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

[alsa-devel] [PATCH] ASoC: Added a missing 32-bit PCM format, to the wm8994 codec.
by Ian Lartey 01 Sep '10

01 Sep '10

Signed-off-by: Ian Lartey <ian(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e03072c..23d9227 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3498,7 +3498,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) #define WM8994_RATES SNDRV_PCM_RATE_8000_96000 #define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: Implement WM8994 DAC data source muxes
by Mark Brown 01 Sep '10

01 Sep '10

Allow selection of the channel used for input to the AIFnDAC signals. This isn't integrated into DAPM since we treat the data as a single mono channel until just beyond this selection so it ends up having no visible effect on the routing. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e03072c..c45ab00 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2072,6 +2072,22 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, return 0; } +static const char *aifdac_src_text[] = { + "Left", "Right" +}; + +static const struct soc_enum aif1dacl_src = + SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 15, 2, aifdac_src_text); + +static const struct soc_enum aif1dacr_src = + SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 14, 2, aifdac_src_text); + +static const struct soc_enum aif2dacl_src = + SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 15, 2, aifdac_src_text); + +static const struct soc_enum aif2dacr_src = + SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aifdac_src_text); + static const struct snd_kcontrol_new wm8994_snd_controls[] = { SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1_ADC1_RIGHT_VOLUME, @@ -2083,6 +2099,11 @@ SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2_ADC_RIGHT_VOLUME, 1, 119, 0, digital_tlv), +SOC_ENUM("AIF1DACL Source", aif1dacl_src), +SOC_ENUM("AIF1DACR Source", aif1dacr_src), +SOC_ENUM("AIF2DACL Source", aif1dacl_src), +SOC_ENUM("AIF2DACR Source", aif1dacr_src), + SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME, -- 1.7.1

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[alsa-devel] problem with bisecting a bug in the alsa driver
by Martin 31 Aug '10

31 Aug '10

hello, im sorry to post in here but i really dont know how to proceed with finding the bug i have. i have an ALC889A codec and since alsa-driver 1.0.22 my microphone is crackling like hell with every recording. it works flawlessly on windows and till 1.0.21 of alsa driver. so i downloaded alsa-driver.git and alsa-kmirror.git and started gitcompile. the error is also present in the most recent git. my next steps were trying to use bisect bad v1.0.22 and bisect good v1.0.21 but im unable to compile the driver with any bisection setting. maybe i run into problems because i have 2 single git repos? i really would appreciate a howto for my problem. i also filed a bug in the bugtracker with more infos included. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5115

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[alsa-devel] [PATCH RESEND v6 0/3] ASoC: Add iPAQ RX1950 support
by Vasily Khoruzhick 31 Aug '10

31 Aug '10

These three patches are necessary to add sound support for iPAQ RX1950 PDA, appropriate mach-rx1950.c patch is missing, as it depends on battery and LEDs patches which are still not merged nor reviewed. First patch improves situation with power consumption for uda1380 and also fixes suspend/resume. Second patch just adds glue driver. Third patch fixes s3c_device_iis name and exports s3c_device_pcm for all s3c24xx-based devices. v2: - make mixer regs dirty after power-down, not after power-up v3: - improve glue driver init code (handle errors) v4: - adapt patches for multi-component v5: - handle return values in rx1950_uda1380 init code; - use gpio_set_value instead of gpio_direction_output in spk_power; - uda1380: improve code alignment in set_bias v6: - use gpio_is_valid to check gpio validity

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[alsa-devel] Questions about alsa driver for i.Mx27 + wm8974: playback time shorter than the original record time
by Jie Jiang 31 Aug '10

