June 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 6/6] ASoC: fsl_ssi: adjust set DAI format in AC'97 mode
by Maciej S. Szmigiero 28 Jun '15

28 Jun '15

Adjust set DAI format function in fsl_ssi driver so it doesn't fail and clears RXDIR in AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e79dc16..d043c7c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,14 +900,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + if (!fsl_ssi_is_ac97(ssi_private)) + return -EINVAL; } stcr |= strcr; srcr |= strcr; - if (ssi_private->cpu_dai_drv.symmetric_rates) { - /* Need to clear RXDIR when using SYNC mode */ + if (ssi_private->cpu_dai_drv.symmetric_rates + || fsl_ssi_is_ac97(ssi_private)) { + /* Need to clear RXDIR when using SYNC or AC97 mode */ srcr &= ~CCSR_SSI_SRCR_RXDIR; scr |= CCSR_SSI_SCR_SYN; }

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[alsa-devel] [PATCH] ASoC: fsl_ssi: fix AC'97 mode
by Maciej S. Szmigiero 28 Jun '15

28 Jun '15

Currently the AC'97 mode in fsl_ssi driver isn't functional. This patch implements the following changes to make it work properly: * IPG clock have to be enabled during AC'97 CODEC register access, * AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params, * AC'97 bus can support asymmetric playback/capture rates, * Check whether setting AC'97 ops succeeded and clean them on removal so the driver can be reloaded, * AC'97 CODEC will be instantiated in AC'97 mode, * Set DAI format function small fixes in AC'97 mode. Tested on UDOO board. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e0..9a63df2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,14 +900,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + if (!fsl_ssi_is_ac97(ssi_private)) + return -EINVAL; } stcr |= strcr; srcr |= strcr; - if (ssi_private->cpu_dai_drv.symmetric_rates) { - /* Need to clear RXDIR when using SYNC mode */ + if (ssi_private->cpu_dai_drv.symmetric_rates + || fsl_ssi_is_ac97(ssi_private)) { + /* Need to clear RXDIR when using SYNC or AC97 mode */ srcr &= ~CCSR_SSI_SRCR_RXDIR; scr |= CCSR_SSI_SCR_SYN; } @@ -1101,6 +1103,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, @@ -1127,10 +1130,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1151,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1163,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1182,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; } @@ -1291,6 +1313,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) struct resource *res; void __iomem *iomem; char name[64]; + bool newbinding; of_id = of_match_device(fsl_ssi_ids, &pdev->dev); if (!of_id || !of_id->data) @@ -1320,7 +1343,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1357,7 +1384,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; } @@ -1405,7 +1434,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) * that the machine driver uses new binding which does not require * SSI driver to trigger machine driver's probe. */ - if (!of_get_property(np, "codec-handle", NULL)) + newbinding = !of_get_property(np, "codec-handle", NULL); + if (newbinding) goto done; /* Trigger the machine driver's probe function. The platform driver @@ -1434,6 +1464,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (newbinding && fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card: @@ -1458,6 +1509,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

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[alsa-devel] PulseAudio and SNDRV_PCM_INFO_BATCH
by Arun Raghavan 28 Jun '15

28 Jun '15

Hi folks, I'd like to bring this one up again, since we are currently in the sub-optimal position of forcing ~100 ms latency on USB devices. The original thread is here -- http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/069666.h… I see two flags that are possibly of consequence here: SNDRV_PCM_INFO_BATCH and SNDRV_PCM_INFO_BLOCK_TRANSFER. I'm not sure what these mean -- the documentation mentions "double buffering" for the batch flag, and just that the block transfer means "block transfer". :-) We've spoken about batch meaning either transfers in period size chunks, or some fixed chunk size. It seems that it would make more sense for it to mean the former, and block transfer to mean the latter. So I guess the first thing that would be nice to have is a clear meaning of these two flags. With this done, we could potentially get to the API to report the transfer size from the driver. I did notice that there is a snd_pcm_hw_params_get_fifo_size(). Is this something we could use for the purpose of transfer size reporting, by any chance? Cheers, Arun

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[alsa-devel] snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 4.
by Adam Nielsen 27 Jun '15

