August 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write
by George Anchev 27 Aug '18

27 Aug '18

Hi, Each time I boot in journalctl I notice the lines which you can see in the attached journal.txt file. Attaching also the output of alsa-info.sh. Both files are compressed to minimize message size. Please let me know if this is an actual bug and what is causing it. P. S. The "Bug Tracker" link on project's home page shows a "not found" page. -- George

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[alsa-devel] snd_usb_usx2y holding onto a lock?
by Joseph Spencer 27 Aug '18

27 Aug '18

When plugging an audio interface in with nomodeset I see the following in the logs: Aug 08 23:19:23 localhost.localdomain kernel: INFO: task pulseaudio:1750 blocked for more than 120 seconds. Aug 08 23:19:23 localhost.localdomain kernel: Tainted: G OE 4.17.11-200.fc28.x86_64+debug #1 Aug 08 23:19:23 localhost.localdomain kernel: "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. Aug 08 23:19:23 localhost.localdomain kernel: pulseaudio D12408 1750 1268 0x00000000 Aug 08 23:19:23 localhost.localdomain kernel: Call Trace: Aug 08 23:19:23 localhost.localdomain kernel: ? __schedule+0x2ba/0xaf0 Aug 08 23:19:23 localhost.localdomain kernel: ? prepare_to_wait_event+0xd2/0x180 Aug 08 23:19:23 localhost.localdomain kernel: schedule+0x2f/0x90 Aug 08 23:19:23 localhost.localdomain kernel: usX2Y_urbs_start+0x208/0x470 [snd_usb_usx2y] Aug 08 23:19:23 localhost.localdomain kernel: ? finish_wait+0x80/0x80 Aug 08 23:19:23 localhost.localdomain kernel: snd_usX2Y_pcm_prepare+0xb4/0xd0 [snd_usb_usx2y] Aug 08 23:19:23 localhost.localdomain kernel: snd_pcm_do_prepare+0x19/0x30 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: snd_pcm_action_single+0x36/0x80 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: snd_pcm_action_nonatomic+0x61/0x70 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: snd_pcm_common_ioctl+0x3d2/0xb70 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: ? selinux_file_ioctl+0x171/0x210 Aug 08 23:19:23 localhost.localdomain kernel: snd_pcm_ioctl+0x23/0x30 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: do_vfs_ioctl+0xa5/0x6d0 Aug 08 23:19:23 localhost.localdomain kernel: ksys_ioctl+0x60/0x90 Aug 08 23:19:23 localhost.localdomain kernel: __x64_sys_ioctl+0x16/0x20 Aug 08 23:19:23 localhost.localdomain kernel: do_syscall_64+0x60/0x1f0 Aug 08 23:19:23 localhost.localdomain kernel: entry_SYSCALL_64_after_hwframe+0x49/0xbe Aug 08 23:19:23 localhost.localdomain kernel: RIP: 0033:0x7f59021c4ec7 Aug 08 23:19:23 localhost.localdomain kernel: RSP: 002b:00007ffd83d81448 EFLAGS: 00000206 ORIG_RAX: 0000000000000010 Aug 08 23:19:23 localhost.localdomain kernel: RAX: ffffffffffffffda RBX: 0000556d69312220 RCX: 00007f59021c4ec7 Aug 08 23:19:23 localhost.localdomain kernel: RDX: 0000556d6931ae20 RSI: 0000000000004140 RDI: 0000000000000016 Aug 08 23:19:23 localhost.localdomain kernel: RBP: 00007ffd83d81b60 R08: 0000000000000000 R09: 0000000000001138 Aug 08 23:19:23 localhost.localdomain kernel: R10: 0000000000000004 R11: 0000000000000206 R12: 00007ffd83d81830 Aug 08 23:19:23 localhost.localdomain kernel: R13: 00007ffd83d81cf4 R14: 0000000000000000 R15: 00007ffd83d815c0 Aug 08 23:19:23 localhost.localdomain kernel: Showing all locks held in the system: Aug 08 23:19:23 localhost.localdomain kernel: 1 lock held by khungtaskd/38: Aug 08 23:19:23 localhost.localdomain kernel: #0: 00000000b04056c7 (tasklist_lock){.+.+}, at: debug_show_all_locks+0x3e/0x165 Aug 08 23:19:23 localhost.localdomain kernel: 2 locks held by pulseaudio/1750: Aug 08 23:19:23 localhost.localdomain kernel: #0: 000000008d30dafb (snd_pcm_link_rwsem){++++}, at: snd_pcm_action_nonatomic+0x1e/0x70 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: #1: 00000000c8fc4ab6 (&usX2Y(card)->pcm_mutex){+.+.}, at: snd_usX2Y_pcm_prepare+0x38/0xd0 [snd_usb_usx2y] The audio interface doesn't work and without nomodeset X freezes. Any help pointing me in the right direction would be greatly appreciated. It looks like an issue with the snd_usb_usx2y driver to me, but I'm not sure. Joe Spencer

