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* drivers/powerpc: Replace _ALIGN_UP() by ALIGN()Christophe Leroy2020-05-111-1/+1
| | | | | | | | | | | | _ALIGN_UP() is specific to powerpc ALIGN() is generic and does the same Replace _ALIGN_UP() by ALIGN() Signed-off-by: Christophe Leroy <christophe.leroy@c-s.fr> Signed-off-by: Michael Ellerman <mpe@ellerman.id.au> Reviewed-by: Joel Stanley <joel@jms.id.au> Link: https://lore.kernel.org/r/a5945463f86c984151962a475a3ee56a2893e85d.1587407777.git.christophe.leroy@c-s.fr
* Merge tag 'sound-5.7-rc2' of ↵Linus Torvalds2020-04-178-68/+99
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "One significant regression fix is for HD-audio buffer preallocation. In 5.6 it was set to non-prompt for x86 and forced to 0, but this turned out to be problematic for some applications, hence it gets reverted. Distros would need to restore CONFIG_SND_HDA_PREALLOC_SIZE value to the earlier values they've used in the past. Other than that, we've received quite a few small fixes for HD-audio and USB-audio. Most of them are for dealing with the broken TRX40 mobos and the runtime PM without HD-audio codecs" * tag 'sound-5.7-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda: call runtime_allow() for all hda controllers ALSA: hda: Allow setting preallocation again for x86 ALSA: hda: Explicitly permit using autosuspend if runtime PM is supported ALSA: hda: Skip controller resume if not needed ALSA: hda: Keep the controller initialization even if no codecs found ALSA: hda: Release resources at error in delayed probe ALSA: hda: Honor PM disablement in PM freeze and thaw_noirq ops ALSA: hda: Don't release card at firmware loading error ALSA: usb-audio: Check mapping at creating connector controls, too ALSA: usb-audio: Don't create jack controls for PCM terminals ALSA: usb-audio: Don't override ignore_ctl_error value from the map ALSA: usb-audio: Filter error from connector kctl ops, too ALSA: hda/realtek - Enable the headset mic on Asus FX505DT ALSA: ctxfi: Remove unnecessary cast in kfree
| * ALSA: hda: call runtime_allow() for all hda controllersHui Wang2020-04-141-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Before the pci_driver->probe() is called, the pci subsystem calls runtime_forbid() and runtime_get_sync() on this pci dev, so only call runtime_put_autosuspend() is not enough to enable the runtime_pm on this device. For controllers with vgaswitcheroo feature, the pci/quirks.c will call runtime_allow() for this dev, then the controllers could enter rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like Intel hda controllers, the runtime_pm is not enabled because the runtime_allow() is not called. Since it is no harm calling runtime_allow() twice, here let hda driver call runtime_allow() for all controllers. Then the runtime_pm is enabled on all controllers after the put_autosuspend() is called. Signed-off-by: Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20200414142725.6020-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda: Allow setting preallocation again for x86Takashi Iwai2020-04-131-3/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit c31427d0d21e ("ALSA: hda: No preallocation on x86 platforms") changed CONFIG_SND_HDA_PREALLOC_SIZE setup and its default to zero for x86, as the preallocation should work almost all cases. However, this expectation was too naive; some applications try to allocate as the max buffer size as possible, and it leads to the memory exhaustion. More badly, the commit changed the kconfig no longer adjustable for x86, so you can't fix it statically (although it can be still adjusted via procfs). So, practically seen, it's more recommended to set a reasonable limit for x86, too. This patch follows to that experience, and changes the default to 2048 and allow the kconfig adjustable again. Fixes: c31427d0d21e ("ALSA: hda: No preallocation on x86 platforms") Cc: <stable@vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223 Link: https://lore.kernel.org/r/20200413201919.24241-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda: Explicitly permit using autosuspend if runtime PM is supportedRoy Spliet2020-04-131-1/+3
| | | | | | | | | | | | | | | | | | | | This fixes runtime PM not working after a suspend-to-RAM cycle at least for the codec-less HDA device found on NVIDIA GPUs. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Signed-off-by: Roy Spliet <nouveau@spliet.org> Link: https://lore.kernel.org/r/20200413082034.25166-7-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda: Skip controller resume if not neededTakashi Iwai2020-04-132-12/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The HD-audio controller does system-suspend and resume operations by directly calling its helpers __azx_runtime_suspend() and __azx_runtime_resume(). However, in general, we don't have to resume always the device fully at the system resume; typically, if a device has been runtime-suspended, we can leave it to runtime resume. Usually for achieving this, the driver would call pm_runtime_force_suspend() and pm_runtime_force_resume() pairs in the system suspend and resume ops. Unfortunately, this doesn't work for the resume path in our case. For handling the jack detection at the system resume, a child codec device may need the (literally) forcibly resume even if it's been runtime-suspended, and for that, the controller device must be also resumed even if it's been suspended. This patch is an attempt to improve the situation. It replaces the direct __azx_runtime_suspend()/_resume() calls with with pm_runtime_force_suspend() and pm_runtime_force_resume() with a slight trick as we've done for the codec side. More exactly: - azx_has_pm_runtime() check is dropped from azx_runtime_suspend() and azx_runtime_resume(), so that it can be properly executed from the system-suspend/resume path - The WAKEEN handling depends on the card's power state now; it's set and cleared only for the runtime-suspend - azx_resume() checks whether any codec may need the forcible resume beforehand. If the forcible resume is required, it does temporary PM refcount up/down for actually triggering the runtime resume. - A new helper function, hda_codec_need_resume(), is introduced for checking whether the codec needs a forcible runtime-resume, and the existing code is rewritten with that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-6-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda: Keep the controller initialization even if no codecs foundTakashi Iwai2020-04-131-5/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently, when the HD-audio controller driver doesn't detect any codecs, it tries to abort the probe. But this abort happens at the delayed probe, i.e. the primary probe call already returned success, hence the driver is never unbound until user does so explicitly. As a result, it may leave the HD-audio device in the running state without the runtime PM. More badly, if the device is a HD-audio bus that is tied with a GPU, GPU cannot reach to the full power down and consumes unnecessarily much power. This patch changes the logic after no-codec situation; it continues probing without the further codec initialization but keep the controller driver running normally. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Tested-by: Roy Spliet <nouveau@spliet.org> Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda: Release resources at error in delayed probeTakashi Iwai2020-04-132-13/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd-hda-intel driver handles the most of its probe task in the delayed work (either via workqueue or via firmware loader). When an error happens in the later delayed probe, we can't deregister the device itself because the probe callback already returned success and the device was bound. So, for now, we set hda->init_failed flag and make the rest untouched until the device gets really unbound. However, this leaves the device up running, keeping the resources without any use that prevents other operations. In this patch, we release the resources at first when a probe error happens in the delayed probe stage, but keeps the top-level object, so that the PM and other ops can still refer to the object itself. Also for simplicity, snd_hda_intel object is allocated via devm, so that we can get rid of the explicit kfree calls. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda: Honor PM disablement in PM freeze and thaw_noirq opsTakashi Iwai2020-04-131-0/+4
| | | | | | | | | | | | | | | | | | | | | | freeze_noirq and thaw_noirq need to check the PM availability like other PM ops. There are cases where the device got disabled due to the error, and the PM operation should be ignored for that. Fixes: 3e6db33aaf1d ("ALSA: hda - Set SKL+ hda controller power at freeze() and thaw()") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda: Don't release card at firmware loading errorTakashi Iwai2020-04-131-14/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | At the error path of the firmware loading error, the driver tries to release the card object and set NULL to drvdata. This may be referred badly at the possible PM action, as the driver itself is still bound and the PM callbacks read the card object. Instead, we continue the probing as if it were no option set. This is often a better choice than the forced abort, too. Fixes: 5cb543dba986 ("ALSA: hda - Deferred probing with request_firmware_nowait()") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: Check mapping at creating connector controls, tooTakashi Iwai2020-04-122-8/+14
| | | | | | | | | | | | | | | | | | | | | | | | Add the mapping check to build_connector_control() so that the device specific quirk can provide the node to skip for the badly behaving connector controls. As an example, ALC1220-VB-based codec implements the skip entry for the broken SPDIF connector detection. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200412081331.4742-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: Don't create jack controls for PCM terminalsTakashi Iwai2020-04-121-3/+6
| | | | | | | | | | | | | | | | | | | | | | | | Some funky firmwares set the connector flag even on PCM terminals although it doesn't make sense (and even actually the firmware doesn't react properly!). Let's skip creation of jack controls in such a case. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200412081331.4742-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: Don't override ignore_ctl_error value from the mapTakashi Iwai2020-04-121-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | The mapping table may contain also ignore_ctl_error flag for devices that are known to behave wild. Since this flag always writes the card's own ignore_ctl_error flag, it overrides the value already set by the module option, so it doesn't follow user's expectation. Let's fix the code not to clear the flag that has been set by user. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200412081331.4742-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: usb-audio: Filter error from connector kctl ops, tooTakashi Iwai2020-04-121-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The ignore_ctl_error option should filter the error at kctl accesses, but there was an overlook: mixer_ctl_connector_get() returns an error from the request. This patch covers the forgotten code path and apply filter_error() properly. The locking error is still returned since this is a fatal error that has to be reported even with ignore_ctl_error option. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200412081331.4742-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda/realtek - Enable the headset mic on Asus FX505DTAdam Barber2020-04-111-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | On Asus FX505DT with Realtek ALC233, the headset mic is connected to pin 0x19, with default 0x411111f0. Enable headset mic by reconfiguring the pin to an external mic associated with the headphone on 0x21. Mic jack detection was also found to be working. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207131 Signed-off-by: Adam Barber <barberadam995@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200410090032.2759-1-barberadam995@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: ctxfi: Remove unnecessary cast in kfreeXu Wang2020-04-091-7/+7
| | | | | | | | | | | | | | | | Remove unnecassary casts in the argument to kfree. Signed-off-by: Xu Wang <vulab@iscas.ac.cn> Link: https://lore.kernel.org/r/20200409112052.13402-1-vulab@iscas.ac.cn Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | 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>
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