27 Jun '15

Hi all, I'm trying to track down a problem with my media player (xmms2) where it has stopped advancing to the next track after reaching the end of the song. I have the same distro with the same package versions on two slightly different computers, and it works fine on one but not the other. One difference between the two is that although both have Intel HD Audio, each is provided by a different type of chip. I was asked to try pulseaudio to see if the problem persisted there and it did, however upon loading the pulseaudio server it told me this: E: [alsa-source-ALC892 Analog] alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 4. E: [alsa-source-ALC892 Analog] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. E: [alsa-source-ALC892 Analog] alsa-util.c: snd_pcm_dump(): E: [alsa-source-ALC892 Analog] alsa-util.c: Direct Snoop PCM E: [alsa-source-ALC892 Analog] alsa-util.c: Its setup is: E: [alsa-source-ALC892 Analog] alsa-util.c: stream : CAPTURE E: [alsa-source-ALC892 Analog] alsa-util.c: access : MMAP_INTERLEAVED E: [alsa-source-ALC892 Analog] alsa-util.c: format : S32_LE E: [alsa-source-ALC892 Analog] alsa-util.c: subformat : STD E: [alsa-source-ALC892 Analog] alsa-util.c: channels : 2 E: [alsa-source-ALC892 Analog] alsa-util.c: rate : 48000 E: [alsa-source-ALC892 Analog] alsa-util.c: exact rate : 48000 (48000/1) E: [alsa-source-ALC892 Analog] alsa-util.c: msbits : 32 E: [alsa-source-ALC892 Analog] alsa-util.c: buffer_size : 16384 E: [alsa-source-ALC892 Analog] alsa-util.c: period_size : 1024 E: [alsa-source-ALC892 Analog] alsa-util.c: period_time : 21333 E: [alsa-source-ALC892 Analog] alsa-util.c: tstamp_mode : ENABLE E: [alsa-source-ALC892 Analog] alsa-util.c: period_step : 1 E: [alsa-source-ALC892 Analog] alsa-util.c: avail_min : 15943 E: [alsa-source-ALC892 Analog] alsa-util.c: period_event : 0 E: [alsa-source-ALC892 Analog] alsa-util.c: start_threshold : -1 E: [alsa-source-ALC892 Analog] alsa-util.c: stop_threshold : 4611686018427387904 E: [alsa-source-ALC892 Analog] alsa-util.c: silence_threshold: 0 E: [alsa-source-ALC892 Analog] alsa-util.c: silence_size : 0 E: [alsa-source-ALC892 Analog] alsa-util.c: boundary : 4611686018427387904 E: [alsa-source-ALC892 Analog] alsa-util.c: Hardware PCM card 1 'HDA Intel PCH' device 0 subdevice 0 E: [alsa-source-ALC892 Analog] alsa-util.c: Its setup is: E: [alsa-source-ALC892 Analog] alsa-util.c: stream : CAPTURE E: [alsa-source-ALC892 Analog] alsa-util.c: access : MMAP_INTERLEAVED E: [alsa-source-ALC892 Analog] alsa-util.c: format : S32_LE E: [alsa-source-ALC892 Analog] alsa-util.c: subformat : STD E: [alsa-source-ALC892 Analog] alsa-util.c: channels : 2 E: [alsa-source-ALC892 Analog] alsa-util.c: rate : 48000 E: [alsa-source-ALC892 Analog] alsa-util.c: exact rate : 48000 (48000/1) E: [alsa-source-ALC892 Analog] alsa-util.c: msbits : 32 E: [alsa-source-ALC892 Analog] alsa-util.c: buffer_size : 16384 E: [alsa-source-ALC892 Analog] alsa-util.c: period_size : 1024 E: [alsa-source-ALC892 Analog] alsa-util.c: period_time : 21333 E: [alsa-source-ALC892 Analog] alsa-util.c: tstamp_mode : ENABLE E: [alsa-source-ALC892 Analog] alsa-util.c: period_step : 1 E: [alsa-source-ALC892 Analog] alsa-util.c: avail_min : 1024 