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[alsa-devel] [PATCH] ALSA: pcm: signedness bug in snd_pcm_plug_alloc()
by Dan Carpenter 27 Aug '18

27 Aug '18

The "frames" variable is unsigned so the error handling doesn't work properly. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 0391cb1a4f19..141c5f3a9575 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->next) { if (plugin->dst_frames) frames = plugin->dst_frames(plugin, frames); - if (snd_BUG_ON(frames <= 0)) + if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) return -ENXIO; plugin = plugin->next; err = snd_pcm_plugin_alloc(plugin, frames); @@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->prev) { if (plugin->src_frames) frames = plugin->src_frames(plugin, frames); - if (snd_BUG_ON(frames <= 0)) + if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) return -ENXIO; plugin = plugin->prev; err = snd_pcm_plugin_alloc(plugin, frames);

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[alsa-devel] [PATCH] ALSA: hdspm: fix spelling mistake "Initializeing" -> "Initializing"
by Colin King 27 Aug '18

27 Aug '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_dbg message and also remove extraneous white space and repeated question marks. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 11b5b5e0e058..679ad0415e3b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6534,7 +6534,7 @@ static int snd_hdspm_create_alsa_devices(struct snd_card *card, dev_dbg(card->dev, "Update mixer controls...\n"); hdspm_update_simple_mixer_controls(hdspm); - dev_dbg(card->dev, "Initializeing complete ???\n"); + dev_dbg(card->dev, "Initializing complete?\n"); err = snd_card_register(card); if (err < 0) { -- 2.17.1

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[alsa-devel] [PATCH 0/8] ALSA: non-cached memalloc helper
by Takashi Iwai 27 Aug '18

27 Aug '18

Hi, this is a patchset to add the non-cached page handling in the memory allocation helpers, which have been implemented individually in several driver codes. This also fixes a long-time bug of HD-audio non-mmap behavior in the non-snoop mode, which was reported recently. Takashi === Takashi Iwai (8): ALSA: memalloc: Don't align the size to power-of-two ALSA: memalloc: Simplify snd_malloc_dev_pages() calls ALSA: memalloc: Add non-cached buffer type ALSA: hda: Check the non-cached stream buffers more explicitly ALSA: hda: Use new non-cached allocation for non-snoop mode ALSA: hda: Remove substream allocation/free ops ALSA: intel8x0: Use the new non-cached allocation for 440MX workaround ALSA: intel_hdmi: Use the new non-cached allocation include/sound/memalloc.h | 3 + sound/core/memalloc.c | 40 ++++++++----- sound/core/sgbuf.c | 15 ++++- sound/pci/hda/hda_controller.c | 13 +++-- sound/pci/hda/hda_controller.h | 7 +-- sound/pci/hda/hda_intel.c | 103 ++++----------------------------- sound/pci/hda/hda_tegra.c | 18 +----- sound/pci/intel8x0.c | 89 ++++------------------------ sound/x86/intel_hdmi_audio.c | 29 +--------- 9 files changed, 75 insertions(+), 242 deletions(-) -- 2.18.0

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[alsa-devel] [PATCH v3] ALSA: pcm: Return 0 when size < start_threshold in capture
by Ricardo Biehl Pasquali 26 Aug '18

26 Aug '18

In __snd_pcm_lib_xfer(), when capture, if state is PREPARED and size is less than start_threshold nothing can be done. As there is no error, 0 is returned. Signed-off-by: Ricardo Biehl Pasquali <pasqualirb(a)gmail.com> --- version 3: Set 'err' variable inside 'else' block. --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2173,11 +2173,16 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(str goto _end_unlock; if (!is_playback && - runtime->status->state == SNDRV_PCM_STATE_PREPARED && - size >= runtime->start_threshold) { - err = snd_pcm_start(substream); - if (err < 0) + runtime->status->state == SNDRV_PCM_STATE_PREPARED) { + if (size >= runtime->start_threshold) { + err = snd_pcm_start(substream); + if (err < 0) + goto _end_unlock; + } else { + /* nothing to do */ + err = 0; goto _end_unlock; + } } runtime->twake = runtime->control->avail_min ? : 1;

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[alsa-devel] [PATCH 2/2] ASoC: rt5682: Change DAC/ADC volume scale
by shumingf＠realtek.com 24 Aug '18

