April 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/1] ASoC: TWL4030: Add support Voice DAI
by Joonyoung Shim 20 Apr '09

20 Apr '09

Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz (voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. If the system master clock is not 26MHz, the voice PCM interface is not available. The PCM voice interface has two modes - PCM mode1 : This uses the rising edge of the clock signal - PCM mode2 : This uses the falling edge of the clock signal PCM mode1 and mode2 have a look of DSP_A and DSP_B, so we used DSP_A and DSP_B. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 135 +++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/twl4030.h | 16 +++++- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- 7 files changed, 150 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index db7fecf..13461d1 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1437,6 +1437,107 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_voice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 infreq; + + /* If the system master clock is not 26MHz, the voice PCM interface is + * not avilable. + */ + infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL) + & TWL4030_APLL_INFREQ; + + if (infreq != TWL4030_APLL_INFREQ_26000KHZ) + return -EPERM; + + return 0; +} + +static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 old_mode, mode; + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; + mode = old_mode; + + switch (params_rate(params)) { + case 8000: + mode &= ~(TWL4030_SEL_16K); + break; + case 16000: + mode |= TWL4030_SEL_16K; + break; + default: + printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + +static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: + format &= ~(TWL4030_VIF_SLAVE_EN); + break; + case SND_SOC_DAIFMT_CBS_CFS: + format |= TWL4030_VIF_SLAVE_EN; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + format &= ~(TWL4030_VIF_FORMAT); + break; + case SND_SOC_DAIFMT_DSP_B: + format |= TWL4030_VIF_FORMAT; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + /* change format and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) @@ -1448,8 +1549,14 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .set_fmt = twl4030_set_dai_fmt, }; -struct snd_soc_dai twl4030_dai = { - .name = "twl4030", +static struct snd_soc_dai_ops twl4030_dai_voice_ops = { + .startup = twl4030_voice_startup, + .hw_params = twl4030_voice_hw_params, + .set_fmt = twl4030_voice_set_dai_fmt, +}; + +struct snd_soc_dai twl4030_dai[] = { +{ .name = "twl4030", .playback = { .stream_name = "Playback", .channels_min = 2, @@ -1463,6 +1570,22 @@ struct snd_soc_dai twl4030_dai = { .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .ops = &twl4030_dai_ops, +}, +{ .name = "twl4030 Voice", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &twl4030_dai_voice_ops, +}, }; EXPORT_SYMBOL_GPL(twl4030_dai); @@ -1503,8 +1626,8 @@ static int twl4030_init(struct snd_soc_device *socdev) codec->read = twl4030_read_reg_cache; codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; - codec->dai = &twl4030_dai; - codec->num_dai = 1; + codec->dai = twl4030_dai; + codec->num_dai = ARRAY_SIZE(twl4030_dai), codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); @@ -1598,13 +1721,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dai(&twl4030_dai); + return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dai(&twl4030_dai); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_exit(twl4030_exit); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index cb63765..305b8ba 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -171,6 +171,17 @@ #define TWL4030_CLK256FS_EN 0x02 #define TWL4030_AIF_EN 0x01 +/* VOICE_IF (0x0F) Fields */ + +#define TWL4030_VIF_SLAVE_EN 0x80 +#define TWL4030_VIF_DIN_EN 0x40 +#define TWL4030_VIF_DOUT_EN 0x20 +#define TWL4030_VIF_SWAP 0x10 +#define TWL4030_VIF_FORMAT 0x08 +#define TWL4030_VIF_TRI_EN 0x04 +#define TWL4030_VIF_SUB_EN 0x02 +#define TWL4030_VIF_EN 0x01 + /* EAR_CTL (0x21) */ #define TWL4030_EAR_GAIN 0x30 @@ -236,7 +247,10 @@ #define TWL4030_SMOOTH_ANAVOL_EN 0x02 #define TWL4030_DIGMIC_LR_SWAP_EN 0x01 -extern struct snd_soc_dai twl4030_dai; +#define TWL4030_DAI_HIFI 0 +#define TWL4030_DAI_VOICE 1 + +extern struct snd_soc_dai twl4030_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_twl4030; #endif /* End of __TWL4030_AUDIO_H__ */ diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 0c2322d..027e1a4 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap2evm_ops, }; diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index fd24a4a..6aa428e 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link omap3beagle_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3beagle_ops, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index fe282d4..ad219aa 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -228,14 +228,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .name = "PCM1773", .stream_name = "HiFi Out", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_out_ops, .init = omap3pandora_out_init, }, { .name = "TWL4030", .stream_name = "Line/Mic In", .cpu_dai = &omap_mcbsp_dai[1], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_in_ops, .init = omap3pandora_in_init, } diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index a72dc4e..ec4f8fd 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link overo_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &overo_ops, }; diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 10f1c86..1c79741 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -197,7 +197,7 @@ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }; -- 1.5.6.3

