4671_PMVCM, AK4671_PMVCM); break; case SND_SOC_BIAS_OFF: snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Fix reg_word_size for ak4104
by Axel Lin 20 Oct '11

20 Oct '11

According to the register map in datasheet, the registers are 8 bit. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4104.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index cbf0b6d..d3b29dc 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -247,7 +247,7 @@ static struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, .reg_cache_size = AK4104_NUM_REGS, - .reg_word_size = sizeof(u16), + .reg_word_size = sizeof(u8), }; static int ak4104_spi_probe(struct spi_device *spi) -- 1.7.5.4

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[alsa-devel] [PATCH v3 2/8] ASoC: da7210: Add support for DAPM
by Ashish Chavan 20 Oct '11

20 Oct '11

This patch adds support for DAPM covering all inputs and outputs as well as ADC and DAC. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - used supply widget for mic bias - removed mic bias hookup - assigned dapm widgets and routes in card struct - removed explicit setting of default mixer settings - removed explicit setting of ios to standby mode --- sound/soc/codecs/da7210.c | 210 +++++++++++++++++++++++++++++++++++++++----- 1 files changed, 186 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 0809f32..b6e5085 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -28,6 +28,8 @@ /* DA7210 register space */ #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 +#define DA7210_STARTUP2 0x04 +#define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 #define DA7210_INMIX_L 0x0D @@ -40,6 +42,7 @@ #define DA7210_DAC_SEL 0x17 #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 @@ -165,6 +168,185 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, 0x3F, 0, hp_out_tlv), }; +/* + * DAPM Controls + */ +/* In Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), + + SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixl_controls[0], + ARRAY_SIZE(da7210_dapm_inmixl_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixr_controls[0], + ARRAY_SIZE(da7210_dapm_inmixr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixl_controls[0], + ARRAY_SIZE(da7210_dapm_outmixl_controls)), + + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixr_controls[0], + ARRAY_SIZE(da7210_dapm_outmixr_controls)), + + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &da7210_dapm_monomix_controls[0], + ARRAY_SIZE(da7210_dapm_monomix_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT1R"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("OUT2"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da7210_audio_map[] = { + /* Dest Connecting Widget source */ + /* Input path */ + {"Mic Left", NULL, "MICL"}, + {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + + {"INPGA Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "INPGA Left"}, + + {"INPGA Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "INPGA Right"}, + + /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, + {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, + {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, + + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, + {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + + {"Out1 Left", NULL, "OUTPGA Left Enable"}, + {"OUT1L", NULL, "Out1 Left"}, + + {"Out1 Right", NULL, "OUTPGA Right Enable"}, + {"OUT1R", NULL, "Out1 Right"}, + + {"Headphone Left", NULL, "OUTPGA Left Enable"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "OUTPGA Right Enable"}, + {"HPR", NULL, "Headphone Right"}, + + {"Out2 Mono", NULL, "Mono Mixer"}, + {"OUT2", NULL, "Out2 Mono"}, +}; + /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; @@ -204,29 +386,6 @@ static int da7210_volatile_register(struct snd_soc_codec *codec, return 0; } } -static int da7210_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct snd_soc_codec *codec = dai->codec; - - if (is_play) { - /* Enable Out */ - snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); - snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); - - } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); - } - - return 0; -} /* * Set PCM DAI word length. @@ -409,7 +568,6 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { - .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, }; @@ -544,6 +702,10 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, .volatile_register = da7210_volatile_register, + .dapm_widgets = da7210_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7210_dapm_widgets), + .dapm_routes = da7210_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.1

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[alsa-devel] ASoC driver parts probing order (MPC5200/MPC5121)
by David Jander 20 Oct '11

20 Oct '11

Hi all, I am writing a AC97 ASoC driver for the MPC5121e SoC from Freescale. This SoC has almost the same PSC (Programmable Serial Controllers) as the MPC5200B, for which there already is an AC97 driver: sound/soc/fsl/mpc5200-ac97.c, so I'd like to extend that one to also support the MPC5121e. This driver is supposed to work together with a second part, implementing the actual PCM driver: sound/soc/fsl/mpc5200_dma.c. Both drivers register themselves via platform_driver_register() in a regular module init function. The problem is, they do share the same driver data struct! So psc_ac97_of_probe() starts off calling psc_dma = dev_get_drvdata(&op->dev); right ahead and accessing it without checking. The DMA part of the driver does indeed create this struct psc_dma and calls dev_set_drvdata() on it at the end of the probe function. So obviously, it is supposed that the DMA driver somehow gets probed before the PSC driver, but I can't see where this is enforced. AFAIK, the order is fairly random, so it could be the other way around.... and indeed, in my case it is. Unfortunately I don't have MPC5200 hardware with AC97 to test this out, but only a MPC5121e based board. If I mimic this on the MPC5121e, it won't work, because the PSC driver gets probed first and obviously crashes with a NULL-pointer dereference in the next line of code! If this is supposed to be broken, I'd like to fix it of course, but: 1.- I can't test it on a MPC5200B, so therefor I need help. 2.- Simply turning the dev_get/set_drvdata and chip initialization order around does work in my case, but doesn't seem a robust solution to me. Any idea how this should be handled? Best regards, -- David Jander Protonic Holland.

