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* Merge tag 'sound-fix-5.7-rc1' of ↵Linus Torvalds2020-04-1029-39/+324
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "A collection of small fixes gathered since the previous update. ALSA core: - Regression fix for OSS PCM emulation ASoC: - Trivial fixes in reg bit mask ops, DAPM, DPCM and topology - Lots of fixes for Intel-based devices - Minor fixes for AMD, STM32, Qualcomm, Realtek Others: - Fixes for the bugs in mixer handling in HD-audio and ice1724 drivers that were caught by the recent kctl validator - New quirks for HD-audio and USB-audio Also this contains a fix for EDD firmware fix, which slipped from anyone's hands" * tag 'sound-fix-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (35 commits) ALSA: hda: Add driver blacklist ALSA: usb-audio: Add mixer workaround for TRX40 and co ALSA: hda/realtek - Add quirk for MSI GL63 ALSA: ice1724: Fix invalid access for enumerated ctl items ALSA: hda: Fix potential access overflow in beep helper ASoC: cs4270: pull reset GPIO low then high ALSA: hda/realtek - Add HP new mute led supported for ALC236 ALSA: hda/realtek - Add supported new mute Led for HP ASoC: rt5645: Add platform-data for Medion E1239T ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet ASoC: stm32: sai: Add missing cleanup ALSA: usb-audio: Add registration quirk for Kingston HyperX Cloud Alpha S ASoC: Intel: atom: Fix uninitialized variable compiler warning ASoC: Intel: atom: Check drv->lock is locked in sst_fill_and_send_cmd_unlocked ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map() ASoC: SOF: Turn "firmware boot complete" message into a dbg message ALSA: usb-audio: Add Pioneer DJ DJM-250MK2 quirk ALSA: pcm: oss: Fix regression by buffer overflow fix (again) ALSA: pcm: oss: Fix regression by buffer overflow fix edd: Use scnprintf() for avoiding potential buffer overflow ...
| * Merge tag 'asoc-fix-v5.7' of ↵Takashi Iwai2020-04-0821-26/+91
| |\ | | | | | | | | | | | | | | | | | | | | | | | | https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.7 A collection of fixes that have been accumilated since the merge window, mainly relating to x86 platform support.
| | * ASoC: cs4270: pull reset GPIO low then highMike Willard2020-04-071-5/+35
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Pull the RST line low then high when initializing the driver, in order to force a reset of the chip. Previously, the line was not pulled low, which could result in the chip registers not resetting to their default values on boot. Signed-off-by: Mike Willard <mwillard@izotope.com> Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20200401205454.79792-1-mwillard@izotope.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: rt5645: Add platform-data for Medion E1239THans de Goede2020-04-061-0/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Medion E1239T uses the default jack-detect mode 3, but instead of using an analog microphone it is using a DMIC on dmic-data-pin 1, like other models following Intel's Brasswell's reference design. This commit adds a DMI quirk pointing to the intel_braswell_platform_data for this model. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200402185257.3355-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tabletHans de Goede2020-04-061-0/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200405133726.24154-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: stm32: sai: Add missing cleanupJulia Lawall2020-04-061-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") converts some function calls to their non-devm equivalents. The appropriate cleanup code was added to the remove function, but not to the probe function. Add a call to snd_dmaengine_pcm_unregister to compensate for the call to snd_dmaengine_pcm_register in case of subsequent failure. Fixes: commit 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr> Acked-by: Olivier Moysan <olivier.moysan@st.com> Link: https://lore.kernel.org/r/1586099028-5104-1-git-send-email-Julia.Lawall@inria.fr Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: Intel: atom: Fix uninitialized variable compiler warningHans de Goede2020-04-031-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | GCC 10 gives a "variable might be used uninitialized" warning for the block variable in sst_prepare_and_post_msg(). This is a false-positive warning, but lets fix it anyways. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: Intel: atom: Check drv->lock is locked in sst_fill_and_send_cmd_unlockedHans de Goede2020-04-031-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | sst_fill_and_send_cmd_unlocked must be called with the drv->lock mutex locked already. In the past there have been cases where this was not the case, add a WARN_ON to check for drv->lock being locked. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()Hans de Goede2020-04-031-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: SOF: Turn "firmware boot complete" message into a dbg messageHans de Goede2020-04-031-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Using a Canon Lake machine with the SOF driver causes dmesg to fill up with a ton of these messages: [ 275.902194] sof-audio-pci 0000:00:1f.3: firmware boot complete [ 351.529358] sof-audio-pci 0000:00:1f.3: firmware boot complete [ 560.049047] sof-audio-pci 0000:00:1f.3: firmware boot complete etc. Since the DSP is powered down when not in used this happens everytime e.g. a notification plays, polluting dmesg. Turn this messages into a debug message, matching what the code already does for the ""booting DSP firmware" message. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200402184948.3014-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: topology: use name_prefix for new kcontrol이경택2020-04-011-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | Current topology doesn't add prefix of component to new kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com> Link: https://lore.kernel.org/r/009b01d60804$ae25c2d0$0a714870$@samsung.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: rt5682: Fix build error without CONFIG_I2CYueHaibing2020-04-011-1/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If I2C is n but SoundWire is m, building fails: sound/soc/codecs/rt5682.c:3716:1: warning: data definition has no type or storage class module_i2c_driver(rt5682_i2c_driver); ^~~~~~~~~~~~~~~~~ sound/soc/codecs/rt5682.c:3716:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] sound/soc/codecs/rt5682.c:3716:1: warning: parameter names (without types) in function declaration Guard this use #ifdef CONFIG_I2C. Fixes: 5549ea647997 ("ASoC: rt5682: fix unmet dependencies") Signed-off-by: YueHaibing <yuehaibing@huawei.com> Link: https://lore.kernel.org/r/20200401091055.34112-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: dpcm: allow start or stop during pause for backend이경택2020-04-011-2/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com> Reviewed-by: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: dapm: connect virtual mux with default value이경택2020-03-311-1/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since a virtual mixer has no backing registers to decide which path to connect, it will try to match with initial state. This is to ensure that the default mixer choice will be correctly powered up during initialization. Invert flag is used to select initial state of the virtual switch. Since actual hardware can't be disconnected by virtual switch, connected is better choice as initial state in many cases. Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com> Link: https://lore.kernel.org/r/01a301d60731$b724ea10$256ebe30$@samsung.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: qcom: q6asm-dai: Add SNDRV_PCM_INFO_BATCH flagStephan Gerhold2020-03-311-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | At the moment, playing audio with PulseAudio with the qdsp6 driver results in distorted sound. It seems like its timer-based scheduling does not work properly with qdsp6 since setting tsched=0 in the PulseAudio configuration avoids the issue. Apparently this happens when the pointer() callback is not accurate enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop PulseAudio from using timer-based scheduling by default. According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html: The flag is being used in the sense explained in the previous audio meeting -- the data transfer granularity isn't fine enough but aligned to the period size (or less). q6asm-dai reports the position as multiple of prtd->pcm_count = snd_pcm_lib_period_bytes(substream) so it indeed just a multiple of the period size. Therefore adding the flag here seems appropriate and makes audio work out of the box. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Stephan Gerhold <stephan@gerhold.net> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20200330175210.47518-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie@kernel.org>
| | * Merge series "ASoC: Intel: boards: Remove ignore_suspend flag from SSP0 dai ↵Mark Brown2020-03-304-4/+0
| | |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | link" from Cezary Rojewski <cezary.rojewski@intel.com>: As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Link to first message in conversation: https://lkml.org/lkml/2020/3/18/54 Cezary Rojewski (4): ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link sound/soc/intel/boards/bdw-rt5650.c | 1 - sound/soc/intel/boards/bdw-rt5677.c | 1 - sound/soc/intel/boards/broadwell.c | 1 - sound/soc/intel/boards/haswell.c | 1 - 4 files changed, 4 deletions(-) -- 2.17.1
| | | * ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai linkCezary Rojewski2020-03-301-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Dominik Brodowski <linux@dominikbrodowski.net> Cc: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20200319204947.18963-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai linkCezary Rojewski2020-03-301-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Dominik Brodowski <linux@dominikbrodowski.net> Cc: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20200319204947.18963-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai linkCezary Rojewski2020-03-301-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Dominik Brodowski <linux@dominikbrodowski.net> Cc: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20200319204947.18963-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai linkCezary Rojewski2020-03-301-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Link to first message in conversation: https://lkml.org/lkml/2020/3/18/54 Reported-by: Dominik Brodowski <linux@dominikbrodowski.net> Suggested-by: Mark Brown <broonie@kernel.org> Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200319204947.18963-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * | ASoC: soc-dai: fix DAI startup/shutdown sequencePierre-Louis Bossart2020-03-301-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The addition of a single flag to track the DAI status prevents the DAI startup sequence from being called on capture if the DAI is already used for playback. Fix by extending the existing code with one flag per direction. Fixes: b56be800f1292 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once") Reported-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Tested-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20200330160602.10180-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * | ASoC: fix regwmask이경택2020-03-301-2/+2
| | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If regwshift is 32 and the selected architecture compiles '<<' operator for signed int literal into rotating shift, '1<<regwshift' became 1 and it makes regwmask to 0x0. The literal is set to unsigned long to get intended regwmask. Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com> Link: https://lore.kernel.org/r/001001d60665$db7af3e0$9270dba0$@samsung.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: AMD: Clear format bits before setting themAkshu Agrawal2020-03-302-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This avoids residual bit form previous format when the format is changed. Hence, the resultant format is not an invalid one. Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com> Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com> Link: https://lore.kernel.org/r/20200328093921.32211-1-akshu.agrawal@amd.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: bcm: Fix pointer cast warningTakashi Iwai2020-03-301-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The NULL check can be done gracefully without cast. It fixes a compile warning like: sound/soc/bcm/bcm63xx-pcm-whistler.c:184:6: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast] Fixes: 88eb404ccc3e ("ASoC: brcm: Add DSL/PON SoC audio driver") Signed-off-by: Takashi Iwai <tiwai@suse.de> Link: https://lore.kernel.org/r/20200330135645.9707-1-tiwai@suse.de Signed-off-by: Mark Brown <broonie@kernel.org>
| * | ALSA: hda: Add driver blacklistTakashi Iwai2020-04-081-0/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent AMD platform exposes an HD-audio bus but without any actual codecs, which is internally tied with a USB-audio device, supposedly. It results in "no codecs" error of HD-audio bus driver, and it's nothing but a waste of resources. This patch introduces a static blacklist table for skipping such a known bogus PCI SSID entry. As of writing this patch, the known SSIDs are: * 1043:874f - ASUS ROG Zenith II / Strix * 1462:cb59 - MSI TRX40 Creator * 1462:cb60 - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200408140449.22319-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: Add mixer workaround for TRX40 and coTakashi Iwai2020-04-081-0/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some recent boards (supposedly with a new AMD platform) contain the USB audio class 2 device that is often tied with HD-audio. The device exposes an Input Gain Pad control (id=19, control=12) but this node doesn't behave correctly, returning an error for each inquiry of GET_MIN and GET_MAX that should have been mandatory. As a workaround, simply ignore this node by adding a usbmix_name_map table entry. The currently known devices are: * 0414:a002 - Gigabyte TRX40 Aorus Pro WiFi * 0b05:1916 - ASUS ROG Zenith II * 0b05:1917 - ASUS ROG Strix * 0db0:0d64 - MSI TRX40 Creator * 0db0:543d - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200408140449.22319-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - Add quirk for MSI GL63Takashi Iwai2020-04-081-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: ice1724: Fix invalid access for enumerated ctl itemsTakashi Iwai2020-04-071-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda: Fix potential access overflow in beep helperTakashi Iwai2020-04-071-1/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - Add HP new mute led supported for ALC236Kailang Yang2020-04-071-0/+44
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | HP new platform has new mute led feature. COEF index 0x34 bit 5 to control playback mute led. COEF index 0x35 bit 2 and bit 3 to control Mic mute led. [ corrected typos by tiwai ] Signed-off-by: Kailang Yang <kailang@realtek.com> Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - Add supported new mute Led for HPKailang Yang2020-04-071-0/+81
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | HP Note Book supported new mute Led. Hardware PIN was not enough to meet old LED rule. JD2 to control playback mute led. GPO3 to control capture mute led. (ALC285 didn't control GPO3 via verb command) This two PIN just could control by COEF registers. [ corrected typos by tiwai ] Signed-off-by: Kailang Yang <kailang@realtek.com> Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: Add registration quirk for Kingston HyperX Cloud Alpha SEmmanuel Pescosta2020-04-041-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud Alpha S (0951:16d8) uses two interfaces, but only the second interface contains the capture stream. This patch delays the registration until the second interface appears. Signed-off-by: Emmanuel Pescosta <emmanuelpescosta099@gmail.com> Link: https://lore.kernel.org/r/20200404153843.9288-1-emmanuelpescosta099@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: Add Pioneer DJ DJM-250MK2 quirkFrantišek Kučera2020-04-031-0/+42