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Re: [alsa-devel] [PATCH v2 2/6] ARM: DaVinci: ASoC: Add the platform devices for ASP
by naresh 20 Apr '09

20 Apr '09

Dave, Thanks for the review. I will re-submit the patch after fixing the comments. Regards, Naresh > >On Thursday 16 April 2009, Medisetty, Naresh wrote: >> arch/arm/mach-davinci/dm355.c | 29 ++++++++++++- >> arch/arm/mach-davinci/dm644x.c | 27 +++++++++++- >> arch/arm/mach-davinci/dm646x.c | 67 >++++++++++++++++++++++++++ >> arch/arm/mach-davinci/include/mach/asp.h | 39 ++++++++++++++++- > >Note the merge dependency: most of those aren't yet in mainline. >I think Kevin sent the dm644x.c changes to the ARM list for their >first review cycle recently. > >My comments below are for the dm355.c changes, but they apply just >as much to the other two SoC-specific setup files. > > >> +static struct resource dm355_evm_snd_resources[] = { >> + { >> + .start = DAVINCI_ASP1_BASE, >> + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, >> + .flags = IORESOURCE_MEM, >> + }, >> +}; >> + >> +static struct evm_snd_platform_data dm355_evm_snd_data = { >> + .clk_name = "asp1", >> + .tx_dma_ch = DAVINCI_DMA_ASP1_TX, >> + .rx_dma_ch = DAVINCI_DMA_ASP1_RX, >> +}; > >If you're going to move to proper driver model support >(good!) ... the DMA channels should be IORESOURCE_DMA >resources, and the clock should be associated directly >with that platform device. > >That might seem to leave that current platform data >struct as useless ... not so! It should tell about how >the audio is configured: whether RX and TX are both >wired, if they're stereo or what, how *audio* clocks >are set up, and so forth. All that is specific to the >specific *board* in use, not to that SoC, so it should >be provided by board setup code. A lot of it is now >encoded in the sound/soc/davinci/davinci-evm.c file. > >Example: dm6446 evm supports six different audio clocks, >but right now one choice is fixed at board setup time. > >ASoC seems to have ways to let drivers make that choice. >That could be handled by board-specific callbacks and data >tables, and would be good to package in platform data. >A different dm6446 board may use a different audio clock >setup, with different choices. > > >> + >> +static struct platform_device dm355_asoc_device = { >> + .name = "davinci-asoc", >> + .id = -1, >> + .dev.platform_data = &dm355_evm_snd_data, >> + .num_resources = ARRAY_SIZE(dm355_evm_snd_resources), >> + .resource = dm355_evm_snd_resources, > >Looks like the alignment goofage there (for "=") is more than >what can be explained by patch mangling from email... > > >> +}; > > >> @@ -555,6 +579,7 @@ static int __init dm355_init_devices(void) >> >> davinci_cfg_reg(DM355_INT_EDMA_CC); >> platform_device_register(&dm355_edma_device); >> + platform_device_register(&dm355_asoc_device); > >This does *NOT* belong in the generic dm355 setup code like that. >Not every DM355 board will even have audio! Or have it wired >exactly like Spectrum's EVM boards. > >It'd be appropriate to add dm355_init_asoc(asp_num, platform_data) >or something similar, so that different boards wouldn't need to >repeat the won't-change parts. That routine might need to set up >some pin, IRQ, and EDMA muxing. Boards with no audio support would >not call that routine. Ones with two audio interfaces might need >to call it once for each interface. > > >> return 0; >> } >> postcore_initcall(dm355_init_devices); > > >Similar comments for the two other SoC chips touched by this patch; >although obviously the dm6446 wouldn't need an asp_num, etc. > >- dave