E: [alsa-source-ALC892 Analog] alsa-util.c: period_event : 0 E: [alsa-source-ALC892 Analog] alsa-util.c: start_threshold : 1 E: [alsa-source-ALC892 Analog] alsa-util.c: stop_threshold : 4611686018427387904 E: [alsa-source-ALC892 Analog] alsa-util.c: silence_threshold: 0 E: [alsa-source-ALC892 Analog] alsa-util.c: silence_size : 0 E: [alsa-source-ALC892 Analog] alsa-util.c: boundary : 4611686018427387904 E: [alsa-source-ALC892 Analog] alsa-util.c: appl_ptr : 0 E: [alsa-source-ALC892 Analog] alsa-util.c: hw_ptr : 8236 So I am reporting this in the hope that someone understands what the problem is and knows how to fix it. The wiki said to include the output of aadebug with a bug report, so here's that: ALSA Audio Debug v0.2.0 - Sat Jun 27 17:26:44 AEST 2015 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/agpl-3.0.txt Kernel ---------------------------------------------------- Linux korath.teln.shikadi.net 4.0.6-1-ARCH #1 SMP PREEMPT Tue Jun 23 14:25:08 CEST 2015 x86_64 GNU/Linux Advanced Linux Sound Architecture Driver Version k4.0.6-1-ARCH. Loaded Modules -------------------------------------------- snd_hrtimer 16384 1 snd_seq 61440 1 snd_seq_device 16384 1 snd_seq snd_hda_codec_hdmi 53248 1 snd_hda_codec_realtek 65536 1 snd_hda_codec_generic 65536 1 snd_hda_codec_realtek snd_hda_intel 28672 9 snd_hda_controller 28672 1 snd_hda_intel snd_hda_codec 114688 5 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_codec_generic,snd_hda_intel,snd_hda_controller snd_hwdep 16384 1 snd_hda_codec snd_pcm 90112 6 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel,snd_hda_controller snd_timer 28672 5 snd_hrtimer,snd_pcm,snd_seq snd 69632 25 snd_hda_codec_realtek,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_hda_codec_generic,snd_hda_codec,snd_hda_intel,snd_seq_device Proc Asound ----------------------------------------------- 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xf7c34000 irq 30 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7c30000 irq 28 1: : sequencer 2: [ 1] : control 3: [ 1- 0]: digital audio playback 4: [ 1- 0]: digital audio capture 5: [ 1- 1]: digital audio playback 6: [ 1- 2]: digital audio capture 7: [ 1- 0]: hardware dependent 8: [ 0] : control 9: [ 0- 3]: digital audio playback 10: [ 0- 7]: digital audio playback 11: [ 0- 0]: hardware dependent 33: : timer 01-00: HDA Codec 0 00-00: HDA Codec 0 00-03: HDMI 0 : HDMI 0 : playback 1 00-07: HDMI 1 : HDMI 1 : playback 1 01-00: ALC892 Analog : ALC892 Analog : playback 1 : capture 1 01-01: ALC892 Digital : ALC892 Digital : playback 1 01-02: ALC892 Alt Analog : ALC892 Alt Analog : capture 1 Client info cur clients : 2 peak clients : 2 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 128 : "TiMidity" [User] Port 0 : "TiMidity port 0" (-We-) Port 1 : "TiMidity port 1" (-We-) Port 2 : "TiMidity port 2" (-We-) Port 3 : "TiMidity port 3" (-We-) Output pool : Pool size : 500 Cells in use : 0 Peak cells in use : 0 Alloc success : 0 Alloc failures : 0 Input pool : Pool size : 1000 Cells in use : 0 Peak cells in use : 0 Alloc success : 0 Alloc failures : 0 Dev Snd --------------------------------------------------- total 0 drwxr-xr-x 2 root root 80 Jun 27 13:31 by-path crw-rw----+ 1 root audio 