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

[alsa-devel] [PATCH] ASoC: atmel: ac97c: Remove unused variable
by Fabio Estevam 02 Apr '15

02 Apr '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Remove the unused 'match' variable to fix the following build warning: sound/atmel/ac97c.c:919:29: warning: unused variable 'match' [-Wunused-variable] Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/atmel/ac97c.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index cf4cedf..6dad042 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -916,7 +916,6 @@ static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) { struct ac97c_platform_data *pdata; struct device_node *node = dev->of_node; - const struct of_device_id *match; if (!node) { dev_err(dev, "Device does not have associated DT data\n"); -- 1.9.1

1 0

[alsa-devel] [RFC 0/11] dw_hdmi cleanups, audio preparation, helpers and ahb audio support
by Russell King - ARM Linux 02 Apr '15

02 Apr '15

I'm sending this series for comments, and to allow people to update their code bases (for those who are using my AHB audio driver on iMX6). This is really several series, and I'm not expecting this to be ready for the upcoming merge window. That said, if anyone wants to say that they're happy with some of the initial four patches, I'm happy to queue those for David, if he's willing to take them. This series applies on top of my previous series, which finished with the patch "drm: bridge/dw_hdmi: adjust n/cts setting order". The first four patches are a few cleanups to the dw_hdmi driver. The next two patches introduce new interfaces to the dw_hdmi driver to support AHB audio, including the errata found on iMX6 which requires N to be programmed to zero when the audio FIFO is not full. I'm expecting some discussion with these last two as we try to work out how to deal with the two variants of audio support for this part (AHB audio vs I2S audio). The following three add various helpers to the DRM and ALSA subsystems for audio support which is non-specific to the AHB audio driver. For DRM, this is a helper macro to obtain the pointer into the short audio descriptors. For ALSA, it's a set of helpers to restrict an audio PCM device's capabilities according to the ELD, and to generate the IEC958 channel status data. If these helpers are acceptable, we can convert a number of drivers to them. The ELD helper isn't quite finished, but should be sufficient for initial testing. The last two patches add support for the dw_hdmi audio. This is an ALSA driver rather than an ASoC driver as it's a pretty simple affair. I couldn't find a suitable location in the sound/ subtree to place it, so I've placed it next to the bridge - yes, we probably need to find it a better home in the sound/ subtree, but it currently makes use of dw_hdmi.h in the directory it lives. drivers/gpu/drm/bridge/Kconfig | 11 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 566 +++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.h | 14 + drivers/gpu/drm/bridge/dw_hdmi.c | 250 ++++++++----- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + include/drm/bridge/dw_hdmi.h | 7 + include/drm/drm_edid.h | 19 + include/sound/pcm_drm_eld.h | 6 + include/sound/pcm_iec958.h | 9 + sound/core/Kconfig | 6 + sound/core/Makefile | 3 + sound/core/pcm_drm_eld.c | 91 +++++ sound/core/pcm_iec958.c | 70 ++++ 14 files changed, 964 insertions(+), 92 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

7 29

[alsa-devel] [PATCH] ASoC: atmel-pcm-dma: increase buffer_bytes_max
by Alexandre Belloni 02 Apr '15

02 Apr '15

atmel-pcm-dma is not limited to a buffer size of 64kB like atmel-pcm-pdc. Increase buffer_bytes_max to 512kB to allow for higher bit rates (i.e. 32bps at 192kHz) to work correctly. By default, keep the prealloc at 64kB. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/atmel-pcm-dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index b8e7bad05eb1..b6625c8c411b 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -54,7 +54,7 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = { .period_bytes_max = 2 * 0xffff, /* if 2 bytes format */ .periods_min = 8, .periods_max = 1024, /* no limit */ - .buffer_bytes_max = ATMEL_SSC_DMABUF_SIZE, + .buffer_bytes_max = 512 * 1024, }; /** @@ -119,7 +119,7 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = { .prepare_slave_config = atmel_pcm_configure_dma, .pcm_hardware = &atmel_pcm_dma_hardware, - .prealloc_buffer_size = ATMEL_SSC_DMABUF_SIZE, + .prealloc_buffer_size = 64 * 1024, }; int atmel_pcm_dma_platform_register(struct device *dev) -- 2.1.0

