September 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 2/4] ASoC: dapm: Refactor the code in snd_soc_dapm_new_pcm
by anish kumar 19 Sep '17

19 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> refactor snd_soc_dapm_new_pcm to reduce the size of this function to facilitate further refactoring. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f51f613..d55cac6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3821,19 +3821,15 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kasprintf(card->dev, GFP_KERNEL, "Anonymous Configuration %d", count); - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, strlen(config->stream_name) + 1, GFP_KERNEL); - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + } + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; } config++; } -- 2.5.4 (Apple Git-61)

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[alsa-devel] Resend: [PATCH 1/4] ASoC: dapm: fix error path in snd_soc_dapm_new_pcm
by anish kumar 19 Sep '17

19 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Before freeing the memory, check if it was allcoated. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..f51f613 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3907,9 +3907,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, (void *)private_value); outfree_link_name: devm_kfree(card->dev, link_name); -outfree_w_param: for (count = 0 ; count < num_params; count++) devm_kfree(card->dev, (void *)w_param_text[count]); +outfree_w_param: devm_kfree(card->dev, w_param_text); return ret; -- 2.5.4 (Apple Git-61)

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[alsa-devel] [RFCv2][PATCH 00/38] alsa-utils: axfer: rewrite aplay
by Takashi Sakamoto 19 Sep '17