116, 8 Jun 27 13:31 controlC0 crw-rw----+ 1 root audio 116, 2 Jun 27 13:31 controlC1 crw-rw----+ 1 root audio 116, 11 Jun 27 13:31 hwC0D0 crw-rw----+ 1 root audio 116, 7 Jun 27 13:31 hwC1D0 crw-rw----+ 1 root audio 116, 9 Jun 27 17:12 pcmC0D3p crw-rw----+ 1 root audio 116, 10 Jun 27 17:12 pcmC0D7p crw-rw----+ 1 root audio 116, 4 Jun 27 17:12 pcmC1D0c crw-rw----+ 1 root audio 116, 3 Jun 27 17:13 pcmC1D0p crw-rw----+ 1 root audio 116, 5 Jun 27 13:31 pcmC1D1p crw-rw----+ 1 root audio 116, 6 Jun 27 13:31 pcmC1D2c crw-rw----+ 1 root audio 116, 1 Jun 27 13:31 seq crw-rw----+ 1 root audio 116, 33 Jun 27 13:31 timer CPU ------------------------------------------------------- model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3097.500 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3000.000 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3295.664 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3120.703 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3001.406 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3000.351 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3645.585 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3000.468 RAM ------------------------------------------------------- MemTotal: 16362900 kB SwapTotal: 0 kB Hardware -------------------------------------------------- 00:02.0 VGA compatible controller: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor Integrated Graphics Controller (rev 06) 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 04) Interupts ------------------------------------------------- CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 0: 19 0 0 0 0 0 0 0 IO-APIC-edge timer 1: 1 0 0 0 0 0 0 1 IO-APIC-edge i8042 7: 9 0 0 0 0 0 0 0 IO-APIC-edge 8: 0 0 0 1 0 0 0 0 IO-APIC-edge rtc0 9: 1 0 0 0 1 0 0 1 IO-APIC-fasteoi acpi 16: 20 0 0 0 4 1 3 1 IO-APIC 16-fasteoi ehci_hcd:usb1 23: 23 0 5 0 2 1 1 1 IO-APIC 23-fasteoi ehci_hcd:usb2 24: 236257 13391 10344 8635 7919 6578 6237 5429 PCI-MSI-edge xhci_hcd 25: 27200 5286 3884 3536 10293 3395 2433 1864 PCI-MSI-edge 0000:00:1f.2 26: 3 6 0 0 0 2 0 2 PCI-MSI-edge mei_me 27: 231790 38059 33089 33480 38333 18853 15356 13578 PCI-MSI-edge eth0 28: 549701 20320 18918 16545 17439 14054 12933 11245 PCI-MSI-edge snd_hda_intel 29: 1936 799 524 368 520 541 356 265 PCI-MSI-edge i915 30: 25 272 13 2 477 200 8 2 PCI-MSI-edge snd_hda_intel NMI: 15 14 14 14 9 7 9 10 Non-maskable interrupts LOC: 1051971 756503 703396 726403 198333 221752 198703 213191 Local timer interrupts SPU: 0 0 0 0 0 0 0 0 Spurious interrupts PMI: 15 14 14 14 9 7 9 10 Performance monitoring interrupts IWI: 0 0 0 0 0 0 1 0 IRQ work interrupts RTR: 7 0 0 0 0 0 0 0 APIC ICR read retries RES: 13573 10138 7672 8132 3954 3624 3500 2935 Rescheduling interrupts CAL: 32694 13897 13827 14113 968 1025 1057 1107 Function call interrupts TLB: 7503 7559 9116 13064 9259 6773 5330 4805 TLB shootdowns TRM: 0 0 0 0 0 0 0 0 Thermal event interrupts THR: 0 0 0 0 0 0 0 0 Threshold APIC interrupts MCE: 0 0 0 0 0 0 0 0 Machine check exceptions MCP: 48 48 48 48 48 48 48 48 Machine check polls HYP: 0 0 0 0 0 0 0 0 Hypervisor callback interrupts ERR: 9 MIS: 0 Please let me know if I can provide any other info. Many thanks, Adam.