1 0

[alsa-devel] Via 1708S no S/PDIF out
by Gus Zernial 19 Apr '09

19 Apr '09

I posted this on alsa-user and got no response. I have an ASUS M4A78T-E motherboard system, which has 7.1 channel HD Audio out using Via VT1708S chip, (plus ATI HDMI out). I'm running Kubuntu Jaunty 9.04 with a custom 2.6.28.9 kernel. I am getting sound out the Line Out/Speaker Out jacks, but no sound through the Toslink S/PDIF Out jack. The S/PDIF out channel does not show in aplay -L command output, and there is no S/PDIF out slider in amixer. This is a dual-boot system and everything, including S/PDIF out, works fine with WinXP. In linux, I'm running alsa version 1.0.19. I can post more information if needed, but I think I have the right modules loaded, and I've tried various "model=" options for snd_hda_intel, so far none provide S/PDIF out. My questions are .... can someone tell me if S/PDIF out is supposed to work with this audio chip, and if so what do I need to do configuration-wise to get it to work? Thankyou ...

1 0

[alsa-devel] [PATCH] ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR
by Philipp Zabel 19 Apr '09

19 Apr '09

Those macros are just screwed as soon as CONFIG_PXA25x is enabled. This patch - changes ssp_set_scr to take an ssp_dev pointer instead of ssp_device - adds a corresponding ssp_get_scr function. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- To me, using ssp_dev seems to be cleaner, as all the places where ssp_set_scr is called, we already have an ssp_dev *ssp = priv->dev.ssp set up, which allows us to call ssp_set_scr(ssp, ...) instead of ssp_set_scr(&priv->dev, ...). Same for ssp_get_scr. --- sound/soc/pxa/pxa-ssp.c | 36 ++++++++++++++++++++++++++++-------- 1 files changed, 28 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 308a657..d979c5c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -280,12 +280,33 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_dev *dev, u32 div) +static void ssp_set_scr(struct ssp_device *ssp, u32 div) { - struct ssp_device *ssp = dev->ssp; - u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + ssp_write_reg(ssp, SSCR0, sscr0); +} + +/** + * ssp_get_clkdiv - get SSP clock divider + */ +static u32 ssp_get_scr(struct ssp_device *ssp) +{ + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 div; - ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; } /* @@ -326,7 +347,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(&priv->dev, 1); + ssp_set_scr(ssp, 1); sscr0 |= SSCR0_ACS; break; default: @@ -387,7 +408,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(&priv->dev, div); + ssp_set_scr(ssp, div); break; default: return -ENODEV; @@ -674,8 +695,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_I2S: sspsp = ssp_read_reg(ssp, SSPSP); - if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && - (width == 16)) { + if ((ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. -- 1.6.2.3

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[alsa-devel] [PATCH] General fix for Palm27x aSoC driver
by Marek Vasut 19 Apr '09

19 Apr '09

Hi, Firstly, this patch makes the palm27x asoc driver a little more sane. Also, since all affected devices use GPIO95 as AC97_nRESET, this patch sets that properly. Affected are PalmT5, TX and LifeDrive. Please consider applying, it's against Russells tree. If Eric wants to merge it, I can remake it. Thanks.

