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2020-05-01Merge tag 'sound-5.7-rc4' of ↵Linus Torvalds
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "Just a collection of small fixes around this time: - One more try for fixing PCM OSS regression - HD-audio: a new quirk for Lenovo, the improved driver blacklisting, a lock fix in the minor error path, and a fix for the possible race at monitor notifiaction - USB-audio: a quirk ID fix, a fix for POD HD500 workaround" * tag 'sound-5.7-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: usb-audio: Correct a typo of NuPrime DAC-10 USB ID ALSA: opti9xx: shut up gcc-10 range warning ALSA: hda/hdmi: fix without unlocked before return ALSA: hda/hdmi: fix race in monitor detection during probe ALSA: hda/realtek - Two front mics on a Lenovo ThinkCenter ALSA: line6: Fix POD HD500 audio playback ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7) ALSA: pcm: oss: Place the plugin buffer overflow checks correctly ALSA: hda: Match both PCI ID and SSID for driver blacklist
2020-04-30ALSA: usb-audio: Correct a typo of NuPrime DAC-10 USB IDTakashi Iwai
The USB vendor ID of NuPrime DAC-10 is not 16b0 but 16d0. Fixes: f656891c6619 ("ALSA: usb-audio: add more quirks for DSD interfaces") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200430124755.15940-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-30ALSA: opti9xx: shut up gcc-10 range warningArnd Bergmann
gcc-10 points out a few instances of suspicious integer arithmetic leading to value truncation: sound/isa/opti9xx/opti92x-ad1848.c: In function 'snd_opti9xx_configure': sound/isa/opti9xx/opti92x-ad1848.c:322:43: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_opti9xx_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow] 322 | (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/opti92x-ad1848.c:351:3: note: in expansion of macro 'snd_opti9xx_write_mask' 351 | snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); | ^~~~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/miro.c: In function 'snd_miro_configure': sound/isa/opti9xx/miro.c:873:40: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_miro_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow] 873 | (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask))) | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/miro.c:1010:3: note: in expansion of macro 'snd_miro_write_mask' 1010 | snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); | ^~~~~~~~~~~~~~~~~~~ These are all harmless here as only the low 8 bit are passed down anyway. Change the macros to inline functions to make the code more readable and also avoid the warning. Strictly speaking those functions also need locking to make the read/write pair atomic, but it seems unlikely that anyone would still run into that issue. Fixes: 1841f613fd2e ("[ALSA] Add snd-miro driver") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20200429190216.85919-1-arnd@arndb.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-29ALSA: hda/hdmi: fix without unlocked before returnWu Bo
Fix the following coccicheck warning: sound/pci/hda/patch_hdmi.c:1852:2-8: preceding lock on line 1846 After add sanity check to pass klockwork check, The spdif_mutex should be unlock before return true in check_non_pcm_per_cvt(). Fixes: 960a581e22d9 ("ALSA: hda: fix some klockwork scan warnings") Signed-off-by: Wu Bo <wubo40@huawei.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/1587907042-694161-1-git-send-email-wubo40@huawei.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-28ALSA: hda/hdmi: fix race in monitor detection during probeKai Vehmanen
A race exists between build_pcms() and build_controls() phases of codec setup. Build_pcms() sets up notifier for jack events. If a monitor event is received before build_controls() is run, the initial jack state is lost and never reported via mixer controls. The problem can be hit at least with SOF as the controller driver. SOF calls snd_hda_codec_build_controls() in its workqueue-based probe and this can be delayed enough to hit the race condition. Fix the issue by invalidating the per-pin ELD information when build_controls() is called. The existing call to hdmi_present_sense() will update the ELD contents. This ensures initial monitor state is correctly reflected via mixer controls. BugLink: https://github.com/thesofproject/linux/issues/1687 Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200428123836.24512-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-27ALSA: hda/realtek - Two front mics on a Lenovo ThinkCenterHui Wang
This new Lenovo ThinkCenter has two front mics which can't be handled by PA so far, so apply the fixup ALC283_FIXUP_HEADSET_MIC to change the location for one of the mics. Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20200427030039.10121-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-26ALSA: line6: Fix POD HD500 audio playbackVasily Khoruzhick
Apparently interface 1 is control interface akin to HD500X, setting LINE6_CAP_CONTROL and choosing it as ctrl_if fixes audio playback on POD HD500. Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200425201115.3430-1-anarsoul@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-24ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7)Takashi Iwai
[ This is again a forward-port of the fix applied for 5.6-base code (commit 4285de0725b1) to 5.7-base, hence neither Fixes nor Cc-to-stable tags are included here -- tiwai ] The checks of the plugin buffer overflow in the previous fix by commit f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") are put in the wrong places mistakenly, which leads to the expected (repeated) sound when the rate plugin is involved. Fix in the right places. Also, at those right places, the zero check is needed for the termination node, so added there as well, and let's get it done, finally. Link: https://lore.kernel.org/r/20200424193843.20397-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-24Merge tag 'sound-5.7-rc3' of ↵Linus Torvalds