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Re: [alsa-devel] max channels of 3 jacks 6 channels motherboard using model=auto realtek codec
by Takashi Iwai 20 Oct '11

20 Oct '11

At Sun, 10 Jul 2011 08:47:46 +0200, Takashi Iwai wrote: > > At Sun, 10 Jul 2011 11:31:44 +0800, > Raymond Yau wrote: > > > > 2011/7/6 Takashi Iwai <tiwai(a)suse.de>: > > > At Wed, 6 Jul 2011 11:53:47 +0800, > > > Raymond Yau wrote: > > >> > > >> Refer to commit ce764ab22e40a046065c9417ee5f04ad2a816ac1 > > >> > > >> ALSA: hda - Add channel-mode support to Realtek auto-parser > > >> > > >> It seem that max channel is six instead of two when the channel mode > > >> is 2ch for those 3 jacks motherboard > > >> > > >> using hda-emu and alsa-info from > > >> http://thread.gmane.org/gmane.linux.alsa.devel/86816 > > >> > > >> > PCM 0 p 44100 2 16 > > >> Open PCM ALC887 Analog for play > > >> Available PCM parameters: > > >> channels: 2/6 > > >> formats: S16_LE S32_LE > > >> rates: 44100 48000 96000 192000 > > >> Prepare PCM, rate=44100, channels=2, format=16 bits > > >> > > >> > > >> because > > >> > > >> spec->multiout.max_channels = spec->multiout.num_dacs * 2; > > >> > > >> is assigned after alc_auto_add_multi_channel_mode() increase > > >> spec->multiout.num_dacs from 1 to 3 > > >> > > >> spec->multiout.num_dacs = num_pins + 1; > > > > > > Right, and this is intentional. Since there should be enough amount > > > of DACs, the driver keeps using the surround DACs statically. > > > In that way, apps can keep opening the device in the surround mode > > > while user can switch the outputs on the fly via channel-mode mixer > > > enum. > > > > > > > > > > How can the driver support 6 channels when the input jacks have not > > been retasked as output ? > > Even if the pins aren't retasked, DACs can receive data. And then > you can change the pins and routing on the fly. > > You can open a two-channel PCM stream even if the master or (any > other) volume is muted. A similar reason. > > > This behaviour seem different from model=3stack-6ch or > > model=3stack-6ch-dig or those via codec with smart51 "OFF" The smart51 case is a bit different (we prefer DAC unused because of independent HP control), but the 3stack-6ch case is a regression, indeed. This was just because of dropped code snippet. Now I fixed with the patch below. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Re-add need_dac_fix check for multi-io jacks of Realtek codecs During the rewrite, the check of spec->need_dac_fix and the corresponding num_dacs change was dropped from the channel-mode control. This patch re-adds it, and also enables need_dac_fix for ALC880 as default, as this feature was originally introduced to fix h/w bugs of this chip. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 124c63f..52ce075 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3196,6 +3196,8 @@ static int alc_auto_ch_mode_put(struct snd_kcontrol *kcontrol, for (i = 0; i < spec->multi_ios; i++) alc_set_multi_io(codec, i, i < ch); spec->multiout.max_channels = spec->ext_channel_count; + if (spec->need_dac_fix && !spec->const_channel_count) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; return 1; } @@ -3642,6 +3644,7 @@ static int patch_alc880(struct hda_codec *codec) codec->spec = spec; spec->mixer_nid = 0x0b; + spec->need_dac_fix = 1; board_config = alc_board_config(codec, ALC880_MODEL_LAST, alc880_models, alc880_cfg_tbl); -- 1.7.6

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[alsa-devel] [PATCH 1/2] ASoC: wm9081: Fix setting soft VMID ramp enable with VMID 2*240k
by Axel Lin 20 Oct '11

20 Oct '11

According to the datasheet: BIT 2:1 VMID_SEL[1:0] VMID Divider Enable and Select 00 = VMID disabled 01 = 2x40k Omh divider 10 = 2x240k Omh divider 11 = 2x5k Omh divider To set VMID 2*240k, we should OR reg with 0x04 instead of 0x40. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm9081.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index b2d3448..66092ef 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -819,7 +819,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*240k */ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x40; + reg |= 0x04; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); /* Standby bias current on */ -- 1.7.4.1

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Re: [alsa-devel] [PATCH] ALSA: firewire: Neaten cmp_error function
by Clemens Ladisch 20 Oct '11

20 Oct '11

Joe Perches wrote: > Use an actual struct va_format instead of casting > an array of pointers. > - dev_err(&c->resources.unit->device, "%cPCR%u: %pV", > - 'i', c->pcr_index, &(struct va_format){ fmt, &va }); There is no cast. This compound literal _is_ an actual struct va_format. > Move a constant char argument to the format string. Good catch, but this module is going to be extended to handle oPCRs, too. Regards, Clemens

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[alsa-devel] Writing a DMA-less PCM audio driver
by David Jander 20 Oct '11

20 Oct '11

Hi all, I am writing a PCM audio driver for a piece of (embedded) hardware that has no DMA, only a FIFO capable of holding 2048 samples. I was trying to use a kthread with high realtime (SCHED_FIFO) priority to keep the FIFO filled, and sleep with schedule_timeout(x), where "x" is depending on the current FIFO fill level. The same thread also calls snd_pcm_period_elapsed() on every completed period. It seems to work, but I get sporadic audio skips and I am trying to figure out where they come from. Most of the time schedule_timeout(1) takes almost exactly 1ms (1 jiffy at HZ=1000) to complete, but sometimes it takes up to 30ms. It seems related to file-IO (via NFS) happening on the system, but the effect is far bigger if the program that is playing the audio (mplayer) itself is producing file-IO. When using a large cache parameter on mplayer (effectively preloading the entire MP3 file into RAM), the problem is almost gone. Before deciding whether I should debug the network driver to see if it could produce such tremendous amounts of latency, is there a better way to solve this problem? Are there any other DMA-less audio drivers I could look at as an example? Any suggestion is welcome. Best regards, -- David Jander Protonic Holland.

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