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

[alsa-devel] [PATCH] ALSA: HDA: Add quirk for Lenovo U350
by David Henningsson 23 May '11

23 May '11

Since the auto-parser for Conexant 5066 is yet not activated, I guess it makes sense to keep adding quirks? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[PATCH] ALSA: HDA: Add quirk for Lenovo U350
by David Henningsson 23 May '11

23 May '11

Add model=asus quirk for Lenovo Ideapad U350 to make internal mic work correctly. Cc: stable(a)kernel.org (2.6.38+) BugLink: http://bugs.launchpad.net/bugs/751681 Reported-by: Kent Baxley <kent.baxley(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4f37477..e8725d2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3098,6 +3098,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} -- 1.7.4.1 --------------090201020202020607060101--

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[alsa-devel] ALSA pcm_lib.c:1789: playback write error (DMA or IRQ trouble?) ALSA pcm_native.c:1531: playback drain error
by Raymond Yau 23 May '11

23 May '11

For those HDA Codec which support rate lower than 44100 Hz with prealloc = 4096 ALSA pcm_lib.c:1789: playback write error (DMA or IRQ trouble?) ALSA hda_intel.c:1678: azx_pcm_prepare: bufsize=0xd7500, format=0x4311 This error occr when aplay using a period time longer than 10 seconds ( usually occur above 12 seconds) aplay -v -Dhw:1 --period-time=12000000 11025.wav Playing WAVE '11025.wav' : Signed 16 bit Little Endian, Rate 11025 Hz, Stereo Hardware PCM card 1 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 11025 exact rate : 11025 (11025/1) msbits : 16 buffer_size : 264576 period_size : 132288 period_time : 11998911 tstamp_mode : NONE period_step : 1 avail_min : 132288 period_event : 0 start_threshold : 264576 stop_threshold : 264576 silence_threshold: 0 silence_size : 0 boundary : 1083703296 appl_ptr : 0 hw_ptr : 0 ALSA pcm_native.c:1531: playback drain error (DMA or IRQ trouble?) ALSA hda_intel.c:1678: azx_pcm_prepare: bufsize=0x400000, format=0x511 This error occur with speaker-test which use maximum buffer size speaker-test -d -c 2 -t sine --rate=8000 -Dhw:CARD=Intel speaker-test 1.0.24.2 Playback device is hw:CARD=Intel Stream parameters are 8000Hz, S16_LE, 2 channels Sine wave rate is 440.0000Hz Rate set to 8000Hz (requested 8000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Using max buffer size 1048576 Periods = 4 was set period_size = 262144 was set buffer_size = 1048576 Hardware PCM card 1 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 1048576 period_size : 262144 period_time : 32768000 tstamp_mode : NONE period_step : 1 avail_min : 262144 period_event : 0 start_threshold : 1048576 stop_threshold : 1048576 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 0 - Front Left 1 - Front Right Time per period = 20.075443

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[alsa-devel] Does amixer has to be used to make a complete path in ASoC?
by Feng Ye 23 May '11

23 May '11

Hello, I wrote ASoC driver with interacts with WM8750 codec chip. It does not work, no sound at all. Some codec registers are not programmed right, and the check_power callback (in soc-dapm.c) always return 0. I think the problem may not be in the driver, instead, amixer should be used to select the codec mixer settings to build a complete audio path (like the LEFT MIXER in wm8750). I have following questions: 1. Is my understanding correct? amixer has to be used before aplay? 2. Is there way to do it in the code instead of using amixer? 3. If amixer has to be used, where I can find sample wm8750 amixer setting usage for a simple playback from I2S in to Lout1 & Rout1 out? Here's the simple code in my driver, just FYI: static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Speaker", NULL), }; static const struct snd_soc_dapm_route wm8750_audio_map[] = { { "Ext Speaker", NULL, "LOUT1" }, { "Ext Speaker", NULL, "ROUT1" }, }; ... (in dai_link init function): err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, ARRAY_SIZE(wm8750_dapm_widgets)); ... snd_soc_dapm_add_routes(codec, wm8750_audio_map, ARRAY_SIZE(wm8750_audio_map)); snd_soc_dapm_sync(codec); ... (in snd_soc_ops startup function): snd_soc_dapm_enable_pin(codec, "Ext Speaker"); Thanks, Feng