24 Aug '18

From: Shuming Fan <shumingf(a)realtek.com> The step of DAC/ADC volume scale changes from 0.375dB to 0.75dB Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 0ea2759..731c6a8 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -750,8 +750,8 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ @@ -1114,7 +1114,7 @@ static const struct snd_kcontrol_new rt5682_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, - RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv), /* IN Boost Volume */ SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL, @@ -1124,7 +1124,7 @@ static const struct snd_kcontrol_new rt5682_snd_controls[] = { SOC_DOUBLE("STO1 ADC Capture Switch", RT5682_STO1_ADC_DIG_VOL, RT5682_L_MUTE_SFT, RT5682_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682_STO1_ADC_DIG_VOL, - RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 127, 0, adc_vol_tlv), + RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 63, 0, adc_vol_tlv), /* ADC Boost Volume Control */ SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682_STO1_ADC_BOOST, -- 2.7.4

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[alsa-devel] [PATCH] ASoC: max98373: Added 10ms delay after amp software reset
by Ryan Lee 24 Aug '18

24 Aug '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes : Applied 10ms delay after amp software reset. 10ms guard time is required for stability. sound/soc/codecs/max98373.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 2764fae..6aaff03 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -730,6 +730,7 @@ static int max98373_probe(struct snd_soc_component *component) /* Software Reset */ regmap_write(max98373->regmap, MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); + mdelay(10); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,6 +819,7 @@ static int max98373_resume(struct device *dev) regmap_write(max98373->regmap, MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); + mdelay(10); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.7.4

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Re: [alsa-devel] [RFC] hda/realtek filtered headphone output
by Takashi Iwai 23 Aug '18

23 Aug '18

[ corrected to the right ML ] On Thu, 23 Aug 2018 14:37:39 +0200, René Rebe wrote: > > Hi, > > On 23 Aug 2018, at 14:31, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Thu, 23 Aug 2018 13:10:39 +0200, > > René Rebe wrote: > >> > >> Hi Takashi, > >> > >> I uploaded the requested log but did not hear back. > >> The limited headphone frequency response (aka high-pass) is still an issue with 4.18. > >> I rebased the patch: > >> > >> https://svn.exactcode.de/t2/trunk/package/base/linux/hda-realtek.patch > >> > >> This affects more devices (e.g. Dell XPS 13, ThinkPads) any maybe many more which I do not have at hand. > >> > >> Further guidance how to proceed with this issue to get it fixed upstream is welcome. > > > > As mentioned, a better way is to define the preferred mapping table. > > Take a look at alc274_fixup_bind_dacs() as an example > > Did you had a chance to look at the log I uploaded to get an idea if it is possible to improve this auto parser to figure out the connection? Did you take a look at the function? It's pretty straightforward, you can specify the mapping you'd need there. Takashi > > René > > > Takashi > > > >> > >> René > >> > >> On 06 Mar 2018, at 16:07, René Rebe <rene(a)exactcode.de> wrote: > >> > >>> Hi Takashi, > >>> > >>> On 03/05/2018 12:55 PM, Takashi Iwai wrote: > >>>> On Sat, 03 Mar 2018 13:06:49 +0100, > >>>> René Rebe wrote: > >>>>> > >>>>> Hi all, > >>>>> > >>>>> testing some new laptops, I noticed the headphone frequency response was lacking > >>>>> bass. First I noticed this in the Lenovo T470s, and later I found this also be the case > >>>>> on Surface Pro 2 and 3. The Dell XPS13 may also be affected, but I had to return it > >>>>> before I found out the solution on the other machines: > >>>>> > >>>>> https://www.youtube.com/watch?v=a1YkPtfC4LI > >>>>> > >>>>> I had not yet sent the patch because a) I was too busy and b) I wanted to review this further. > >>>>> > >>>>> As the other day I got a Lenovo ideapad 720s to test with exactly the same issue, here is the > >>>>> current hacked patch: > >>>>> > >>>>> https://svn.exactcode.de/t2/trunk/package/base/linux/surface-audio.patch > >>>>> > >>>>> However, given that all newer realtek codecs that came on my desk are affected, > >>>>> maybe there is something wrong with the hda_auto_parser.c - I did not yet fully grasp > >>>>> all the logic of that. > >>>>> > >>>>> As you can see in the patch and video (…) the problem is the signal routing of NID 0x15 > >>>>> or 0x21 (depending whether it is an older, or newer codec). This is the headphone out, > >>>>> and as far as I can see the auto_parser assigned the index 0 pcm output (NID 0x0c) > >>>>> which result in this filtered frequency response, likely intended for the laptop’s speakers? > >>>>> Using the next NID (I guess that was 0xd?) results in a full frequency response and at > >>>>> least to my ears is immediately hearable and obvious. > >>>>> > >>>>> Without this fixed routing the headphone sounds like a PC speaker in a coke can. > >>>>> > >>>>> So the question for the authors and HDA experts: Can this be fixed in the auto_parser, > >>>>> or is my patch (which btw. has to small unrelated cleanups) modulo potentially some > >>>>> renaming of the enum and function, now that more than the Surface is affected, the > >>>>> way to go? If this can be fixed more generally I would be happy to hear! > >>>> Could you give the output of alsa-info.sh? We need to check the > >>>> details of the codec topology. > >>> > >>> To avoid sending the huge output to the list, I pressed the upload button, and it gave me this Url: > >>> > >>> > >>> http://www.alsa-project.org/db/?f=c68cad9c06aa999f017f1e43b52448d778d2903f > >>> > >>> This is with the linked patch, if you want to see it without the change, I need to re-build my kernel again :-/ > >>> > >>>> In general, the right approach is to fix at parsing the routing, not > >>>> fiddling with the codec routing dynamically at initialization. > >>>> Currently the easy fix would be to provide the fixed DAC connection > >>>> list. Maybe we can enhance the generic parser to receive some hints > >>>> for DAC selection. The routes are determined by calculating "badness" > >>>> values per route, so in theory, we can give some badness values > >>>> depending on the target pin type (headphone, speaker, etc). > >>> > >>> Let me know if I should try some alternative solution, ... > >>> > >>> René > >>> > >>> -- > >>> René Rebe, ExactCODE GmbH, Lietzenburger Str. 42, DE-10789 Berlin > >>> https://exactcode.com | https://t2sde.org | https://rene.rebe.de > >> > >> -- > >> ExactCODE GmbH, Lietzenburger Str. 42, DE-10789 Berlin > >> DE Legal: Amtsgericht Berlin (Charlottenburg) HRB 105123B, Tax-ID#: DE251602478 > >> Managing Director: René Rebe > >> http://exactcode.com | http://exactscan.com | http://ocrkit.com | http://t2-project.org | http://rene.rebe.de > >> > > -- > ExactCODE GmbH, Lietzenburger Str. 42, DE-10789 Berlin > DE Legal: Amtsgericht Berlin (Charlottenburg) HRB 105123B, Tax-ID#: DE251602478 > Managing Director: René Rebe > http://exactcode.com | http://exactscan.com | http://ocrkit.com | http://t2-project.org | http://rene.rebe.de >