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[alsa-devel] [PATCH] ALSA: hda - Add headset support to Acer Aspire V5
by Takashi Iwai 27 Jun '15

27 Jun '15

Acer Aspire V5 with ALC282 codec needs the similar quirk like Dell laptops to support the headset mic. The headset mic pin is 0x19 and it's not exposed by BIOS, thus we need to fix the pincfg as well. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=96201 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 78dfec9d34c8..cab09c42598b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4484,6 +4484,7 @@ enum { ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, + ALC269_FIXUP_ASPIRE_HEADSET_MIC, ALC269_FIXUP_ASUS_X101_FUNC, ALC269_FIXUP_ASUS_X101_VERB, ALC269_FIXUP_ASUS_X101, @@ -4761,6 +4762,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_no_hp_mic, }, + [ALC269_FIXUP_ASPIRE_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* headset mic w/o jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, [ALC286_FIXUP_SONY_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5096,6 +5106,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), + SND_PCI_QUIRK(0x1025, 0x072d, "Acer Aspire V5-571G", ALC269_FIXUP_ASPIRE_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), -- 2.4.4

1 0

[alsa-devel] Sennheiser HDVD 800 Support
by Armin Ranjbar 26 Jun '15

26 Jun '15

Dear everyone, i have noticed that alas doesn't support HDVD-800 USB DAC, i can not hear anything. note that this device requires it's own driver under windows/mac. this is my alsa-info output: http://www.alsa-project.org/db/?f=fe56e65add73248baf5307a04b6b9deb2b236004 output of pasuspender -- aplay -Dplughw:2 some_file.wav: ALSA lib pcm_hw.c:1667:(_snd_pcm_hw_open) Invalid value for card main:722: audio open error: No such file or directory output of aplay -v -fdat --dump-hw-params -d1 -Dhw:S20 /dev/zero: http://pastebin.com/p2Daxza6 speaker-test -c2 -twav -Dplughw:S20 starts OK, but i can't hear anything/ product page: http://en-us.sennheiser.com/headphone-audio-amplification-hdvd-800-acoustic I'm open to help if you need anything to be done. --- Armin ranjbar

1 1

[alsa-devel] Audio Driver Works In Ubuntu but Crash in Debian After 2nd Run
by Azizul Hakim 26 Jun '15

26 Jun '15

Dear altruists, I'm writing a sound card driver based on ALSA API. My driver works perfectly on Ubuntu 12.04, 13.10 and 14.04. There is no problem on how many times I play a sound file using my driver. Everything works great. But same driver has some issue on debian distribution. I've been trying to test it on BeagleBone and during the first run the driver works pretty well. But as soon as I play a second sound file, it crash. I thought may be I did something wrong in releasing resources after 1st run. But so far from my debugging it looks like there is no problem with resource release. Also same driver works on Ubuntu as many times as I test. I even tested with the same sound file. On debian, I get the following message is kernel log- Apr 23 20:26:41 beaglebone kernel: [ 183.631905] ALSA sound/core/pcm_lib.c:1910 playback write error (DMA or IRQ trouble?) It would be highly appreciated if I can get some suggestions on which direction I should try to resolve this issue. Is there any known issue with debian distribution or anyone faced similar kind of problem before? Thanks Azizul Hakim

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[alsa-devel] Using plughw device and samplerate conversion with an ALSA driver not supporting memory mapped access
by Christian Gruber 26 Jun '15

26 Jun '15

Hello, I have problems using the plughw device with samplerate conversion on an embedded board having an ALSA driver, which does not support memory mapped access. The audio codec only supports 48 kHz sampling rate. The problem occurs, when I want to playback for instance a 16 kHz file via plughw device, which is the default device. Interestingly the behaviour of the ALSA system is as follows: 1. aplay -D default audio_16kHz.wav does not work 2. aplay -D default -M audio_16_kHz.wav works correctly This is quite strange, since although the ALSA driver does not support memory mapped access, using the option -M works, but the default read write access method without option -M does not, although the driver supports read write access. In this case the function snd_pcm_hw_params() returns with an error (EINVAL). I started debugging this and could see, that in the second case, when using option -M, the plughw device establishes the following signal processing chain: plughw -> rate -> mmap_emul -> hw This is what I expected from the plughw device. It first introduces a samplerate converter for conversion from 16 kHz to 48 kHz, since the codec only supports 48 kHz. And second it introduces a memory mapped access emulator to be able to use memory mapped access method on the application side and read write access method on the driver side (hw device). In the first case without using option -M the plughw device establishes the following processing chain: plughw -> rate -> hw This means, it does not introduce a memory mapped access emulator, which is comprehensible as well, since the application already uses read write access method, the same as the ALSA driver does. But nevertheless an error occurs. Further debugging showed, that the error occurs in the function hw_refine_call() (pcm_hw.c) during the call to _snd_pcm_hw_params_internal() within the function snd_pcm_plug_hw_params() (pcm_plug.c). For some reason the rate plugin wants to use memory mapped access only on the hw device. So I ask myself, if this is the expected behaviour of the rate plugin. Does the rate plugin only support memory mapped access? If yes, why is no mmap_emul plugin introduced into the signal processing chain in this case? If not, why does the rate plugin choose memory mapped access method? Is the ALSA driver maybe buggy giving wrong information about its hw capabilities? Can anybody help me? Regards, Christian

