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

[alsa-devel] [PATCH 1/2] ALSA: line6: use msecs_to_jiffies for conversion
by Nicholas Mc Guire 03 Feb '15

03 Feb '15

This is only an API consolidation and should make things more readable it replaces var * HZ / 1000 by msecs_to_jiffies(var). Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Converting milliseconds to jiffies by val * HZ / 1000 is technically not wrong but msecs_to_jiffies(val) is the cleaner solution and handles all corner cases correctly. This is a minor API cleanup only. Patch was only compile tested for x86_64_defconfig + CONFIG_SOUND=y, CONFIG_SND_USB_POD=m (implies CONFIG_SND_USB_LINE6=y) Patch is against 3.0.19-rc6 (localversion = -next-20150202) sound/usb/line6/driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 2328ec9..3cf12cc 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -166,7 +166,7 @@ void line6_start_timer(struct timer_list *timer, unsigned int msecs, void (*function)(unsigned long), unsigned long data) { setup_timer(timer, function, data); - mod_timer(timer, jiffies + msecs * HZ / 1000); + mod_timer(timer, jiffies + msecs_to_jiffies(msecs)); } EXPORT_SYMBOL_GPL(line6_start_timer); -- 1.7.10.4

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[alsa-devel] [PATCH v4 0/3] Sound support for Exynos4412 Trats2 board
by Inha Song 03 Feb '15

03 Feb '15

This patch-set adds basic sound support for the Trats2 boards. It just support primary I2s and external speaker playback. Changes in v4: - Move to clock out parent settings in exynos4412-trats2.dts - Remove unnecessary lines in Trats2 machine driver. - Change name of dai stream to more cleary. - Add MFD_WM8994 to select in machine driver kconfig. Changes in v3: - Set clock out parent to XUSBXTI in pmu_system_controller DT node. Changes in v2: - Remove unnecessary lines in Trats2 machine driver. - move clock enable/disable to .startup / .shutdown. - Fix code style issues. - Remove mclk DT property in Trats2 sound and add MCLK1 property in WM1811 node. (MCLK1 is optional property in WM8994 node - refer: Document/devicetree/bindings/sound/wm8994.txt) - Use mclk that defined in WM1811 DT node. - Add "samsung,audio-codec" property to required properties. Inha Song (3): ASoC: samsung: Add machine driver for Trats2 ASoC: samsung: Document Trats2 audio subsystem bindings ARM: dts: Add sound nodes for exynos4412-trats2 .../bindings/sound/samsung,trats2-wm1811.txt | 25 +++ arch/arm/boot/dts/exynos4412-trats2.dts | 42 ++++ sound/soc/samsung/Kconfig | 9 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/trats2_wm1811.c | 211 +++++++++++++++++++++ 5 files changed, 289 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,trats2-wm1811.txt create mode 100644 sound/soc/samsung/trats2_wm1811.c -- 2.0.0.390.gcb682f8

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[alsa-devel] [PATCH v2] ASoC: Add sysfs entries via static attribute groups
by Takashi Iwai 02 Feb '15

02 Feb '15

Instead of calling device_create_file() manually, assign the static attribute group entries at the device registration. This simplifies the error handling and avoids the possible races. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Enable attributes conditionally via is_visible callback include/sound/soc-dapm.h | 2 +- sound/soc/soc-core.c | 58 ++++++++++++++++++++++++++++-------------------- sound/soc/soc-dapm.c | 14 ++++-------- 3 files changed, 39 insertions(+), 35 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 89823cfe6f04..e8e0220404d4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -405,7 +405,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_update *update); /* dapm sys fs - used by the core */ -int snd_soc_dapm_sys_add(struct device *dev); +extern struct attribute *soc_dapm_dev_attrs[]; void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2c62620abca6..4bd4d77d4ba8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -191,6 +191,39 @@ static ssize_t pmdown_time_set(struct device *dev, static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); +static struct attribute *soc_dev_attrs[] = { + &dev_attr_codec_reg.attr, + &dev_attr_pmdown_time.attr, + NULL +}; + +static umode_t soc_dev_attr_is_visible(struct kobject *kobj, + struct attribute *attr, int idx) +{ + struct device *dev = kobj_to_dev(kobj); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + + if (attr == &dev_attr_pmdown_time.attr) + return attr->mode; /* always visible */ + return rtd->codec ? attr->mode : 0; /* enabled only with codec */ +} + +static const struct attribute_group soc_dapm_dev_group = { + .attrs = soc_dapm_dev_attrs, + .is_visible = soc_dev_attr_is_visible, +}; + +static const struct attribute_group soc_dev_roup = { + .attrs = soc_dev_attrs, + .is_visible = soc_dev_attr_is_visible, +}; + +static const struct attribute_group *soc_dev_attr_groups[] = { + &soc_dapm_dev_group, + &soc_dev_roup, + NULL +}; + #ifdef CONFIG_DEBUG_FS static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) @@ -949,8 +982,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(rtd->dev, &dev_attr_pmdown_time); - device_remove_file(rtd->dev, &dev_attr_codec_reg); device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -1120,6 +1151,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; + rtd->dev->groups = soc_dev_attr_groups; dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); @@ -1136,23 +1168,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, return ret; } rtd->dev_registered = 1; - - if (rtd->codec) { - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", - ret); - - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", - ret); - } - return 0; } @@ -1308,11 +1323,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } #endif - ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); - if (ret < 0) - dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n", - ret); - if (cpu_dai->driver->compress_dai) { /*create compress_device"*/ ret = soc_new_compress(rtd, num); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb1f982..6c5cf7d1ff54 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2127,15 +2127,10 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); -int snd_soc_dapm_sys_add(struct device *dev) -{ - return device_create_file(dev, &dev_attr_dapm_widget); -} - -static void snd_soc_dapm_sys_remove(struct device *dev) -{ - device_remove_file(dev, &dev_attr_dapm_widget); -} +struct attribute *soc_dapm_dev_attrs[] = { + &dev_attr_dapm_widget.attr, + NULL +}; static void dapm_free_path(struct snd_soc_dapm_path *path) { @@ -3903,7 +3898,6 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) */ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_sys_remove(dapm->dev); dapm_debugfs_cleanup(dapm); dapm_free_widgets(dapm); list_del(&dapm->list); -- 2.2.2