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "This became a slightly big pull request, as the accumulated ASoC fixes are included here. Some highlights: - Revert of ASoC DAI startup changes that caused regression on some x86 platforms - Regression fix in HD-audio power management and driver blacklist - A collection of ASoC DAPM and topology fixes - Continued USB-audio fixes and quirks - Lots of small device-specific fixes - Rockchip S/PDIF DT stuff update for validation issues" * tag 'sound-5.7-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (51 commits) ALSA: hda: Always use jackpoll helper for jack update after resume ALSA: hda/realtek - Add new codec supported for ALC245 ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif ALSA: usb-audio: Add connector notifier delegation ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd gen ASoC: wm8960: Fix wrong clock after suspend & resume ALSA: usx2y: Fix potential NULL dereference ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2 ASoC: wm89xx: Add missing dependency ASoC: dapm: fixup dapm kcontrol widget ASoC: rsnd: Fix "status check failed" spam for multi-SSI ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent ASoC: meson: gx-card: fix codec-to-codec link setup ASoC: meson: axg-card: fix codec-to-codec link setup ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos ALSA: hda: Remove ASUS ROG Zenith from the blacklist ALSA: hda/realtek - Fix unexpected init_amp override ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell ASoC: stm32: sai: fix sai probe ...
2020-04-24ALSA: hda: Match both PCI ID and SSID for driver blacklistTakashi Iwai
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. Since the empty codec problem appear on the certain AMD platform (PCI ID 1022:1487), this patch changes the blacklist matching to both PCI ID and SSID using pci_match_id(). Also, the entry that was removed by the previous fix for ASUS ROG Zenigh II is re-added. Link: https://lore.kernel.org/r/20200424061222.19792-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-23ALSA: hda: Always use jackpoll helper for jack update after resumeTakashi Iwai
HD-audio codec driver applies a tricky procedure to forcibly perform the runtime resume by mimicking the usage count even if the device has been runtime-suspended beforehand. This was needed to assure to trigger the jack detection update after the system resume. And recently we also applied the similar logic to the HD-audio controller side. However this seems leading to some inconsistency, and eventually PCI controller gets screwed up. This patch is an attempt to fix and clean up those behavior: instead of the tricky runtime resume procedure, the existing jackpoll work is scheduled when such a forced codec resume is required. The jackpoll work will power up the codec, and this alone should suffice for the jack status update in usual cases. If the extra polling is requested (by checking codec->jackpoll_interval), the manual update is invoked after that, and the codec is powered down again. Also, we filter the spurious wake up of the codec from the controller runtime resume by checking codec->relaxed_resume flag. If this flag is set, basically we don't need to wake up explicitly, but it's supposed to be done via the audio component notifier. Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed") Link: https://lore.kernel.org/r/20200422203744.26299-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-23ALSA: hda/realtek - Add new codec supported for ALC245Kailang Yang
Enable new codec supported for ALC245. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/8c0804738b2c42439f59c39c8437817f@realtek.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-23ALSA: usb-audio: Fix usb audio refcnt leak when getting spdifXiyu Yang
snd_microii_spdif_default_get() invokes snd_usb_lock_shutdown(), which increases the refcount of the snd_usb_audio object "chip". When snd_microii_spdif_default_get() returns, local variable "chip" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in several exception handling paths of snd_microii_spdif_default_get(). When those error scenarios occur such as usb_ifnum_to_if() returns NULL, the function forgets to decrease the refcnt increased by snd_usb_lock_shutdown(), causing a refcnt leak. Fix this issue by jumping to "end" label when those error scenarios occur. Fixes: 447d6275f0c2 ("ALSA: usb-audio: Add sanity checks for endpoint accesses") Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/1587617711-13200-1-git-send-email-xiyuyang19@fudan.edu.cn Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-22ALSA: usb-audio: Add connector notifier delegationTakashi Iwai
It turned out that ALC1220-VB USB-audio device gives the interrupt event to some PCM terminals while those don't allow the connector state request but only the actual I/O terminals return the request. The recent commit 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack controls for PCM terminals") excluded those phantom terminals, so those events are ignored, too. My first thought was that this could be easily deduced from the associated terminals, but some of them have even no associate terminal ID, hence it's not too trivial to figure out. Since the number of such terminals are small and limited, this patch implements another quirk table for the simple mapping of the connectors. It's not really scalable, but let's hope that there will be not many such funky devices in future. Fixes: 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack controls for PCM terminals") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Link: https://lore.kernel.org/r/20200422113320.26664-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-21Merge tag 'asoc-fix-v5.7-rc2' of ↵Takashi Iwai
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.7 Quite a lot of fixes here, a lot of driver specific ones but the biggest one is the revert of changes to the startup and shutdown sequence for DAIs that went in during the merge window - they broke some older x86 platforms and attempts to fix them didn't succeed so it's safer to just roll them back and try to make sure those platforms are handled properly in any future attempt. The rockchip S/PDIF DT stuff was IIRC for validation issues.