19 Sep '17

Hi, This large patchset is to introduce a new command, named as 'axfer' (ALSA transfer). This comes from my work to rewrite aplay. Initial patchset was already posted as RFCv1. [alsa-devel][PATCH 00/23] alsa-utils: rewrite aplay http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/124132.html Unlike the initial patchset, this patchset includes support for the most of options in aplay, and simple unit tests for functionalities to handle data frames for files. At patch 17, transmission of data frame with blocking mode is supported. You can actually test playback/capture. * axfer: add a sub-command to transfer data frames At patch 24, transmission with non-blocking mode is supported. * axfer: add support for non-blocking operation At patch 25, transmission by MMAP operation is supported. * axfer: add support for MMAP PCM operation Between patch 30-36, some minor features are supported, but they includes own issues. I'd like to abandon them in future patchset. * axfer: add an option to support volume unit meter * axfer: add a unit test for vumeter calculation * axfer: add an option for formatted filename * axfer: add an option to handle key events * axfer: add a parser for channel map API * axfer: add a feature to generate a file for process id * axfer: obsolete some test options At patch 37/38, an integration to select waiter type is added. You can select one of poll(2) and epoll(2) to wait for I/O event notification. At present, I found some problems of this program: * When MMAP operation is used for PCM nodes with 'plug' plugin, XRUN occurs for capture and noisy sound for playback. I don't found this in the other operations such as blocking. * When using two axfer processes for capture/playback with pipe, some frames are truncated in the end of playback. I leave some tasks such as documentation and localization, but this patchset covers major use cases, I think. Regards Takashi Sakamoto (38): axfer: add an entry point for this command axfer: add a sub-command to print list of PCMs/devices axfer: add a common interface to handle a file with audio-specific data format axfer: add support for a container of Microsoft/IBM RIFF/Wave format axfer: add support for a container of Sparc AU format axfer: add support for a container of Creative Tech. voice format axfer: add support for a container of raw data axfer: add unit test for container interface axfer: add a common interface to align data frames on different layout axfer: add support for a mapper for single target axfer: add support for a mapper for multiple target axfer: add a unit test for mapper interface axfer: add a parser for command-line options axfer: add a common interface to transfer data frames axfer: add support to transfer data frames by alsa-lib PCM APIs axfer: add support for blocking data transmission operation of alsa-lib PCM API axfer: add a sub-command to transfer data frames axfer: add informative output and an option to suppress it axfer: add an option to dump available hardware parameters axfer: add options related to duration and obsolete '--max-file-size' option axfer: add an option to finish transmission at XRUN axfer: add a common interface of waiter for I/O event notification axfer: add support of waiter for poll(2) axfer: add support for non-blocking operation axfer: add support for MMAP PCM operation axfer: add an option to suppress event waiting axfer: add options for buffer arrangement axfer: add options for software parameters of PCM substream axfer: add options for plugins in alsa-lib axfer: add an option to support volume unit meter axfer: add a unit test for vumeter calculation axfer: add an option for formatted filename axfer: add an option to handle key events axfer: add a parser for channel map API axfer: add a feature to generate a file for process id axfer: obsolete some test options axfer: add an implementation of waiter for epoll(7) axfer: add an option for waiter type Makefile.am | 2 +- axfer/Makefile.am | 60 +++ axfer/container-au.c | 207 +++++++++ axfer/container-raw.c | 66 +++ axfer/container-riff-wave.c | 581 ++++++++++++++++++++++++ axfer/container-voc.c | 843 +++++++++++++++++++++++++++++++++++ axfer/container.c | 470 ++++++++++++++++++++ axfer/container.h | 126 ++++++ axfer/key-event.c | 121 +++++ axfer/key-event.h | 19 + axfer/main.c | 194 ++++++++ axfer/mapper-multiple.c | 271 +++++++++++ axfer/mapper-single.c | 195 ++++++++ axfer/mapper.c | 152 +++++++ axfer/mapper.h | 87 ++++ axfer/misc.h | 16 + axfer/options.c | 963 ++++++++++++++++++++++++++++++++++++++++ axfer/options.h | 83 ++++ axfer/subcmd-list.c | 234 ++++++++++ axfer/subcmd-transfer.c | 520 ++++++++++++++++++++++ axfer/subcmd.h | 18 + axfer/test/Makefile.am | 39 ++ axfer/test/container-test.c | 299 +++++++++++++ axfer/test/generator.c | 260 +++++++++++ axfer/test/generator.h | 47 ++ axfer/test/mapper-test.c | 491 ++++++++++++++++++++ axfer/test/vumeter-test.c | 128 ++++++ axfer/vumeter.c | 565 +++++++++++++++++++++++ axfer/vumeter.h | 56 +++ axfer/waiter-epoll.c | 81 ++++ axfer/waiter-poll.c | 68 +++ axfer/waiter.c | 84 ++++ axfer/waiter.h | 52 +++ axfer/xfer-libasound-irq-mmap.c | 224 ++++++++++ axfer/xfer-libasound-irq-rw.c | 518 +++++++++++++++++++++ axfer/xfer-libasound.c | 682 ++++++++++++++++++++++++++++ axfer/xfer-libasound.h | 57 +++ axfer/xfer.c | 191 ++++++++ axfer/xfer.h | 76 ++++ configure.ac | 3 +- 40 files changed, 9147 insertions(+), 2 deletions(-) create mode 100644 axfer/Makefile.am create mode 100644 axfer/container-au.c create mode 100644 axfer/container-raw.c create mode 100644 axfer/container-riff-wave.c create mode 100644 axfer/container-voc.c create mode 100644 axfer/container.c create mode 100644 axfer/container.h create mode 100644 axfer/key-event.c create mode 100644 axfer/key-event.h create mode 100644 axfer/main.c create mode 100644 axfer/mapper-multiple.c create mode 100644 axfer/mapper-single.c create mode 100644 axfer/mapper.c create mode 100644 axfer/mapper.h create mode 100644 axfer/misc.h create mode 100644 axfer/options.c create mode 100644 axfer/options.h create mode 100644 axfer/subcmd-list.c create mode 100644 axfer/subcmd-transfer.c create mode 100644 axfer/subcmd.h create mode 100644 axfer/test/Makefile.am create mode 100644 axfer/test/container-test.c create mode 100644 axfer/test/generator.c create mode 100644 axfer/test/generator.h create mode 100644 axfer/test/mapper-test.c create mode 100644 axfer/test/vumeter-test.c create mode 100644 axfer/vumeter.c create mode 100644 axfer/vumeter.h create mode 100644 axfer/waiter-epoll.c create mode 100644 axfer/waiter-poll.c create mode 100644 axfer/waiter.c create mode 100644 axfer/waiter.h create mode 100644 axfer/xfer-libasound-irq-mmap.c create mode 100644 axfer/xfer-libasound-irq-rw.c create mode 100644 axfer/xfer-libasound.c create mode 100644 axfer/xfer-libasound.h create mode 100644 axfer/xfer.c create mode 100644 axfer/xfer.h -- 2.11.0

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[alsa-devel] [PATCH] ASoC: Intel: Atom: Remove unneeded linux/miscdevice.h include
by Corentin Labbe 18 Sep '17