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[alsa-devel] [PATCH] ASoC: rx51: Correct DAPM route for capture
by Peter Ujfalusi 02 Feb '15

02 Feb '15

"Mic Bias" is DAPM_SUPPLY so it has to be connected in the route accordingly. Fixes audio capture on the board. Reported-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Tested-by: Pavel Machek <pavel(a)ucw.cz> --- sound/soc/omap/rx51.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 04896d6252a2..7f299357c2d2 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -250,14 +250,14 @@ static const struct snd_soc_dapm_route audio_map[] = { {"FM Transmitter", NULL, "LLOUT"}, {"FM Transmitter", NULL, "RLOUT"}, - {"DMic Rate 64", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "DMic"}, + {"DMic Rate 64", NULL, "DMic"}, + {"DMic", NULL, "Mic Bias"}, {"b LINE2R", NULL, "MONO_LOUT"}, {"Earphone", NULL, "b HPLOUT"}, - {"LINE1L", NULL, "b Mic Bias"}, - {"b Mic Bias", NULL, "HS Mic"} + {"LINE1L", NULL, "HS Mic"}, + {"HS Mic", NULL, "b Mic Bias"}, }; static const char * const spk_function[] = {"Off", "On"}; -- 2.2.2

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Re: [alsa-devel] Digital Input on ALC3661 (from laptop HDMI IN) not working under Linux
by Oliver Freyermuth 02 Feb '15