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card. The MIDI controller part is standard but the PCM part is "vendor specific". Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE. Input is not working. Signed-off-by: František Kučera <franta-linux@frantovo.cz> Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.cz Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: pcm: oss: Fix regression by buffer overflow fix (again)Takashi Iwai2020-04-031-10/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ This is essentially the same fix as commit ae769d355664, but it's adapted to the latest code for 5.7; hence it contains no Fixes or other tags for avoid backport confusion -- tiwai ] The recent fix for the OOB access in PCM OSS plugins (commit f2ecf903ef06: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a regression on OSS applications. The patch introduced the size check in client and slave size calculations to limit to each plugin's buffer size, but I overlooked that some code paths call those without allocating the buffer but just for estimation. This patch fixes the bug by skipping the size check for those code paths while keeping checking in the actual transfer calls. Link: https://lore.kernel.org/r/20200403073818.27943-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - Add quirk for Lenovo Carbon X1 8th genHans de Goede2020-04-021-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The audio setup on the Lenovo Carbon X1 8th gen is the same as that on the Lenovo Carbon X1 7th gen, as such it needs the same ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk. This fixes volume control of the speaker not working among other things. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196 Cc: stable@vger.kernel.org Suggested-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Hans de Goede <hdegoede@redhat.com> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Link: https://lore.kernel.org/r/20200402174311.238614-1-hdegoede@redhat.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge tag 'spdx-5.7-rc1' of ↵Linus Torvalds2020-04-031-1/+1
|\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/spdx Pull SPDX updates from Greg KH: "Here are three SPDX patches for 5.7-rc1. One fixes up the SPDX tag for a single driver, while the other two go through the tree and add SPDX tags for all of the .gitignore files as needed. Nothing too complex, but you will get a merge conflict with your current tree, that should be trivial to handle (one file modified by two things, one file deleted.) All three of these have been in linux-next for a while, with no reported issues other than the merge conflict" * tag 'spdx-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/spdx: ASoC: MT6660: make spdxcheck.py happy .gitignore: add SPDX License Identifier .gitignore: remove too obvious comments
| * | | ASoC: MT6660: make spdxcheck.py happyLukas Bulwahn2020-03-251-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The SPDX-License-Identifier shall not be suffixed with anything further. This makes ./scripts/spdxcheck.py complain: sound/soc/codecs/mt6660.c: 1:36 Invalid token: // Clean up SPDX-License-Identifier line to make spdxcheck.py happy. Signed-off-by: Lukas Bulwahn <lukas.bulwahn@gmail.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| * | | .gitignore: add SPDX License IdentifierMasahiro Yamada2020-03-251-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Add SPDX License Identifier to all .gitignore files. Signed-off-by: Masahiro Yamada <masahiroy@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| * | | .gitignore: remove too obvious commentsMasahiro Yamada2020-03-251-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some .gitignore files have comments like "Generated files", "Ignore generated files" at the header part, but they are too obvious. Signed-off-by: Masahiro Yamada <masahiroy@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* | | | Merge tag 'sound-5.7-rc1' of ↵Linus Torvalds2020-04-02378-3925/+17505
|\ \ \ \ | | |/ / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "This became again a busy development cycle. There are few ALSA core updates (merely API cleanups and sparse fixes), with the majority of other changes are found in ASoC scene. Here are some highlights: ALSA core: - More helper macros for sparse warning fixes (e.g. bitwise types) - Slight optimization of PCM OSS locks - Make common handling for PCM / compress buffers (for SOF) ASoC: - Lots of code refactoring and modernization for (still ongoing) componentization works - Conversion of SND_SOC_ALL_CODECS to use imply - Continued refactoring and fixing of the Intel SOF/SST support, including the initial (but still incomplete) SoundWire support - SoundWire and more advanced clocking support for Realtek RT5682 - Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and TLV320ADCX140 HD-audio: - Optimizations in HDMI jack handling - A few new quirks and fixups for Realtek codecs USB-audio: - Delayed registration support - New quirks for Motu, Kingston, Presonus" * tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (415 commits) ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h" ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups ALSA: hda/realtek - Set principled PC Beep configuration for ALC256 ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256 ALSA: hda/realtek - a fake key event is triggered by running shutup ALSA: hda: default enable CA0132 DSP support ASoC: amd: acp3x-pcm-dma: clean up two indentation issues ASoC: tlv320adcx140: Remove undocumented property ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver ASoC: Intel: boards: add sof_sdw machine driver ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms ASoC: rt5682: move DAI clock registry to I2S mode ASoC: pxa: magician: convert to use i2c_new_client_device() ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers ...
| * | | ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra ↵Andreas Steinmetz2020-03-311-5/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | endpoint descriptor The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz <ast@domdv.de> Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@domdv.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixupsThomas Hebb2020-03-311-34/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | patch_realtek.c has historically failed to properly configure the PC Beep Hidden Register for the ALC256 codec (among others). Depending on your kernel version, symptoms of this misconfiguration can range from chassis noise, picked up by a poorly-shielded PCBEEP trace, getting amplified and played on your internal speaker and/or headphones to loud feedback, which responds to the "Headphone Mic Boost" ALSA control, getting played through your headphones. For details of the problem, see the patch in this series titled "ALSA: hda/realtek - Set principled PC Beep configuration for ALC256", which fixes the configuration. These symptoms have been most noticed on the Dell XPS 13 9350 and 9360, popular laptops that use the ALC256. As a result, several model-specific fixups have been introduced to try and fix the problem, the most egregious of which locks the "Headphone Mic Boost" control as a hack to minimize noise from a feedback loop that shouldn't have been there in the first place. Now that the underlying issue has been fixed, remove all these fixups. Remaining fixups needed by the XPS 13 are all picked up by existing pin quirks. This change should, for the XPS 13 9350/9360 - Significantly increase volume and audio quality on headphones - Eliminate headphone popping on suspend/resume - Allow "Headphone Mic Boost" to be set again, making the headphone jack fully usable as a microphone jack too. Fixes: 8c69729b4439 ("ALSA: hda - Fix headphone noise after Dell XPS 13 resume back from S3") Fixes: 423cd785619a ("ALSA: hda - Fix headphone noise on Dell XPS 13 9360") Fixes: e4c9fd10eb21 ("ALSA: hda - Apply headphone noise quirk for another Dell XPS 13 variant") Fixes: 1099f48457d0 ("ALSA: hda/realtek: Reduce the Headphone static noise on XPS 9350/9360") Cc: stable@vger.kernel.org Signed-off-by: Thomas Hebb <tommyhebb@gmail.com> Link: https://lore.kernel.org/r/b649a00edfde150cf6eebbb4390e15e0c2deb39a.1585584498.git.tommyhebb@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda/realtek - Set principled PC Beep configuration for ALC256Thomas Hebb2020-03-311-6/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Realtek PC Beep Hidden Register[1] is currently set by patch_realtek.c in two different places: In alc_fill_eapd_coef(), it's set to the value 0x5757, corresponding to non-beep input on 1Ah and no 1Ah loopback to either headphones or speakers. (Although, curiously, the loopback amp is still enabled.) This write was added fairly recently by commit e3743f431143 ("ALSA: hda/realtek - Dell headphone has noise on unmute for ALC236") and is a safe default. However, it happens in the wrong place: alc_fill_eapd_coef() runs on module load and cold boot but not on S3 resume, meaning the register loses its value after suspend. Conversely, in alc256_init(), the register is updated to unset bit 13 (disable speaker loopback) and set bit 5 (set non-beep input on 1Ah). Although this write does run on S3 resume, it's not quite enough to fix up the register's default value of 0x3717. What's missing is a set of bit 14 to disable headphone loopback. Without that, we end up with a feedback loop where the headphone jack is being driven by amplified samples of itself[2]. This change eliminates the update in alc256_init() and replaces it with the 0x5757 write from alc_fill_eapd_coef(). Kailang says that 0x5757 is supposed to be the codec's default value, so using it will make debugging easier for Realtek. Affects the ALC255, ALC256, ALC257, ALC235, and ALC236 codecs. [1] Newly documented in Documentation/sound/hd-audio/realtek-pc-beep.rst [2] Setting the "Headphone Mic Boost" control from userspace changes this feedback loop and has been a widely-shared workaround for headphone noise on laptops like the Dell XPS 13 9350. This commit eliminates the feedback loop and makes the workaround unnecessary. Fixes: e1e8c1fdce8b ("ALSA: hda/realtek - Dell headphone has noise on unmute for ALC236") Cc: stable@vger.kernel.org Signed-off-by: Thomas Hebb <tommyhebb@gmail.com> Link: https://lore.kernel.org/r/bf22b417d1f2474b12011c2a39ed6cf8b06d3bf5.1585584498.git.tommyhebb@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | Merge tag 'asoc-v5.7' of ↵Takashi Iwai2020-03-30332-3431/+16234
| |\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Updates for v5.7 This is a very big update for the core since Morimoto-san has been rather busy continuing his refactorings to clean up a lot of the cruft that we have accumilated over the years. We've also gained several new drivers, including initial (but still not complete) parts of the Intel SoundWire support. - Lots of refactorings to modernize the code from Morimoto-san. - Conversion of SND_SOC_ALL_CODECS to use imply from Geert Uytterhoeven. - Continued refactoring and fixing of the Intel support. - Soundwire and more advanced clocking support for Realtek RT5682. - Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and TLV320ADCX140.