18 Sep '17

No file in sound/soc/intel/ use any miscdevice. This patch remove this uncessary include. Signed-off-by: Corentin Labbe <clabbe.montjoie(a)gmail.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 0e928d54305d..4166925a689b 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -23,7 +23,6 @@ #include <linux/interrupt.h> #include <linux/slab.h> #include <linux/io.h> -#include <linux/miscdevice.h> #include <linux/platform_device.h> #include <linux/firmware.h> #include <linux/pm_runtime.h> -- 2.13.5

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[alsa-devel] [PATCH] ALSA: oxygen: Xonar DG(X): make model_xonar_dg const
by Bhumika Goyal 18 Sep '17

18 Sep '17

Make this const as it not modified anywhere. It is only used during a copy operation. Also, add const to the declaration in header. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/oxygen/xonar_dg.h | 2 +- sound/pci/oxygen/xonar_dg_mixer.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/oxygen/xonar_dg.h b/sound/pci/oxygen/xonar_dg.h index d461df3..5a07cda 100644 --- a/sound/pci/oxygen/xonar_dg.h +++ b/sound/pci/oxygen/xonar_dg.h @@ -51,6 +51,6 @@ void dump_cs4245_registers(struct oxygen *chip, void dg_resume(struct oxygen *chip); void dg_cleanup(struct oxygen *chip); -extern struct oxygen_model model_xonar_dg; +extern const struct oxygen_model model_xonar_dg; #endif diff --git a/sound/pci/oxygen/xonar_dg_mixer.c b/sound/pci/oxygen/xonar_dg_mixer.c index b885dac..d22fbe8 100644 --- a/sound/pci/oxygen/xonar_dg_mixer.c +++ b/sound/pci/oxygen/xonar_dg_mixer.c @@ -449,7 +449,7 @@ static int dg_mixer_init(struct oxygen *chip) return 0; } -struct oxygen_model model_xonar_dg = { +const struct oxygen_model model_xonar_dg = { .longname = "C-Media Oxygen HD Audio", .chip = "CMI8786", .init = dg_init, -- 1.9.1

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[alsa-devel] [PATCH] ALSA: compress: Remove unused variable
by Guneshwor Singh 18 Sep '17

18 Sep '17

Commit 04c5d5a430fc ("ALSA: compress: Embed struct device") removed the statement that used 'str' but didn't remove the variable itself. So remove it. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/core/compress_offload.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index fec1dfdb14ad..4490a699030b 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -948,14 +948,13 @@ static const struct file_operations snd_compr_file_ops = { static int snd_compress_dev_register(struct snd_device *device) { int ret = -EINVAL; - char str[16]; struct snd_compr *compr; if (snd_BUG_ON(!device || !device->device_data)) return -EBADFD; compr = device->device_data; - pr_debug("reg %s for device %s, direction %d\n", str, compr->name, + pr_debug("reg device %s, direction %d\n", compr->name, compr->direction); /* register compressed device */ ret = snd_register_device(SNDRV_DEVICE_TYPE_COMPRESS, -- 2.14.1

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[alsa-devel] [PATCH] ALSA: hwdep: prevent a harmless shift wrapping bug
by Dan Carpenter 18 Sep '17

18 Sep '17

The "info.index" variable represents a bit in hw->dsp_loaded which is an unsigned int. If it's higher than 31 we hit a shift wrapping bug. This seems harmless, but I wanted to silence the static checker warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index a73baa1242be..8faae3d1455d 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -228,6 +228,8 @@ static int snd_hwdep_dsp_load(struct snd_hwdep *hw, memset(&info, 0, sizeof(info)); if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; + if (info.index >= 32) + return -EINVAL; /* check whether the dsp was already loaded */ if (hw->dsp_loaded & (1 << info.index)) return -EBUSY;

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[alsa-devel] [PATCH v2] pcm: softvol: add support for S24_LE
by Jörg Krause 18 Sep '17