31 Aug '10

Hi, I have been working on developing alsa driver for wm8974 codec on i.mx27 platform. The wm8974 codec is connected to i.mx27 via SSI1 port. And the OS kernel is linux-2.6.19.2.By now, the driver modules can be successfully built and installed/loaded into kernel. I tested the driver playback function with a wav file (8KHz, 16-bit, mono) recorded by Audacity on my laptop.The playback of this wave file on my laptop lasts about 51 seconds. But when I play it (via aplay) on the mx27 platform, the playback time lasts about 34 seconds, and the playback speed is quite faster than the normal speed and the speaker output sounds like being compressed. In my driver, wm8974 works in I2S master mode and mx27 ssi1 is configured as I2S slave. The configuration of wm9874 audio interface register is 0x10, which set wm8974 as I2S master, word length is 16bits, normal bclk and frame clock polarity. The configuration of mx27 ssi1 transmission channel is as follows(using ssi api from kernel 2.6.19.2): ssi_tx_shift_direction(ssi, 0); // MSB first shift ssi_tx_clock_polarity(ssi, 0); // clock on RISING edge ssi_tx_frame_sync_active(ssi, 1);// frame sync LOW active ssi_tx_early_frame_sync(ssi, 1);//init frame sync one bit before data ssi_tx_frame_sync_length(ssi, 0); //one word length frame ssi_tx_bit0(ssi, 1); // shifting w.r.t. bit 0 of TXSR ssi_tx_word_length(ssi, 0x7); //16bit WL ssi_tx_frame_rate(ssi, 2); // DC[4:0] non-zero ssi_tx_clock_direction(ssi, 0); // external bit clock ssi_tx_frame_direction(ssi, 0); // external frame sync ssi_tx_clock_divide_by_two(ssi, 0); ssi_tx_clock_prescaler(ssi, 0); ssi_tx_fifo_empty_watermark(ssi, 0, TX_WATERMARK);//TX_WATERMARK=4 ssi_tx_fifo_enable(ssi, 0, 1); // tx fifo0 enable ssi_transmit_enable(ssi, 1); // transmit section enable ssi_interrupt_enable(ssi, ssi_tx_dma_interrupt_enable); // enable tx dma ssi_interrupt_enable(ssi, ssi_tx_fifo_0_empty); ssi_interrupt_enable(ssi, ssi_tx_fifo_1_empty); ssi_i2s_mode(ssi, 2); //i2s slave Here I'm not very sure about the setting of frame rate and frame sync length. Is the above setting right? But even I set frame rate to 1, it does not make any difference. What's the possible reason causing the incorrect faster playback speed? Any suggestion is appreciated. Regards, Jie Jiang

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[alsa-devel] [patch 1/1] ASoC: OpenRD Client : Fix naming breakage due to multicomponent support
by Arnaud Patard 31 Aug '10

31 Aug '10

multicomponent support added/changed some device name but added some typos, breaking existing OpenRD Client support. Signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org> Index: sound-2.6/arch/arm/mach-kirkwood/common.c =================================================================== --- sound-2.6.orig/arch/arm/mach-kirkwood/common.c 2010-08-23 10:13:13.000000000 +0200 +++ sound-2.6/arch/arm/mach-kirkwood/common.c 2010-08-23 10:13:14.000000000 +0200 @@ -905,7 +905,7 @@ }; static struct platform_device kirkwood_pcm_device = { - .name = "kirkwood-pcm", + .name = "kirkwood-pcm-audio", .id = -1, }; Index: sound-2.6/sound/soc/kirkwood/kirkwood-openrd.c =================================================================== --- sound-2.6.orig/sound/soc/kirkwood/kirkwood-openrd.c 2010-08-23 10:11:33.000000000 +0200 +++ sound-2.6/sound/soc/kirkwood/kirkwood-openrd.c 2010-08-23 10:13:14.000000000 +0200 @@ -66,7 +66,7 @@ .stream_name = "CS42L51 HiFi", .cpu_dai_name = "kirkwood-i2s", .platform_name = "kirkwood-pcm-audio", - .codec_dai_name = "cs42l51_hifi", + .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", .ops = &openrd_client_ops, }, Index: sound-2.6/sound/soc/codecs/cs42l51.c =================================================================== --- sound-2.6.orig/sound/soc/codecs/cs42l51.c 2010-08-30 15:56:38.000000000 +0200 +++ sound-2.6/sound/soc/codecs/cs42l51.c 2010-08-30 15:56:53.000000000 +0200 @@ -622,7 +622,7 @@ static struct i2c_driver cs42l51_i2c_driver = { .driver = { - .name = "cs42L51-codec", + .name = "cs42l51-codec", .owner = THIS_MODULE, }, .id_table = cs42l51_id,

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[alsa-devel] [PATCH 0/2] ASoC: fsi: modify bug fix
by Kuninori Morimoto 31 Aug '10

31 Aug '10

Dear Mark, Liam These are bug fix for FSI ASoC: fsi: modify compile error ASoC: fsi-ak4642/fsi-da7210: modify dai link settings for card detect 2nd patch is quick hack for now. I will modify it more wisely in future. but no idea now Best regards -- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: Swap bias level enumeration
by Jarkko Nikula 31 Aug '10

31 Aug '10

Swapping the bias level enumeration is only meant to help debugging. It is easier if number 0 means bias off and bigger number means bigger bias level. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- include/sound/soc.h | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d31e8b7..493b3a4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -214,10 +214,10 @@ * @OFF: Power Off. No restrictions on transition times. */ enum snd_soc_bias_level { - SND_SOC_BIAS_ON, - SND_SOC_BIAS_PREPARE, - SND_SOC_BIAS_STANDBY, SND_SOC_BIAS_OFF, + SND_SOC_BIAS_STANDBY, + SND_SOC_BIAS_PREPARE, + SND_SOC_BIAS_ON, }; struct snd_jack; -- 1.7.1

