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| | * ASoC: topology: Check return value of soc_tplg_*_createAmadeusz Sławiński2020-04-091-6/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Functions soc_tplg_denum_create, soc_tplg_dmixer_create, soc_tplg_dbytes_create can fail, so their return values should be checked and error should be propagated. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200327204729.397-4-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: topology: Check return value of soc_tplg_create_tlvAmadeusz Sławiński2020-04-091-2/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Function soc_tplg_create_tlv can fail, so we should check if it succeded or not and proceed appropriately. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200327204729.397-3-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: topology: Add missing memory checksAmadeusz Sławiński2020-04-091-13/+49
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | kstrdup is an allocation function and it can fail, so its return value should be checked and handled appropriately. In order to check all cases, we need to modify set_stream_info to return a value, so check that everything went correctly when doing kstrdup(). Later add proper checks and error handlers. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200327204729.397-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| * | ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd genAlexander Tsoy2020-04-211-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Due to rounding error driver sometimes incorrectly calculate next packet size, which results in audible clicks on devices with synchronous playback endpoints. For example on a high speed bus and a sample rate 44.1 kHz it loses one sample every ~40.9 seconds. Fortunately playback interface on Scarlett 2i4 2nd gen has a working explicit feedback endpoint, so we can switch playback data endpoint to asynchronous mode as a workaround. Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200421190908.462860-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usx2y: Fix potential NULL dereferenceTakashi Iwai2020-04-211-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The error handling code in usX2Y_rate_set() may hit a potential NULL dereference when an error occurs before allocating all us->urb[]. Add a proper NULL check for fixing the corner case. Reported-by: Lin Yi <teroincn@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200420075529.27203-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2Gregor Pintar2020-04-212-84/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Force it to use asynchronous playback. Same quirk has already been added for Focusrite Scarlett Solo (2nd gen) with a commit 46f5710f0b88 ("ALSA: usb-audio: Add quirk for Focusrite Scarlett Solo"). This also seems to prevent regular clicks when playing at 44100Hz on Scarlett 2i2 (2nd gen). I did not notice any side effects. Moved both quirks to snd_usb_audioformat_attributes_quirk() as suggested. Signed-off-by: Gregor Pintar <grpintar@gmail.com> Reviewed-by: Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200420214030.2361-1-grpintar@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobosTakashi Iwai2020-04-203-6/+44
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | TRX40 mobos from MSI and others with ALC1220-VB USB-audio device need yet more quirks for the proper control names. This patch provides the mapping table for those boards, correcting the FU names for volume and mute controls as well as the terminal names for jack controls. It also improves build_connector_control() not to add the directional suffix blindly if the string is given from the mapping table. With this patch applied, the new UCM profiles will be effective. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Link: https://lore.kernel.org/r/20200420062036.28567-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda: Remove ASUS ROG Zenith from the blacklistTakashi Iwai2020-04-191-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a new blacklist for the devices that are known to have empty codecs, and one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f). However, it turned out that the very same PCI SSID is used for the previous model that does have the valid HD-audio codecs and the change broke the sound on it. This patch reverts the corresponding entry as a temporary solution. Although Zenith II and co will see get the empty HD-audio bus again, it'd be merely resource wastes and won't affect the functionality, so it's no end of the world. We'll need to address this later, e.g. by either switching to DMI string matching or using PCI ID & SSID pairs. Fixes: 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") Reported-by: Johnathan Smithinovic <johnathan.smithinovic@gmx.at> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200419071926.22683-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - Fix unexpected init_amp overrideTakashi Iwai2020-04-181-3/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") changed the way to assign spec->init_amp field that specifies the way to initialize the amp. Along with the change, the commit also replaced a few fixups that set spec->init_amp in HDA_FIXUP_ACT_PROBE with HDA_FIXUP_ACT_PRE_PROBE. This was rather aligning to the other fixups, and not supposed to change the actual behavior. However, this change turned out to cause a regression on FSC S7020, which hit exactly the above. The reason was that there is still one place that overrides spec->init_amp after HDA_FIXUP_ACT_PRE_PROBE call, namely in alc_ssid_check(). This patch fixes the regression by adding the proper spec->init_amp override check, i.e. verifying whether it's still ALC_INIT_UNDEFINED. Fixes: 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") Cc: <stable@vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207329 Link: https://lore.kernel.org/r/20200418190639.10082-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devicesAlexander Tsoy2020-04-181-0/+51
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Many Focusrite devices supports a limited set of sample rates per altsetting. These includes audio interfaces with ADAT ports: - Scarlett 18i6, 18i8 1st gen, 18i20 1st gen; - Scarlett 18i8 2nd gen, 18i20 2nd gen; - Scarlett 18i8 3rd gen, 18i20 3rd gen; - Clarett 2Pre USB, 4Pre USB, 8Pre USB. Maximum rate is exposed in the last 4 bytes of Format Type descriptor which has a non-standard bLength = 10. Tested-by: Alexey Skobkin <skobkin-ru@ya.ru> Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200418175815.12211-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/hdmi: Add module option to disable audio component bindingTakashi Iwai2020-04-171-0/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As the recent regression showed, we want sometimes to turn off the audio component binding just for debugging. This patch adds the module option to control it easily without compilation. Fixes: ade49db337a9 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200415162523.27499-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | 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>
generated by cgit v1.2.3 (git 2.25.1) at 2024-10-08 01:36:35 +0000