18 Sep '17

Tested with the Wolfson WM8524 DAC on a i.MX6UL board and the following ALSA configuration file using the pcm test utility from alsa-lib: """ $ cat /etc/asound.conf pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "hw:0" } control { name "Master" card 0 } } ctl.!default { type hw card 0 } ctl.softvol { type hw card 0 } $ pcm -D softvol -o S24_LE -c 2 -r 48000 """ The data in the Synchronous Audio Interface (SAI) of the i.MX6UL is aligned the following way: """ 31 30 29 28 | 27 26 25 24 | 23 22 21 20 | .. | 3 2 1 0 ## ## ## ## ## ## ## ## [ DATA[23:0] ] """ Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- v2: * add CONVERT_AREA_S24_LE algorithm instead of using the same one as S32_LE --- src/pcm/pcm_softvol.c | 56 ++++++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 53 insertions(+), 3 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 1fe5784d..68a35b56 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -251,6 +251,44 @@ static inline short MULTI_DIV_short(short a, unsigned int b, int swap) } \ } \ } while (0) + +#define CONVERT_AREA_S24_LE() do { \ + unsigned int ch, fr; \ + int *src, *dst; \ + int tmp; \ + for (ch = 0; ch < channels; ch++) { \ + src_area = &src_areas[ch]; \ + dst_area = &dst_areas[ch]; \ + src = snd_pcm_channel_area_addr(src_area, src_offset); \ + dst = snd_pcm_channel_area_addr(dst_area, dst_offset); \ + src_step = snd_pcm_channel_area_step(src_area) \ + / sizeof(int); \ + dst_step = snd_pcm_channel_area_step(dst_area) \ + / sizeof(int); \ + GET_VOL_SCALE; \ + fr = frames; \ + if (! vol_scale) { \ + while (fr--) { \ + *dst = 0; \ + dst += dst_step; \ + } \ + } else if (vol_scale == 0xffff) { \ + while (fr--) { \ + *dst = *src; \ + src += dst_step; \ + dst += src_step; \ + } \ + } else { \ + while (fr--) { \ + tmp = *src << 8; \ + tmp = (signed int) tmp >> 8; \ + *dst = MULTI_DIV_24(tmp, vol_scale); \ + src += dst_step; \ + dst += src_step; \ + } \ + } \ + } \ +} while (0) #define GET_VOL_SCALE \ switch (ch) { \ @@ -315,6 +353,10 @@ static void softvol_convert_stereo_vol(snd_pcm_softvol_t *svol, CONVERT_AREA(int, !snd_pcm_format_cpu_endian(svol->sformat)); break; + case SND_PCM_FORMAT_S24_LE: + /* 24bit samples */ + CONVERT_AREA_S24_LE(); + break; case SND_PCM_FORMAT_S24_3LE: CONVERT_AREA_S24_3LE(); break; @@ -366,6 +408,10 @@ static void softvol_convert_mono_vol(snd_pcm_softvol_t *svol, CONVERT_AREA(int, !snd_pcm_format_cpu_endian(svol->sformat)); break; + case SND_PCM_FORMAT_S24_LE: + /* 24bit samples */ + CONVERT_AREA_S24_LE(); + break; case SND_PCM_FORMAT_S24_3LE: CONVERT_AREA_S24_3LE(); break; @@ -422,6 +468,7 @@ static int snd_pcm_softvol_hw_refine_cprepare(snd_pcm_t *pcm, { (1ULL << SND_PCM_FORMAT_S16_LE) | (1ULL << SND_PCM_FORMAT_S16_BE) | + (1ULL << SND_PCM_FORMAT_S24_LE) | (1ULL << SND_PCM_FORMAT_S32_LE) | (1ULL << SND_PCM_FORMAT_S32_BE), (1ULL << (SND_PCM_FORMAT_S24_3LE - 32)) @@ -577,10 +624,11 @@ static int snd_pcm_softvol_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * param if (slave->format != SND_PCM_FORMAT_S16_LE && slave->format != SND_PCM_FORMAT_S16_BE && slave->format != SND_PCM_FORMAT_S24_3LE && + slave->format != SND_PCM_FORMAT_S24_LE && slave->format != SND_PCM_FORMAT_S32_LE && slave->format != SND_PCM_FORMAT_S32_BE) { - SNDERR("softvol supports only S16_LE, S16_BE, S24_3LE, S32_LE " - " or S32_BE"); + SNDERR("softvol supports only S16_LE, S16_BE, S24_LE, S24_3LE, " + "S32_LE or S32_BE"); return -EINVAL; } svol->sformat = slave->format; @@ -863,6 +911,7 @@ int snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, sformat != SND_PCM_FORMAT_S16_LE && sformat != SND_PCM_FORMAT_S16_BE && sformat != SND_PCM_FORMAT_S24_3LE && + sformat != SND_PCM_FORMAT_S24_LE && sformat != SND_PCM_FORMAT_S32_LE && sformat != SND_PCM_FORMAT_S32_BE) return -EINVAL; @@ -1082,9 +1131,10 @@ int _snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, sformat != SND_PCM_FORMAT_S16_LE && sformat != SND_PCM_FORMAT_S16_BE && sformat != SND_PCM_FORMAT_S24_3LE && + sformat != SND_PCM_FORMAT_S24_LE && sformat != SND_PCM_FORMAT_S32_LE && sformat != SND_PCM_FORMAT_S32_BE) { - SNDERR("only S16_LE, S16_BE, S24_3LE, S32_LE or S32_BE format is supported"); + SNDERR("only S16_LE, S16_BE, S24_LE, S24_3LE, S32_LE or S32_BE format is supported"); snd_config_delete(sconf); return -EINVAL; } -- 2.14.1