02 Feb '15

Am 02.02.2015 um 08:05 schrieb Raymond Yau: > >> >> >> I have an Alienware 17 laptop (sometimes called M17RX5) which has a >> > Realtek ALC3661. >> >>The problem I describe here should also be valid for the Alienware 18 >> > (all the late 2013 models, which are still the most current). At least >> > some scattered forum reports suggest so. >> > >> >>The card works almost perfect in Linux, but one functionality is >> > missing: The laptop features an HDMI input, and the Realtek card is >> > supposed to take care of the incoming digital audio. >> > This works fine in Windows, but in Linux, I only get the video and the >> > audio stays missing. >> > >> >> Here you can find my alsa-info: >> >>http://www.alsa-project.org/db/?f=0a0502e49212a0436a30fde9d4ee56163bbc2ebc > <http://www.alsa-project.org/db/?f=0a0502e49212a0436a30fde9d4ee56163bbc2ebc> >> > >> > The only digial input is node 0x0a >> > >> > Try hda-jack-retask to change node 0x1f with pin default [Jack] > digital In >> >> Thanks a lot for your suggestion! >> I tried, 0x0a stays as "not present" but I can select it in alsa-mixer >> as expected. Sadly, I can only record silence. >> Also setting Jack Detection to "Present" in the advanced overrides of >> hdajackretask did not help. > > http://git.alsa-project.org/?p=alsa-tools.git;a=blob;f=hdajackretask/README… > > You have to select "show unconnected pin" and "advanced override" > > Change the pin default of node 0x1f to 0x3c61130 [Jack] SPDIF In > > Set as boot default and reboot > > Digital capture device 1 will be created after you fix the pin default > of node 0x1f > > Turn on iec958 capture switch and try > > arecord -D IEC958:CARD=PCH -f dat -v test.wav Thanks a lot for your continued help! Here's what I did - the sad news, I still record only silence: 1. use hdajackretask as you explained, apply as boot default. 2. I got the following files: /etc/modprobe.d/hda-jack-retask.conf (below the automatic comment): options snd-hda-intel patch=hda-jack-retask.fw,hda-jack-retask.fw,hda-jack-retask.fw,hda-jack-retask.fw /lib64/firmware/hda-jack-retask.fw: [codec] 0x10ec0668 0x102805aa 0 [pincfg] 0x12 0x99a30140 0x14 0x90170110 0x15 0x0321101f 0x16 0x03211020 0x18 0x40030008 0x19 0x03a11030 0x1a 0x411111f0 0x1b 0x411111f0 0x1d 0x41000001 0x1e 0x411111f0 0x1f 0x03c61130 It contains exactly the line 0x1f 0x03c61130, so I hope I clicked correctly in the GUI. 3. After reboot, I see in dmesg (I assume this means it worked): [ 6.021374] snd_hda_intel 0000:00:1b.0: Applying patch firmware 'hda-jack-retask.fw' [ 6.021468] snd_hda_intel 0000:01:00.1: Disabling MSI [ 6.021471] snd_hda_intel 0000:01:00.1: Applying patch firmware 'hda-jack-retask.fw' [ 6.022749] snd_hda_intel 0000:00:1b.0: irq 29 for MSI/MSI-X 4. I activate S/PDIF capture in alsamixer (IEC958 is not shown there, but S/PDIF appeared new after the change). 5. arecord -L tells me at the end: iec958:CARD=PCH,DEV=0 HDA Intel PCH, ALC3661 Digital IEC958 (S/PDIF) Digital Audio Output 6. I run, as suggested: arecord -D iec958:CARD=PCH -f dat -v test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hooks PCM Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 Slave: Hardware PCM card 1 'HDA Intel PCH' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 But I get only silence, I tried to first attach the HDMI + switch to the video and start recording when everything is connected (i.e. when I would hear audio in Windows). I also tried to just record with the HDMI-cable connected and audio coming in (but not switched to video - in this case, I would hear nothing in Windows, but the audio should still arrive). Both cases, I record just silence. Any further ideas? Thanks a lot for your assistance! > >> > >> > Node 0x1f [Pin Complex] wcaps 0x400681: Stereo Digital >> > Pincap 0x00000020: IN >> > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear >> > Conn = 1/8, Color = Black >> > DefAssociation = 0xf, Sequence = 0x0 >> > Misc = NO_PRESENCE >> > Pin-ctls: 0x20: IN >> > Unsolicited: tag=00, enabled=0 >> > Power states: D0 D1 D2 D3 EPSS >> > >> >>In Windows, I observe the HDMI-IN audio routing to the speakers is only >> > done if the Realtek Audio Service is running. In Linux, I observe >> > keypress-events when I either switch manually to the HDMI-in signal > and also >> > if a new signal is detected (something (the UEFI?) injects a keypress >> > then). >> > I assume the service in Windows catches these events and changes >> > "something" in the Realtek Audio Settings. I see no special recording >> > device in Windows, nor any explicit HDMI-input setting, nor any related >> > setting in the driver. Dell links a standard Realtek driver in their >> > support section. >> >

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[alsa-devel] [PATCH 0/2] ASoC: davinci-mcasp: Fixes for 3.20
by Peter Ujfalusi 02 Feb '15

02 Feb '15

Hi, While doing the conversion of davinci DAI drivers to use edma-pcm I have noticed couple of issues with the driver regarding to old daVinci device support which resulted in the following two patch. Some SoC uses shared IRQ line for all McASP instances There are SoCs where a given McASP instance only supports DIT mode. Currently the arch code feeds dummy resource in place of RX (which is not available). With the second patch this is not going to be needed anymore. If it is not too late for 3.20.. regards, Peter --- Peter Ujfalusi (2): ASoC: davicni-mcasp: Mark the common irq line as shared ASoC: davinci-mcasp: Fix DIT only McASP instance support sound/soc/davinci/davinci-mcasp.c | 75 +++++++++++++++++++++------------------ 1 file changed, 41 insertions(+), 34 deletions(-) -- 2.2.2

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[alsa-devel] [PATCH 0/2] ASoC: davinci-evm: Remove not needed header files
by Peter Ujfalusi 02 Feb '15