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[alsa-devel] ask about audio latency
by 周勇发 22 May '11

22 May '11

hi I want to get the latency between record and playback, so i use following solution to calculate it. #audio.sh arecord -d 3 -r 96000 in.wav & sleep 1 aplay mode.wav model.wav is a sine wave. this file is used for calculate the delta timestamp of record and playback. I have append do_gettimeofday() function in azx_stream_start() in hda_intel.c file. it will printk the dma trigger time after trigger dma. the result is about 12ms. as far as i known. fifo size is 192, sample rate is 96000, so it will take about 2ms latency, my question is where does the other 10ms latency come from? Thanks -- Best Regards！

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[alsa-devel] ASoC updates for 2.6.40
by Mark Brown 22 May '11

22 May '11

The following changes since commit 4a787a3ff3f419c23ab0a5cef677fa441356b818: Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2011-05-20 11:25:32 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.40 A few more small fixes. Jarkko Nikula (1): ASoC: core: Don't set "(null)" as a driver name Mark Brown (1): Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.40 Stephen Warren (2): ASoC: spdif-dit: Add missing MODULE_* ASoC: Tegra: Fix compile when debugfs not enabled sound/soc/codecs/spdif_transciever.c | 8 ++++++-- sound/soc/soc-core.c | 2 +- sound/soc/tegra/tegra_i2s.c | 2 +- 3 files changed, 8 insertions(+), 4 deletions(-)

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Re: [alsa-devel] playback write error(DMA or IRQ trouble?)
by innoway＠mymelody.com 22 May '11

22 May '11

I'll try to fix the programs from this Monday and I think I have to learn alsa system more deeply because I 'm a newcomer about it. Thank you 2011/5/20 <innoway(a)mymelody.com>: > I develop an embedded system on the Linux board with Freescale i.MX27 > (ARM processor). I succeed on sound, which is transferred using > Synchronous Serial Interface (SSI). It,s fine and then I try to do test > of inserting or pulling out a USB memory stick in the USB host connector > while sounding. I get the following error message from > snd_pcm_lib_write1() of <linux src>/sound /core/pcm_lib.c after waiting > for a while (about 10sec). playback write error (DMA or IRQ > trouble?) and then from snd_pcm_drain() of <linux > src>/sound/core/pcm_native.c playback drain error (DMA or IRQ > trouble?) (No sound during this period) > > and then it is sounding again. > > I want to prevent the sound from stopping even if something happens > suddenly such as the USB memory stick is plugged in or out. > > I try to fix the above programs, but I don't succeed. > > Can someone give me the clue? [13:21:38]:/proc# cat interrupts > CPU0 > 12: 3456 - mxc_i2c > 16: 141 - CSPI_IRQ > 18: 952 - mxcintuart > 26: 1051255 - MXC Timer Tick > 28: 29446 - PCMCIA IRQ, ide0 > 50: 33 - fec > 54: 793 - ehci_hcd:usb1 > 56: 1 - arc_udc > 61: 0 - LCDC > 117: 0 - m66291_udc > 147: 0 - PMIC_IRQ > 157: 337179 - SMSC_LAN911x_ISR > 176: 0 - fec(MII) > > [13:35:03]:/proc# cat dma > dma channel 0: unused > dma channel 1: unused > dma channel 2: unused > dma channel 3: unused > dma channel 4: unused > dma channel 5: unused > dma channel 6: unused > dma channel 7: unused > dma channel 8: unused > dma channel 9: unused > dma channel 10: unused > dma channel 11: unused > dma channel 12: unused > dma channel 13: unused > dma channel 14: unused > dma channel 15: unused Thank you > > Masanori Inoue > This error also occur in snd-hda-intel echo 32768 > /proc/asound/Intel/pcm0p/sub0/prealloc speaker-test which use maximum buffer size also have (DMA or IRQ error in system log THis seem to be related to the schedule_timeout tout = schedule_timeout(msecs_to_jiffies(10000)); I guess this is why PA server treat avail > 10 seconds is invalid _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] UCM files for tegra-wm8903 on seaboard
by Stephen Warren 21 May '11