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[alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Gustavo Vieira 19 Apr '09

19 Apr '09

Hi all, I've got sound problems in my HP Pavilion dv4-1125br. The sound card is a Intel 82801I with the IDT 92HD71B7X codec. Sound works ok, including the internal mic and external mic jack, but I still have a couple of problems. The main problem is that I don't have jack sense in the headphone jack. Headphones do not work and I'm unable even to make them work manually through the mixer. This machine got two headphone jacks in front and none of them works. I've tested it since alsa-driver 1.0.17 and it has never worked. The other problem is that the speaker volume is low. Even if I increase all available mixer controls (PCM and Master), the volume is still tiny. This is a regression. In alsa-driver 1.0.17 (Fedora 10, kernel 2.6.27.21) the volume is ok, the problem appeared sometime after that and alsa-driver 1.0.18a (Fedora 11 beta, kernel 2.6.29.1). One thing I noticed between these versions is the disappearance of the Front mixer control present in 1.0.17. Maybe that is the cause of the problem. I've tried all models available to my codec and none of them solved the issue. Actually, all of them seemed to created other problems. The no model version seems the best right now. I'm attaching the alsa-info output for the Fedora 10 and Fedora 11 beta configurations. I've tried to download and compile the alsa-driver snapshot, but the compilation failed early on with an a patch failure in my kernel 2.6.29.1 system. Abraços, Gustavo

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[alsa-devel] how to find the last plugged in USB headset ?
by reetesh chandran 18 Apr '09

18 Apr '09

Hi all, Is there a way to find the card number associated with the USB headset that was plugged in last. I am listening to udev events and I can find the card number by comparing the VendorId product Id combination. But if there are multiple devices of same VendorId and productId, how do I find the card number associated with the device that was plugged in the last ? Thanks in advance ? Reetesh

1 0

[alsa-devel] SoC pxa2xx-ac97 + wm9705 + touchscreen suspend/resume
by Russell King - ARM Linux 18 Apr '09

18 Apr '09

Trying Linus' latest (as of this morning) out on a PXA255 platform with a WM9705 codec reveals a number of problems: 1. when suspend occurs, we turn the AC97 link off by setting the GCR_ACLINK_OFF bit, and stopping the functional units clock. On resume, we reconfigure the pins and re-enable the clock. Unfortunately, this results in GCR being left with zeros, and GSR remains all zeros, and the codec is inaccessible, with lots of warnings from the read/write functions scrolling by: pxa2xx_ac97_write: write error (ac97_reg=118 GSR=0x0) pxa2xx_ac97_read: read error (ac97_reg=118 GSR=0x0) pxa2xx_ac97_read: read error (ac97_reg=118 GSR=0x0) ... Setting GCR to '2' (to release cold reset) using devmem2 starts things moving again, shutting up this warning. 2. maybe as a result of the above problem, the wm9705 touchscreen driver doesn't reinitialize, causing loss of touchscreen. Unbinding and re-binding the driver restores the touchscreen, but with a very long lag between touching the screen and it being registered. I bring (2) up because I notice that the resume actions in (1) are deferred. Given that codecs have shared functions (such as touchscreens) need to access the codec from their own resume functions, how can this deferral be safe? Liam - this problem has been noticed on a NetbookPro (I'm sure you remember that platform...)