3 3

[alsa-devel] ASoC: cs4271: init/timing problem
by Pascal Huerst 02 Apr '15

02 Apr '15

Hey all, We have a custom built hw, based on am335x and from time to time we need to rmmod/modprobe the ASoC machine driver. Depending on the hw, the first call to regmap_update_bits(..) in cs4271_codec_probe(..) fails with -EREMOTEIO. The error is originated in: drivers/i2c/busses/i2c-omap.c and happens if no i2c package acknowledge is received by the host. I think this is a timing issue and on some devices, the codec is just not ready yet, for communication. What is the right way to fix that? My attempt would be something like this: const int retries = 5; while (retries) { ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret == -EREMOTEIO) { retries--; udelay(1); continue; } else if (ret) return ret; break; } But I might also just add a proper delay before the first call to regmap_update_bits(..) So what is the proper way to handle this? Thanks Pascal

3 7

Re: [alsa-devel] Realtek ALC662 issues - Intel DG41CN
by Raymond Yau 02 Apr '15

02 Apr '15

> > >> You don't need any model since the pin defaults of your Intel DG41CN are correct unless you want to use CD audio in connector > > Originally, after the install there was no "model" there. It did not work even then. This entry that you see is because of my trials to make it work. You have to post output of alsa-info.sh without specify any model > > >>You should check the front audio panel of your computer chassis if jack state of headphone is alway plugged > Don't think that is the case. Because it works just fine on the Windows OS. In fact, as I have mentioned, there is some choppiness/stutter sound that I hear occasionally when I play in right now on Fedora/Ubuntu (while most of the time there is no sound). This behaviour goes away when I put the headphone (and then there is no sound at all from the main speaker) Do the front panel headphone still work when you use rear panel jacks for 5.1 sound in windows ? > > >>You have to provide pulseaudio verbose log > Pulseaudio is not a problem because I tested it by killing Pulseaudio too. The same problem persists. Though, if you insist I can surely get you the output. Killing pulseaudio won't work since it can auto-spwan http://lists.freedesktop.org/archives/pulseaudio-discuss/2015-March/023568.… Check the pulseaudio verbose log , as your line out and headphone share the same playback volume control, dB min of line out path and headphone path should be same value http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/m… <http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix…> http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… If Front is shared by HP and Line out, Why set volume to zero dB ? volume should remain unchanged , only need to restore save values after reboot > > >>You can disable multi channels by using early patching and hint multi_io=0 to get back the headphone playback volume contro > How do I do that? Appreciate your help here! > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… Codec: Realtek ALC662 rev1 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0662 Subsystem Id: 0x8086434e Revision Id: 0x100101 Early Patching ~~~~~~~~~~~~~~ When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a firmware file for modifying the HD-audio setup before initializing the codec. This can work basically like the reconfiguration via sysfs in the above, but it does it before the first codec configuration. A patch file is a plain text file which looks like below: ------------------------------------------------------------------------ [codec] 0x12345678 0xabcd1234 2 [hint] multi_io = no ------------------------------------------------------------------------ The file needs to have a line `[codec]`. The next line should contain three numbers indicating the codec vendor-id (0x12345678 in the example), the codec subsystem-id (0xabcd1234) and the address (2) of the codec. The rest patch entries are applied to this specified codec until another codec entry is given. Passing 0 or a negative number to the first or the second value will make the check of the corresponding field be skipped. It'll be useful for really broken devices that don't initialize SSID properly. The hd-audio driver reads the file via request_firmware(). Thus, a patch file has to be located on the appropriate firmware path, typically, /lib/firmware. For example, when you pass the option `patch=hda-init.fw`, the file /lib/firmware/hda-init.fw must be present Post output of alsa-info.sh after you reboot with early patching You will get headphone playback volume control but loss those surround , center and lfe playback volume controls http://www.intel.com/support/motherboards/desktop/sb/CS-034206.htm Seem your motherboard does not support multistreaming as it is not in the supported list Do this mean that indep_hp should be explicitly disabled for alc662 ?