2020-04-21ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd genAlexander Tsoy
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>
2020-04-21ASoC: wm8960: Fix wrong clock after suspend & resumeShengjiu Wang
After suspend & resume, wm8960_hw_params may be called when bias_level is not SND_SOC_BIAS_ON, then wm8960_configure_clocking is not called. But if sample rate is changed at that time, then the output clock rate will be not correct. So judgement of bias_level is SND_SOC_BIAS_ON in wm8960_hw_params is not necessary and it causes above issue. Fixes: 3176bf2d7ccd ("ASoC: wm8960: update pll and clock setting function") Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/1587468525-27514-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-21ALSA: usx2y: Fix potential NULL dereferenceTakashi Iwai
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>
2020-04-21ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2Gregor Pintar
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>
2020-04-20ASoC: wm89xx: Add missing dependencyYueHaibing
sound/soc/codecs/wm8900.o: In function `wm8900_i2c_probe': wm8900.c:(.text+0xa36): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8900.o: In function `wm8900_modinit': wm8900.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8900.o: In function `wm8900_exit': wm8900.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' sound/soc/codecs/wm8988.o: In function `wm8988_i2c_probe': wm8988.c:(.text+0x857): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8988.o: In function `wm8988_modinit': wm8988.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8988.o: In function `wm8988_exit': wm8988.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' sound/soc/codecs/wm8995.o: In function `wm8995_i2c_probe': wm8995.c:(.text+0x1c4f): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8995.o: In function `wm8995_modinit': wm8995.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8995.o: In function `wm8995_exit': wm8995.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' Add SND_SOC_I2C_AND_SPI dependency to fix this. Fixes: ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS") Reported-by: Hulk Robot <hulkci@huawei.com> Signed-off-by: YueHaibing <yuehaibing@huawei.com> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200420125343.20920-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-20Merge series "ASoC: rsnd: multi-SSI setup fixes" from Matthias Blankertz ↵Mark Brown
<matthias.blankertz@cetitec.com>: Fix rsnd_dai_call() operations being performed twice for the master SSI in multi-SSI setups, and fix the rsnd_ssi_stop operation for multi-SSI setups. The only visible effect of these issues was some "status check failed" spam when the rsnd_ssi_stop was called, but overall the code is cleaner now, and some questionable writes to the SSICR register which did not lead to any observable misbehaviour but were contrary to the datasheet are fixed. Mark: The first patch kind of reverts my "ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode" from a few days ago and achieves the same effect in a simpler fashion, if you would prefer a clean patch series based on v5.6 drop me a note. Greetings, Matthias Matthias Blankertz (2): ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent ASoC: rsnd: Fix "status check failed" spam for multi-SSI sound/soc/sh/rcar/ssi.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) base-commit: 15a5760cb8b6d5c1ebbf1d2e1f0b77380ab68a82 -- 2.26.1
2020-04-20Merge series "ASoC: meson: fix codec-to-codec link setup" from Jerome Brunet ↵Mark Brown
<jbrunet@baylibre.com>: This patchset fixes the problem reported by Marc in this thread [0] The problem was due to an error in the meson card drivers which had the "no_pcm" dai_link property set on codec-to-codec links [0]: https://lore.kernel.org/r/20200417122732.GC5315@sirena.org.uk Jerome Brunet (2): ASoC: meson: axg-card: fix codec-to-codec link setup ASoC: meson: gx-card: fix codec-to-codec link setup sound/soc/meson/axg-card.c | 4 +++- sound/soc/meson/gx-card.c | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) -- 2.25.2
2020-04-20ASoC: dapm: fixup dapm kcontrol widgetGyeongtaek Lee