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[alsa-devel] Re-2: [PATCH 2/6] ASoC: ad1980: fix names of a few kcontrols
by marcus.weigelt＠fwd-online.de 31 Aug '10

31 Aug '10

Subject: Re: [alsa-devel] [PATCH 2/6] ASoC: ad1980: fix names of a few kcontrols (27-Aug-2010 20:32) From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> To: marcus.weigelt(a)fwd-online.de > On Tue, Aug 24, 2010 at 03:09:40PM +0200, Uwe Kleine-K??nig wrote: > > From: Marcus Weigelt <marcus.weigelt(a)fwd-online.de> > > > > The names used before keep alsamixer from showing the appropriate > > controls for the AD1980. Mic and phone volume controls were shown in > > alsamixers' capture tab where they don??t belong to. Phone playback > > It's not massively obvious that microphone and phone aren't inputs and > therefore don't sensibly appear in the capture tab? The volume controls for mic and phone are actually hw mixing volumes that do not affect the recording path, therefore only the record selector belongs to capture tab, but not the volume control. > > > volume did not appear in playback tab and the microphone playback could > > never be unmuted. Further it was not possible to select phone as > > capture source. Changing the names fixes these problems. > > This doesn't sound real - misnamed controls might cause UI problems but > you'd have to try *really* hard to make the control name completely > prevent the setting of the control. May sound unreal, but if you look at the specific changes I´ve done, you will notice that I just added "Playback" to the mic volume and mic switch and changed "Phone Capture" to "Phone Playback". From the ALSA amixer.c I know, that there is some string compare on "playback" and "capture", don´t know if it is the same with the alsamixer, but this would explain the different behaviour after changing the names. I can see the result here - the controls are in the right place now and they are doing what they were supposed to. > > What is the actual problem that is is being fixed here? - microphone playback could not be unmuted before, that means that the hw mixing path from mic in to spk out/line out could never work - fixed by this - microphone volume was shown twice in "All" tab of Alsamixer, but running synchroniously, if you change one, the other will follow with the same value - fixed by this - if microphone voulme was changed in capture tab, the volume also changed in playback tab - fixed by this - microphone was never shown as active capture source, even if it was selected correctly, neither in capture tab nor in all-tab - fixed by this - phone was shown as selected capture source in capture tab, but actually it was not selected, instead it was muted/unmuted everytime one tries to select it as capture source - fixed by this > > > Signed-off-by: Marcus Weigelt <marcus.weigelt(a)fwd-online.de> > > Signed-off-by: Uwe Kleine-K??nig <u.kleine-koenig(a)pengutronix.de> > > Please remember to CC maintaintainers on patches. > > > --- > > sound/soc/codecs/ad1980.c | 8 ++++---- > > 1 files changed, 4 insertions(+), 4 deletions(-) > > > > diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c > > index 35c0580..c1550b8 100644 > > --- a/sound/soc/codecs/ad1980.c > > +++ b/sound/soc/codecs/ad1980.c > > SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, > > 1), > > SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), > > SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), > > > > -SOC_SINGLE("Phone Capture Volume", AC97_PHONE, 0, 31, 1), > > -SOC_SINGLE("Phone Capture Switch", AC97_PHONE, 15, 1, 1), > > +SOC_SINGLE("Phone Playback Volume", AC97_PHONE, 0, 31, 1), > > +SOC_SINGLE("Phone Playback Switch", AC97_PHONE, 15, 1, 1), > > > > -SOC_SINGLE("Mic Volume", AC97_MIC, 0, 31, 1), > > -SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), > > +SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1), > > +SOC_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1), > > > > SOC_SINGLE("Stereo Mic Switch", AC97_AD_MISC, 6, 1, 0), > > SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0), > > -- > > 1.7.1 > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- > "You grabbed my hand and we fell into it, like a daydream - or a fever." To: broonie(a)opensource.wolfsonmicro.com u.kleine-koenig(a)pengutronix.de Cc: alsa-devel(a)alsa-project.org sonic.zhang(a)analog.com lrg(a)slimlogic.co.uk

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[alsa-devel] [PATCH 0/2] ASoC: Add iPAQ RX1950 sound support
by Vasily Khoruzhick 31 Aug '10

31 Aug '10

These two patches are necessary to add sound support for iPAQ RX1950 PDA, appropriate mach-rx1950.c patch is missing, as it depends on battery and LEDs patches which are still not merged nor reviewed. First patch improves situation with power consumption for uda1380 and also fixes suspend/resume. Second patch just adds glue driver.

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