1 0

Re: [alsa-devel] [PATCH] fix dig out type to SPDIF when there are two digital pins with different type - VT1708S lost SPDIF device after upgrade
by Raymond Yau 02 Apr '15

02 Apr '15

> > > > Thank you for your reply, > > > > Please see below the output of alsa-info.sh before and after the headphone plugged in > > > > Before: http://www.alsa-project.org/db/?f=a316eebf7b150b5897838be37bce4e8c9d95eaae > > After: http://www.alsa-project.org/db/?f=0f163003e9005972db5f97f21a45be67816e3801 > > > > I have 6 Jacks at the rear panel and two at the top (one output one input) > > I am afraid I do not know how to switch channel mode to retask. > > > > Also, the computer is of a certain age, and the issue has started several month ago but I never really looked into it, so I am not sure that change in power management will be relevant has it seems quite recent. > > What is the model of your asus motherbroad ? > > Seem lost the digial device 1 for SPDIF out > > Pin Default 0x985601f0: [Fixed] Digital Out at Int HDMI > > Is this internal spdif connector ? > > Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 > Device: name="VT1708S Digital", type="HDMI", device=3 > > APLAY > > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: VT1708S Analog [VT1708S Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: Intel [HDA Intel], device 3: VT1708S Digital [VT1708S Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > > Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital > Control: name="IEC958 Playback Con Mask", index=0, device=0 > Control: name="IEC958 Playback Pro Mask", index=0, device=0 > Control: name="IEC958 Playback Default", index=0, device=0 > Control: name="IEC958 Playback Switch", index=0, device=0 > Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 > Device: name="VT1708S Digital", type="HDMI", device=3 > Converter: stream=5, channel=0 > Digital: > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > > Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=5, channel=0 > Digital: > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > > Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital > Control: name="HDMI Phantom Jack", index=0, device=0 > Pincap 0x00000010: OUT > Pin Default 0x985601f0: [Fixed] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > Connection: 1 > 0x12 > Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital > Control: name="SPDIF Phantom Jack", index=0, device=0 > Pincap 0x00000010: OUT > Pin Default 0x074511f0: [Jack] SPDIF Out at Ext Rear Panel > Conn = Optical, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > Connection: 1 > 0x15 Do the driver need a pin fixup for the digital pin or more check when assign dig out type ? If codec has two digital pin complex with diffferent type, check whether the primary digital pin complex support DETECT case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) { codec_info(codec, "ignore pin 0x%x, too many assigned pins\n", nid); continue; } cfg->dig_out_pins[cfg->dig_outs] = nid; cfg->dig_out_type[cfg->dig_outs] = (loc == AC_JACK_LOC_HDMI) ? HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF; cfg->dig_outs++; + if (cfg->dig_outs == 2) + if (cfg->dig_out_type[0] != cfg->dig_out_type[1]) + if (! snd_hda_query_pin_caps(codec, cfg->dig_out_pins[0]) & AC_PINCAP_PRES_DETECT)) + cfg-> dig_out_type[0] = HDA_PCM_TYPE_SPDIF; break;