21 May '11

Liam, I've come up with a basic set of UCM files for Tegra Seaboard, included below. I briefly tested parts of them with alsaucm. I do have a few questions on UCM and how to write these files before I go too far: Some of the answers may depend on, or be impacted by, policy decisions by the user of UCM; general desktop PulseAudio may have different requirements to a more tightly controlled ChromeOS configuration. I'm not sure how much that impacts the UCM files themselves, or whether the UCM files are supposed to be fully general and hence the same for all OSs and policy setups. 1) Can multiple verbs be active at once? In other words, I have cases defined for HiFi (playback) and Record (capture). Should these be merged together such that both playback and capture can operate at once, or can an application just active both verbs at once when it wants to do multiple things? 2) In the ChromeOS kernel, there's a separate PCM for playback to WM8903 and playback to the SPDIF controller, and hence on to HDMI. Since these use different values for PlaybackPCM, I defined separate use-cases for analog and HDMI/SPDIF playback. Is this the typical way of splitting things up, or would one typically have a unified use-case with the PlaybackPCM Value being part of the SectionDevice? 3) For the HiFi case, there are a couple of volume controls in serial; i.e. "Digital Playback Volume" affects both speaker/headphone output whereas "Speaker Volume" and "Headphone Volume" affect just one of the devices. I've defined a PlaybackVolume Value for the overall verb and the same Value for each device. Are applications able to query the Value separately for verb and devices, or will using the same name cause conflicts? Is there any kind of (de-facto) standard for how to represent this situation? Thanks very much for any feedback. ------------------ tegra-wm8903-seaboard.conf: # UCM configuration for Tegra WM8903 driver for Seaboard SectionUseCase."Record" { File "record" Comment "Record" } SectionUseCase."HiFi" { File "hifi" Comment "Play HiFi quality Music" } SectionUseCase."HiFi HDMI" { File "hdmi" Comment "Play HiFi quality Music over HDMI" } SectionDefaults [ cdev "hw:0" cset "name='Headphone Switch' off,off" cset "name='Left Output Mixer DACL Switch' on" cset "name='Left Output Mixer DACR Switch' off" cset "name='Left Output Mixer Left Bypass Switch' off" cset "name='Left Output Mixer Right Bypass Switch' off" cset "name='Right Output Mixer DACL Switch' off" cset "name='Right Output Mixer DACR Switch' on" cset "name='Right Output Mixer Left Bypass Switch' off" cset "name='Right Output Mixer Right Bypass Switch' off" cset "name='Int Spk Switch' off" cset "name='Speaker Switch' off,off" cset "name='Left Speaker Mixer DACL Switch' on" cset "name='Left Speaker Mixer DACR Switch' off" cset "name='Left Speaker Mixer Left Bypass Switch' off" cset "name='Left Speaker Mixer Right Bypass Switch' off" cset "name='Right Speaker Mixer DACL Switch' off" cset "name='Right Speaker Mixer DACR Switch' on" cset "name='Right Speaker Mixer Left Bypass Switch' off" cset "name='Right Speaker Mixer Right Bypass Switch' off" cset "name='Left Input PGA Switch' off" cset "name='Right Input PGA Switch' off" cset "name='Right Inverting Input Mux' 'IN2R'" ] ------------------ hifi: SectionVerb { Value { PlaybackPCM "hw:0,0" PlaybackVolume "Digital Playback Volume" } } SectionDevice."Speakers".0 { Comment "Internal Speakers" EnableSequence [ cdev "hw:0" cset "name='Speaker Switch' on,on" cset "name='Int Spk Switch' on" ] DisableSequence [ cdev "hw:0" cset "name='Speaker Switch' off,off" cset "name='Int Spk Switch' off" ] Value { PlaybackVolume "Speaker Volume" PlaybackSwitch "Speaker Switch" } } SectionDevice."Headphones".0 { Comment "Headphone Jack" EnableSequence [ cdev "hw:0" cset "name='Headphone Switch' on,on" ] DisableSequence [ cdev "hw:0" cset "name='Headphone Switch' off,off" ] Value { PlaybackVolume "Headphone Volume" PlaybackSwitch "Headphone Switch" } } ------------------ hifi-hdmi: SectionVerb { Value { PlaybackPCM "hw:0,1" } } SectionDevice."HDMI".0 { Comment "HDMI" } ------------------ record: SectionVerb { Value { CapturePCM "hw:0,0" CaptureVolume "Digital Capture Volume" } } SectionDevice."AMIC".0 { Comment "Analog Microphone Jack" EnableSequence [ cdev "hw:0" cset "name='ADC Input' 'ADC'" cset "name='Left Capture Mux' 'Right'" cset "name='Right Capture Mux' 'Right'" ] Value { CaptureVolume "Right Input PGA Volume" } } SectionDevice."DMIC".0 { Comment "Internal Digital Microphone" EnableSequence [ cdev "hw:0" cset "name='ADC Input' 'DMIC'" cset "name='Left Capture Mux' 'Left'" cset "name='Right Capture Mux' 'Right'" ] Value { } } -- nvpublic