| | * \ \ Merge branch 'asoc-5.7' into asoc-nextMark Brown2020-03-27326-3338/+16088
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| | | * | Merge series "ASoC: Intel: add SoundWire machine driver" from Pierre-Louis ↵Mark Brown2020-03-2714-11/+1896
| | | |\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Bossart <pierre-louis.bossart@linux.intel.com>: To handle multiple hardware combinations, this patchset suggests a single machine driver which will create and initialize dailinks dynamically. This allows us to support new configurations easily, as shown with the TigerLake rt5682 example. Each configuration updates the card component string, and UCM can test for the presence of components to configure them as needed. Since we use a single the machine driver name, all previous ACPI tables need to be updated. That should have no impact since the machine drivers listed at the time were not upstreamed and are no longer maintained. Naveen Manohar (2): ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function Pierre-Louis Bossart (1): ASoC: Intel: boards: add sof_sdw machine driver Rander Wang (1): ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms sound/soc/intel/boards/Kconfig | 24 + sound/soc/intel/boards/Makefile | 8 +- sound/soc/intel/boards/sof_sdw.c | 962 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 114 +++ sound/soc/intel/boards/sof_sdw_dmic.c | 42 + sound/soc/intel/boards/sof_sdw_hdmi.c | 97 ++ sound/soc/intel/boards/sof_sdw_rt1308.c | 151 +++ sound/soc/intel/boards/sof_sdw_rt5682.c | 126 +++ sound/soc/intel/boards/sof_sdw_rt700.c | 125 +++ sound/soc/intel/boards/sof_sdw_rt711.c | 156 +++ sound/soc/intel/boards/sof_sdw_rt715.c | 42 + .../intel/common/soc-acpi-intel-cml-match.c | 24 +- .../intel/common/soc-acpi-intel-icl-match.c | 6 +- .../intel/common/soc-acpi-intel-tgl-match.c | 30 +- 14 files changed, 1896 insertions(+), 11 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw.c create mode 100644 sound/soc/intel/boards/sof_sdw_common.h create mode 100644 sound/soc/intel/boards/sof_sdw_dmic.c create mode 100644 sound/soc/intel/boards/sof_sdw_hdmi.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt1308.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt5682.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt700.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt711.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c -- 2.20.1
| | | | * | ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper functionNaveen Manohar2020-03-275-0/+152
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add support for Google Volteer device. As per new unified soundwire machine driver, add rt5682-sdw helper function, which configures codec to Link0. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Naveen Manohar <naveen.m@intel.com> Link: https://lore.kernel.org/r/20200325220746.29601-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | | | * | ASoC: Intel: common: add match table for TGL RT5682 SoundWire driverNaveen Manohar2020-03-271-0/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | RT5682 is in SoundWire mode on link0. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Naveen Manohar <naveen.m@intel.com> Link: https://lore.kernel.org/r/20200325220746.29601-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | | | * | ASoC: Intel: boards: add sof_sdw machine driverPierre-Louis Bossart2020-03-2710-2/+1693
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This machine driver provides support for different configurations: RT700, RT711, RT1308 (1x and 2x, I2S or SoundWire mode), and RT715 CometLake, Icelake, TigerLake. PDM digital microphones HDMI To avoid introducing one driver per configuration, this common machine driver relies on platform-specific information, tables and quirks to dynamically create the relevant dailinks. Unlike a lot of machine drivers, we use different DAI links for SoundWire capture and playback since the Cadence PDIs can do capture OR playback, not both simultaneously. For each configuration, the card component string is updated so that UCM can select the relevant parts. Signed-off-by: Rander Wang <rander.wang@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Link: https://lore.kernel.org/r/20200325220746.29601-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | | | * | ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platformsRander Wang2020-03-273-9/+27
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Update topology and reflect change to unified machine driver for SoundWire. Signed-off-by: Rander Wang <rander.wang@intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200325220746.29601-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
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