1 0

[alsa-devel] [PATCH v2 0/5] Intel SST driver restructure
by Jie Yang 02 Apr '15

02 Apr '15

In order to prepare for the upstreaming of Skylake drivers and to allow reuse of code between different Intel audio DSP products, this patch series restructures the Intel directory as follows: 1. Create "common" directory. Move common files into this directory. 2. Create "boards" directory. Move all machine drivers into this directory. 3. Create "haswell" directory. Move all remaining hsw/bdw code into this directory. 4. Create "baytrail" directory. Move remaining baytrail driver to this directory. 5. Create "atom" directory. Move all MCG platform drivers(including sst/ folder) to this directory. Changes in v2: add "Reviewed-by" and "Acked-by" items from each platforms' maintainer. Todo next step: 1. Change and move atom related codes(in atom/sst/) to corresponding and try share as many as possible codes(e.g. sst_acpi.c). 2. Share more common codes about IPC for different platforms, e.g. The methods for queueing and sending/reciving Tx/Rx messages, The methods for mailbox IO, The IPC error return codes, etc. Jie Yang (5): ASoC: Intel: create common folder and move common files in ASoC: Intel: create haswell folder and move haswell platform files in ASoC: Intel: create boards folder and move sst boards files in ASoC: Intel: create baytrail folder and move baytrail platform files in ASoC: Intel: create atom folder and move atom platform files in sound/soc/intel/Makefile | 42 +++------------------- sound/soc/intel/atom/Makefile | 7 ++++ sound/soc/intel/{ => atom}/sst-atom-controls.c | 0 sound/soc/intel/{ => atom}/sst-atom-controls.h | 0 sound/soc/intel/{ => atom}/sst-mfld-dsp.h | 0 .../intel/{ => atom}/sst-mfld-platform-compress.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform.h | 0 sound/soc/intel/{ => atom}/sst/Makefile | 0 sound/soc/intel/{ => atom}/sst/sst.c | 0 sound/soc/intel/{ => atom}/sst/sst.h | 0 sound/soc/intel/{ => atom}/sst/sst_acpi.c | 0 sound/soc/intel/{ => atom}/sst/sst_drv_interface.c | 0 sound/soc/intel/{ => atom}/sst/sst_ipc.c | 0 sound/soc/intel/{ => atom}/sst/sst_loader.c | 0 sound/soc/intel/{ => atom}/sst/sst_pci.c | 0 sound/soc/intel/{ => atom}/sst/sst_pvt.c | 0 sound/soc/intel/{ => atom}/sst/sst_stream.c | 0 sound/soc/intel/baytrail/Makefile | 4 +++ sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h | 0 sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c | 4 +-- sound/soc/intel/boards/Makefile | 15 ++++++++ sound/soc/intel/{ => boards}/broadwell.c | 6 ++-- sound/soc/intel/{ => boards}/byt-max98090.c | 2 +- sound/soc/intel/{ => boards}/byt-rt5640.c | 4 +-- .../{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5645.c | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5672.c | 0 sound/soc/intel/{ => boards}/haswell.c | 6 ++-- sound/soc/intel/{ => boards}/mfld_machine.c | 0 sound/soc/intel/common/Makefile | 6 ++++ sound/soc/intel/{ => common}/sst-acpi.c | 0 sound/soc/intel/{ => common}/sst-dsp-priv.h | 0 sound/soc/intel/{ => common}/sst-dsp.c | 0 sound/soc/intel/{ => common}/sst-dsp.h | 0 sound/soc/intel/{ => common}/sst-firmware.c | 0 sound/soc/intel/haswell/Makefile | 4 +++ sound/soc/intel/{ => haswell}/sst-haswell-dsp.c | 6 ++-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.c | 4 +-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.h | 0 sound/soc/intel/{ => haswell}/sst-haswell-pcm.c | 6 ++-- 43 files changed, 64 insertions(+), 60 deletions(-) create mode 100644 sound/soc/intel/atom/Makefile rename sound/soc/intel/{ => atom}/sst-atom-controls.c (100%) rename sound/soc/intel/{ => atom}/sst-atom-controls.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-dsp.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-compress.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform.h (100%) rename sound/soc/intel/{ => atom}/sst/Makefile (100%) rename sound/soc/intel/{ => atom}/sst/sst.c (100%) rename sound/soc/intel/{ => atom}/sst/sst.h (100%) rename sound/soc/intel/{ => atom}/sst/sst_acpi.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_drv_interface.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_ipc.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_loader.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pci.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pvt.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_stream.c (100%) create mode 100644 sound/soc/intel/baytrail/Makefile rename sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h (100%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c (99%) create mode 100644 sound/soc/intel/boards/Makefile rename sound/soc/intel/{ => boards}/broadwell.c (98%) rename sound/soc/intel/{ => boards}/byt-max98090.c (99%) rename sound/soc/intel/{ => boards}/byt-rt5640.c (98%) rename sound/soc/intel/{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5645.c (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5672.c (100%) rename sound/soc/intel/{ => boards}/haswell.c (98%) rename sound/soc/intel/{ => boards}/mfld_machine.c (100%) create mode 100644 sound/soc/intel/common/Makefile rename sound/soc/intel/{ => common}/sst-acpi.c (100%) rename sound/soc/intel/{ => common}/sst-dsp-priv.h (100%) rename sound/soc/intel/{ => common}/sst-dsp.c (100%) rename sound/soc/intel/{ => common}/sst-dsp.h (100%) rename sound/soc/intel/{ => common}/sst-firmware.c (100%) create mode 100644 sound/soc/intel/haswell/Makefile rename sound/soc/intel/{ => haswell}/sst-haswell-dsp.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.h (100%) rename sound/soc/intel/{ => haswell}/sst-haswell-pcm.c (99%) -- 1.9.1