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Re: [alsa-devel] hda Regression: jack doesn't mute speaker
by Takashi Iwai 17 Apr '09

17 Apr '09

At Thu, 16 Apr 2009 09:23:25 +0200, Ortwin Glück wrote: > > Takashi Iwai wrote: > >>>> Also the "Front" > >>>> and "Surround" outputs are now both labeled "Speaker" (see attached images of kmix). > >>> What I/O are they actually corresponding to? Are they speaker outputs > >>> or line-outs, or anything else? > >> They correspond to the built-in speakers. > > > > So they are really front/surround 4.0-speaker outputs? > > Probably. They seem to control two distinct speakers inside the laptop. More I > don't know. In OS-X there is only a single volume control (a sacrifice to > usability I think. who would want to control individual volumes of these on a > laptop anyway). The problem is likely because hp_detect field isn't set on your machine because the headphone shares the same audio-out widget with the speaker out. Could you try the patch below? thanks, Takashi --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ce30b45..917bc5d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3076,6 +3076,11 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, unsigned int wid_caps; for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) { + if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { + wid_caps = get_wcaps(codec, pins[i]); + if (wid_caps & AC_WCAP_UNSOL_CAP) + spec->hp_detect = 1; + } nid = dac_nids[i]; if (!nid) continue; @@ -3119,11 +3124,6 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, err = create_controls_idx(codec, name, idx, nid, 3); if (err < 0) return err; - if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { - wid_caps = get_wcaps(codec, pins[i]); - if (wid_caps & AC_WCAP_UNSOL_CAP) - spec->hp_detect = 1; - } } } return 0;

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[alsa-devel] Cirrus Logic CS4207
by Peter-Jan Deweirdt 17 Apr '09

17 Apr '09

Hi, We are in the progress of developming a new platform and we are using the audio chip CS4207 from Cirrus Logic (very new). I've noticed that the audio is not working as it should. I'm using the moblin menlow kernel and I'm able to see 1 playback channel and 1 capture channel (which is not correct according to the chip). I have retreived from our vendor a patch for the kernel (2.6.27.7) but since the moblin kernel is based on the 2.6.29 I cannot use it, even not verify if it's working. Would it be possible to look to the patch and maybe even integrate it into a new build? Or would it be possible to guide me trough so that the audio is working properly? This is what I'm seeing when using the current kernel from moblin (2.6.29.rc7-40.1) >>> chunk from dmesg [ 5.166641] Advanced Linux Sound Architecture Driver Version 1.0.18a. [ 5.171792] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 5.216898] ALSA device list: [ 5.216916] #0: HDA Intel MID at 0xd0010000 irq 22 [ 5.217024] nf_conntrack version 0.5.0 (3947 buckets, 15788 max) <<<end chunk of dmesg [root@localhost ~]# cat /proc/asound/cards 0 [MID ]: HDA-Intel - HDA Intel MID HDA Intel MID at 0xd0010000 irq 22 [root@localhost ~]# ls -al /dev/snd/ total 0 drwxr-xr-x 2 root root 160 Mar 24 20:00 . drwxr-xr-x 15 root root 3360 Mar 24 20:00 .. crw-rw---- 1 root audio 116, 7 Mar 24 20:00 controlC0 crw-rw---- 1 root audio 116, 6 Mar 24 20:00 hwC0D0 crw-rw---- 1 root audio 116, 5 Mar 24 20:00 pcmC0D0c crw-rw---- 1 root audio 116, 4 Mar 24 20:00 pcmC0D0p crw-rw---- 1 root audio 116, 3 Mar 24 20:00 seq crw-rw---- 1 root audio 116, 2 Mar 24 20:00 timer [root@localhost ~]# [root@localhost ~]# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: MID [HDA Intel MID], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 [root@localhost ~]# [root@localhost ~]# aplay -L front:CARD=MID,DEV=0 HDA Intel MID, HDA Generic Front speakers surround40:CARD=MID,DEV=0 HDA Intel MID, HDA Generic 4.0 Surround output to Front and Rear speakers surround41:CARD=MID,DEV=0 HDA Intel MID, HDA Generic 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=MID,DEV=0 HDA Intel MID, HDA Generic 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=MID,DEV=0 HDA Intel MID, HDA Generic 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=MID,DEV=0 HDA Intel MID, HDA Generic 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default Default [root@localhost ~]# [root@localhost ~]# alsamixer ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: Connection refused alsamixer: function snd_ctl_open failed for default: Connection refused

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