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[alsa-devel] [PATCH] ASoC: spdif-dit: Add missing MODULE_*
by Stephen Warren 21 May '11

21 May '11

MODULE_ALIAS is required so that the module will auto-load based on a platform_device registration in the board file. While we're at it, add some other MODULE_*. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/spdif_transciever.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 4c32b54..6a1a7e7 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -21,7 +21,7 @@ #include <sound/pcm.h> #include <sound/initval.h> -MODULE_LICENSE("GPL"); +#define DRV_NAME "spdif-dit" #define STUB_RATES SNDRV_PCM_RATE_8000_96000 #define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE @@ -56,7 +56,7 @@ static struct platform_driver spdif_dit_driver = { .probe = spdif_dit_probe, .remove = spdif_dit_remove, .driver = { - .name = "spdif-dit", + .name = DRV_NAME, .owner = THIS_MODULE, }, }; @@ -74,3 +74,7 @@ static void __exit dit_exit(void) module_init(dit_modinit); module_exit(dit_exit); +MODULE_AUTHOR("Steve Chen <schen(a)mvista.com>"); +MODULE_DESCRIPTION("SPDIF dummy codec driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- 1.7.0.4

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[alsa-devel] [PATCH v1 RESEND] audio: tlv320aic26: fix PLL register configuration
by Michael Williamson 21 May '11

21 May '11

The current PLL configuration code for the tlc320aic26 codec appears to assume a hardcoded system clock of 12 MHz. Use the clock value provided by the DAI_OPS API for the calculation. Tested using a MityDSP-L138 platform providing a 24.576 MHz clock. Signed-off-by: Michael Williamson <michael.williamson(a)criticallink.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This got bounced by the alsa-devel list (I wasn't on list). I'm not sure whose tree this needs to go through, but given the lack of response I'm guessing alsa-devel. If I'm missing a list, any advice would be appreciated. Changes from V0: - Added some parens for clarification - Added some comments to describe what the computations were attempting to do sound/soc/codecs/tlv320aic26.c | 14 +++++++++++--- 1 files changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index e2a7608..7859bdc 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -161,10 +161,18 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL; } - /* Configure PLL */ + /** + * Configure PLL + * fsref = (mclk * PLLM) / 2048 + * where PLLM = J.DDDD (DDDD register ranges from 0 to 9999, decimal) + */ pval = 1; - jval = (fsref == 44100) ? 7 : 8; - dval = (fsref == 44100) ? 5264 : 1920; + /* compute J portion of multiplier */ + jval = fsref / (aic26->mclk / 2048); + /* compute fractional DDDD component of multiplier */ + dval = fsref - (jval * (aic26->mclk / 2048)); + dval = (10000 * dval) / (aic26->mclk / 2048); + dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval); qval = 0; reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg); -- 1.7.0.4

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