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[alsa-devel] [RESENT PATCH v9 1/2] ASoC: rt5514: Add devicetree bindings for rt5514-spi
by Jeffy Chen 18 Sep '17

18 Sep '17

Add devicetree bindings documentation file for rt5514 spi dsp codec. Also update rt5514 i2c dt-binding's compatible to distinguish it from rt5514 spi. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- Changes in v9: Address comments from Brian. .../devicetree/bindings/sound/rt5514-spi.txt | 29 ++++++++++++++++++++++ Documentation/devicetree/bindings/sound/rt5514.txt | 4 +-- 2 files changed, 31 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5514-spi.txt diff --git a/Documentation/devicetree/bindings/sound/rt5514-spi.txt b/Documentation/devicetree/bindings/sound/rt5514-spi.txt new file mode 100644 index 000000000000..7911d1adbd29 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5514-spi.txt @@ -0,0 +1,29 @@ +RT5514 SPI audio CODEC + +This device supports SPI only. + +Required properties: + +- compatible : "realtek,rt5514" or "realtek,rt5514", "realtek,rt5514-spi". + +- reg : The SPI address of the device. + +Optional properties: + +- interrupt-parent: Should be the phandle for the interrupt controller +- interrupts: The interrupt number to the cpu. The interrupt specifier format + depends on the interrupt controller. + +Example: + +codec: rt5514-spi@0 { + compatible = "realtek,rt5514", "realtek,rt5514-spi"; + reg = <0>; + + pinctrl-names = "default"; + pinctrl-0 = <&mic_int>; + + interrupt-parent = <&gpio1>; + interrupts = <13 IRQ_TYPE_LEVEL_HIGH>; + wakeup-source; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt index 929ca6756b02..841a1159e234 100644 --- a/Documentation/devicetree/bindings/sound/rt5514.txt +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -4,7 +4,7 @@ This device supports I2C only. Required properties: -- compatible : "realtek,rt5514". +- compatible : "realtek,rt5514" or "realtek,rt5514", "realtek,rt5514-i2c". - reg : The I2C address of the device. @@ -28,6 +28,6 @@ Pins on the device (for linking into audio routes) for RT5514: Example: codec: rt5514@57 { - compatible = "realtek,rt5514"; + compatible = "realtek,rt5514", "realtek,rt5514-i2c"; reg = <0x57>; }; -- 2.11.0

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[alsa-devel] Problems with Chromebook Pixel 2015 audio
by Tom Rini 18 Sep '17

18 Sep '17

Hey folks, Entirely unrelated to the regression on this board that we dealt with the other week, I have (and have had for a bit), a problem with audio on this laptop. After a while of use, I'll have a problem like this: Sep 12 20:59:30 oliver kernel: haswell-pcm-audio haswell-pcm-audio: FW loaded, mailbox readback FW info: type 01, - version: 00.00, build 77, source commit id: 876ac6906f31a43b6772b23c7c983ce9dcb18a19 (everything is fine, audio works, is in use) Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: error: audio DSP boot timeout IPCD 0x0 IPCX 0x0 Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: ipc: --message timeout-- ipcx 0x83000000 isr 0x00000000 ipcd 0x00000000 imrx 0x7fff0000 Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: error: stream commit failed Sep 12 21:24:04 oliver kernel: System PCM: error: failed to commit stream -110 Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: ASoC: haswell-pcm-audio hw params failed: -110 Sep 12 21:24:04 oliver kernel: System PCM: ASoC: hw_params FE failed -110 With those last 4 messages repeating constantly (~3 sets per second, and the above are from v4.13, but I've had similar for as long as the driver has been mainline I think) and it only stops when I reboot. Rebooting may or may not clear the problem, last night I had to power off to get audio to come back up. I'm not quite sure where to begin debugging this issue. Does anyone have any suggestions? Thanks! -- Tom

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