02 Feb '15

Hi, Trivial changes to remove headers should not be included in the machine driver. The first patch was sent as part of the edma header cleanup series in November: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084734.h… http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084735.h… The series missed 3.19 and it seams to not going to make it to 3.20 either.. With these header files gone from the davinci-evm I will be able to send my davinci audio conversion from davinci-pcm to edma-pcm for 3.21. Mark: is it possible to take these for 3.20 still? Regards, Peter --- Peter Ujfalusi (2): ASoC: davinci-evm: Do not include edma headers ASoC: davinci-evm: Do not include davinci-pcm.h and davinci-i2s.h sound/soc/davinci/davinci-evm.c | 6 ------ 1 file changed, 6 deletions(-) -- 2.2.2

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[alsa-devel] [PATCH v4 00/10] audio timestamping evolutions
by Pierre-Louis Bossart 02 Feb '15

02 Feb '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields V3/4: Addressed feedback from Jaroslav: no change to STATUS ioctl, new functionality introduced in STATUS_EXT ioctl with r/w params. bumped PCM protocol for detection on STATUS_EXT in userspace used 32-bit word for accuracy report, no mantissa/exponent packing rolled back info_ext changes, all INFO fields remain in same word Merged comments from Liam (code simplifications) fixed packing Added driver_tstamp field in case there is a delay in passing the tstamp and audio_tstamp over IPC. Pierre-Louis Bossart (10): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info ALSA: bump PCM protocol to 2.0.13 Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 67 +++++++++- include/uapi/sound/asound.h | 36 +++++- sound/core/pcm_compat.c | 36 +++++- sound/core/pcm_lib.c | 88 ++++++++----- sound/core/pcm_native.c | 58 ++++++++- sound/pci/hda/hda_controller.c | 43 +++++-- sound/usb/pcm.c | 9 ++ 8 files changed, 478 insertions(+), 59 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

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[alsa-devel] [PATCH 00/14] Refactoring ALSA core device management
by Takashi Iwai 02 Feb '15

02 Feb '15

Hi, this is a series of relatively small patches for refactoring the ALSA core device management. I experimented a similar thing some time ago but didn't merge it in the end due to too intrusive changes. At this time, the approach is rather minimalistic: only the device registration part is replaced by embedding the struct device into each ALSA device object. To the outside of ALSA core, there shouldn't be many visible changes. The only possible breakage would be an access to some renamed / dropped fields by a downstream driver. But it must be easy to correct. Some patches in the series are merely cleanup patches that can be applied individually, but included just for completeness. The whole patches are found in test/snd-device branch of sound git tree, too. Takashi === Takashi Iwai (14): ALSA: Allow to pass the device object to snd_register_device*() ALSA: control: Provide a helper to look for the preferred subdevice ALSA: Add a helper to initialize device ALSA: control: Embed struct device ALSA: hwdep: Embed struct device ALSA: pcm: Embed struct device ALSA: rawmidi: Embed struct device ALSA: rawmidi: Use rawmidi device file for kernel messages ALSA: timer: Propagate the error at initialization ALSA: timer: Handle the device directly ALSA: seq: Handle the device directly ALSA: compress: Embed struct device ALSA: Simplify snd_device_register() variants ALSA: Drop snd_get_device() helper include/sound/compress_driver.h | 4 +- include/sound/control.h | 11 +++- include/sound/core.h | 41 ++---------- include/sound/hwdep.h | 3 +- include/sound/pcm.h | 2 +- include/sound/rawmidi.h | 4 +- sound/aoa/soundbus/i2sbus/pcm.c | 5 +- sound/core/compress_offload.c | 26 ++++++-- sound/core/control.c | 66 +++++++++++-------- sound/core/hwdep.c | 88 ++++++++++--------------- sound/core/init.c | 19 ++++++ sound/core/pcm.c | 70 ++++++-------------- sound/core/rawmidi.c | 47 +++++++------- sound/core/seq/seq_clientmgr.c | 14 +++- sound/core/sound.c | 112 ++++++++++---------------------- sound/core/timer.c | 42 ++++++++---- sound/pci/hda/hda_controller.c | 3 +- sound/pci/hda/hda_hwdep.c | 7 +- sound/soc/intel/sst-mfld-platform-pcm.c | 1 - 19 files changed, 265 insertions(+), 300 deletions(-) -- 2.2.2

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[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: Fix data delay configuration
by Peter Ujfalusi 02 Feb '15

02 Feb '15

The CTRLC register were not receiving the correct delay configuration, which will corrupt DSP_A audio mode. Fixes: 368494093354 ("ASoC: tlv320aic3x: Add TDM support") Reported-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Tested-by: Pavel Machek <pavel(a)ucw.cz> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- Changes since v1: Fixes tag and Tested-by from Pavel and Jarkko added to commit message. Regards, Peter sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a78ad361f494..07603d142923 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1046,7 +1046,7 @@ static int aic3x_prepare(struct snd_pcm_substream *substream, delay += aic3x->tdm_delay; /* Configure data delay */ - snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } -- 2.2.2

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