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[alsa-devel] [PATCH v7 0/7] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 22 Aug '18

22 Aug '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. The default HDaudio legacy driver does not allow for the use of the DSP, this patch set makes it possible while reusing existing code for HDAudio codecs and without significant changes to the legacy driver. This v7 is based on Marks' for-next branch - the merge of topic/drm_audio_component is not included here. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). Both the HDaudio and HDMI outputs were tested. Credits: all the initial code was written by Rakesh Ughreja, the rebase to broonie/for-next, cleanups and additional tests were done by Pierre Bossart. TODO in future update: 1. fix the HDMI jack detection which only works after the mixer values are set, which isn't practical for headless devices always connected (this is a problem in the hdac_hdmi codec that was present before this series was submitted) 2. when the screen goes blank, HDMI playback stops. This isn't related to this patchset since we rely on the existing hdac_hdmi codec, but will need additional investigations. Changes v7 - fix route count for headless boards such as Up2 to enable all 3 iDisp routes (only iDisp3 was set-up due to a bad macro). Changes v6 (comments from Mark and Takashi) - simplify Kconfig options, remove FORCE for now and make code simpler - change SPDIX style to C++ for C files - add log message if invalid configuration reached - remove static int counter, added private counter in context instead - fixed a couple of indendation issues to make checkpatch.pl --strict happy Changes v5 (comments from Vinod) - fix SPDIF style across patches - patch 1: update commit message to remove reference to "fix" - patch 3: remove platform name assignment and Xmas tree style - patch 7: simplify error handling, add missing _put() on errors, remove tests on ops, add switch statement, fix alignment - patch 8: fix indentation issues, use bool instead of tristate Changes v4: - rebase/update on Takashi's topic/drm_audio_component branch - changes in the HDaudio detection to avoid adding a fake ACPI ID - new Kconfigs to control HDaudio detection Changes v3: - port to component model - additional tests on ApolloLake and KabyLake NUC devices - cleanups (alignment, typos, etc) Changes v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Pierre-Louis Bossart (2): ASoC: Intel: common: add table for HDA-based platforms ASoC: Intel: Skylake: add option to control HDAudio + DSP usage Rakesh Ughreja (5): ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers include/sound/soc-acpi-intel-match.h | 6 + sound/pci/hda/hda_bind.c | 12 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 485 ++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 + sound/soc/intel/Kconfig | 9 + sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 127 +++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 38 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 182 +++++++ sound/soc/intel/common/Makefile | 3 +- .../intel/common/soc-acpi-intel-hda-match.c | 40 ++ sound/soc/intel/skylake/skl-pcm.c | 70 ++- sound/soc/intel/skylake/skl.c | 104 +++- sound/soc/intel/skylake/skl.h | 12 +- 17 files changed, 1103 insertions(+), 27 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c -- 2.17.1

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