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[alsa-devel] [PATCH] ASoC: dapm: Fix deadlock on auto-disable mux controls
by Charles Keepax 26 Jun '15

26 Jun '15

The commit 02aa78abec6e ("ASoC: DAPM: Add APIs to create individual DAPM controls.") added locking to the snd_soc_dapm_new_control function but did not update the call to snd_soc_dapm_new_control in the auto-disable mux code, this appears to be because the patches were sent at fairly similar times. This patch change the call in the auto-disable mux code to use the new snd_soc_dapm_new_control_unlocked function instead. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c9..a47a8ce 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -389,8 +389,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->value = template.on_val; - data->widget = snd_soc_dapm_new_control(widget->dapm, - &template); + data->widget = snd_soc_dapm_new_control_unlocked( + widget->dapm, &template); if (!data->widget) { ret = -ENOMEM; goto err_name; -- 1.7.2.5

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[alsa-devel] DPCM: BE to BE cpu DAIs question‏
by srv_kaichieh.chaung 26 Jun '15

26 Jun '15

Hi, Our platform use ASoC structure. Currently, we want to refactor our driver to use DPCM structure. I followed the documentation of DPCM, but have some trouble implementing hostless BE to BE dai links. My question is: 1. Is there any code example for this(hostless BE to BE)? Since I didn't see any existing code on linux kernel. 2. From documentation, i have to assgin a snd_soc_pcm_stream for BE dais. It seems that the BE<->BE dai links can only support one spec (format/rates/ch). Or am i getting it wrong, does it support dynamically change of the spec? Thanks in advanced! -------------------------------------------------------------------------------------- I have tried some implementations, but failed. My linux version is 3.18. Illustration, PCM -----------------> BE1 ↑ -------< BE2 Code snippet, static const struct snd_soc_dapm_route machine_routes[] = { {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, FE_STNAME}, {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, BE2_CAPTURE_CPU_DAI_STNAME}, }; static const struct snd_soc_pcm_stream dai_params = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rate_min = 48000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, }; static struct snd_soc_dai_link mt_soc_dai_common[] = { { .name = FE_PCM, .stream_name = FE_STNAME, .cpu_dai_name = FE_CPUDAINAME, // cannot use dummy!!!! .platform_name = FE_PCMNAME, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .init = mt_soc_audio_init3, .ops = &kc_snd_soc_ops, .dynamic = 1, .dpcm_playback = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, }, /* BE DAIs */ { .name = "BE2", .cpu_dai_name = BE2_PLAYBACK_CPU_DAI, //"snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, { .name = BE1_PLAYBACK, .cpu_dai_name = BE1_PLAYBACK_CPU_DAI, .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, 1. when only routes PCM-->BE1, If i add .params in BE1 soc dai link, when i start PCM, it will crash. and get "Unable to handle kernel NULL pointer dereference at virtual address 000000d0" Seems to crash in dpcm_be_dai_startup() If i removed .params, it will not crash, and play normally. 2. when only with BE2--> BE1, BE2 is never trigger, where did i get wrong? Thanks. ************* Email Confidentiality Notice ******************** The information contained in this e-mail message (including any attachments) may be confidential, proprietary, privileged, or otherwise exempt from disclosure under applicable laws. It is intended to be conveyed only to the designated recipient(s). Any use, dissemination, distribution, printing, retaining or copying of this e-mail (including its attachments) by unintended recipient(s) is strictly prohibited and may be unlawful. If you are not an intended recipient of this e-mail, or believe that you have received this e-mail in error, please notify the sender immediately (by replying to this e-mail), delete any and all copies of this e-mail (including any attachments) from your system, and do not disclose the content of this e-mail to any other person. Thank you!

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