2 7

Re: [alsa-devel] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Dan Carpenter 01 Apr '15

01 Apr '15

Hello Jyri Sarha, The patch a75a053f1eef: "ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used" from Mar 20, 2015, leads to the following Smatch warning: sound/soc/davinci/davinci-mcasp.c:1152 davinci_mcasp_startup() error: buffer overflow 'mcasp->ruledata' 2 <= 2 sound/soc/davinci/davinci-mcasp.c 1138 1139 /* 1140 * Limit the maximum allowed channels for the first stream: 1141 * number of serializers for the direction * tdm slots per serializer 1142 */ 1143 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) 1144 dir = TX_MODE; 1145 else 1146 dir = RX_MODE; dir is either 1 or 2. 1147 1148 for (i = 0; i < mcasp->num_serializer; i++) { 1149 if (mcasp->serial_dir[i] == dir) 1150 max_channels++; 1151 } 1152 mcasp->ruledata[dir].serializers = max_channels; ^^^^^^^^^^^^^ ->ruledata[] has only two elements so ->ruledata[2] is beyond the end of the array. 1153 max_channels *= mcasp->tdm_slots; regards, dan carpenter

3 2

[alsa-devel] [PATCH] ASoC: omap-hdmi-audio: No not use IEC958_AES1_PRO_MODE_NOTID
by Jyri Sarha 01 Apr '15

01 Apr '15

No IEC958_AES?_PRO_* macros should be used in HDMI consumer audio mode and IEC958_AES1_PRO_MODE_NOTID should be applied to byte 1 when applicable. However IEC958_AES1_PRO_MODE_NOTID is defined as 0 so this fix does not affect the functionality in any way. Reported-by: Russell King <linux(a)arm.linux.org.uk> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- When I refactored the OMAP HDMI audio functionality I intentionally wanted to keep all the HW bits identical to the earlier implementation so that I could compare the register values of the old and the new implementations one to one. One of these days I should go trough and check all the bits now that new implementation has settled in. Best regards, Jyri sound/soc/omap/omap-hdmi-audio.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index f7eb42a..4775da4 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -142,8 +142,6 @@ static int hdmi_dai_hw_params(struct snd_pcm_substream *substream, iec->status[0] |= IEC958_AES0_CON_EMPHASIS_NONE; - iec->status[0] |= IEC958_AES1_PRO_MODE_NOTID; - iec->status[1] = IEC958_AES1_CON_GENERAL; iec->status[2] |= IEC958_AES2_CON_SOURCE_UNSPEC; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/8] ASoC: wm8971: Use system_power_efficient_wq instead of custom workqueue
by Lars-Peter Clausen 01 Apr '15

01 Apr '15

The delayed work used by the wm8971 driver to manage the caps charging doesn't have any special requirements that would justify using a custom workqueue, just use the generic system_power_efficient_wq instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8971.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 39ddb9b..44baacd 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -31,8 +31,6 @@ #define WM8971_REG_COUNT 43 -static struct workqueue_struct *wm8971_workq = NULL; - /* codec private data */ struct wm8971_priv { unsigned int sysclk; @@ -636,7 +634,8 @@ static int wm8971_resume(struct snd_soc_codec *codec) reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_ON; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, msecs_to_jiffies(1000)); } @@ -649,9 +648,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) u16 reg; INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work); - wm8971_workq = create_workqueue("wm8971"); - if (wm8971_workq == NULL) - return -ENOMEM; wm8971_reset(codec); @@ -659,7 +655,8 @@ static int wm8971_probe(struct snd_soc_codec *codec) reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ @@ -681,8 +678,6 @@ static int wm8971_remove(struct snd_soc_codec *codec) { wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (wm8971_workq) - destroy_workqueue(wm8971_workq); return 0; } -- 1.8.0

3 9