snd_soc_dapm_kcontrol widget which is created by autodisable control should contain correct on_val, mask and shift because it is set when the widget is powered and changed value is applied on registers by following code in dapm_seq_run_coalesced(). mask |= w->mask << w->shift; if (w->power) value |= w->on_val << w->shift; else value |= w->off_val << w->shift; Shift on the mask in dapm_kcontrol_data_alloc() is removed to prevent double shift. And, on_val in dapm_kcontrol_set_value() is modified to get correct value in the dapm_seq_run_coalesced(). Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com> Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/000001d61537$b212f620$1638e260$@samsung.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-20ASoC: rsnd: Fix "status check failed" spam for multi-SSIMatthias Blankertz
Fix the rsnd_ssi_stop function to skip disabling the individual SSIs of a multi-SSI setup, as the actual stop is performed by rsnd_ssiu_stop_gen2 - the same logic as in rsnd_ssi_start. The attempt to disable these SSIs was harmless, but caused a "status check failed" message to be printed for every SSI in the multi-SSI setup. The disabling of interrupts is still performed, as they are enabled for all SSIs in rsnd_ssi_init, but care is taken to not accidentally set the EN bit for an SSI where it was not set by rsnd_ssi_start. Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20200417153017.1744454-3-matthias.blankertz@cetitec.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-20ASoC: rsnd: Don't treat master SSI in multi SSI setup as parentMatthias Blankertz
The master SSI of a multi-SSI setup was attached both to the RSND_MOD_SSI slot and the RSND_MOD_SSIP slot of the rsnd_dai_stream. This is not correct wrt. the meaning of being "parent" in the rest of the SSI code, where it seems to indicate an SSI that provides clock and word sync but is not transmitting/receiving audio data. Not treating the multi-SSI master as parent allows removal of various special cases to the rsnd_ssi_is_parent conditions introduced in commit a09fb3f28a60 ("ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode"). It also fixes the issue that operations performed via rsnd_dai_call() were performed twice for the master SSI. This caused some "status check failed" spam when stopping a multi-SSI stream as the driver attempted to stop the master SSI twice. Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20200417153017.1744454-2-matthias.blankertz@cetitec.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-20ASoC: meson: gx-card: fix codec-to-codec link setupJerome Brunet
Since the addition of commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported"), meson-axg cards which have codec-to-codec links fail to init and Oops. Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 Internal error: Oops: 96000044 [#1] PREEMPT SMP CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 pc : invalidate_paths_ep+0x30/0xe0 lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 Call trace: invalidate_paths_ep+0x30/0xe0 snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 dpcm_path_get+0x38/0xd0 dpcm_fe_dai_open+0x70/0x920 snd_pcm_open_substream+0x564/0x840 snd_pcm_open+0xfc/0x228 snd_pcm_capture_open+0x4c/0x78 snd_open+0xac/0x1a8 ... While this error was initially reported the axg-card type, it also applies to the gx-card type. While initiliazing the links, ASoC treats the codec-to-codec links of this card type as a DPCM backend. This error eventually leads to the Oops. Most of the card driver code is shared between DPCM backends and codec-to-codec links. The property "no_pcm" marking DCPM BE was left set on codec-to-codec links, leading to this problem. This commit fixes that. Fixes: e37a0c313a0f ("ASoC: meson: gx: add sound card support") Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20200420114511.450560-3-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-20ASoC: meson: axg-card: fix codec-to-codec link setupJerome Brunet
Since the addition of commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported"), meson-axg cards which have codec-to-codec links fail to init and Oops: Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 Internal error: Oops: 96000044 [#1] PREEMPT SMP CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 pc : invalidate_paths_ep+0x30/0xe0 lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 Call trace: invalidate_paths_ep+0x30/0xe0 snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 dpcm_path_get+0x38/0xd0 dpcm_fe_dai_open+0x70/0x920 snd_pcm_open_substream+0x564/0x840 snd_pcm_open+0xfc/0x228 snd_pcm_capture_open+0x4c/0x78 snd_open+0xac/0x1a8 ... While initiliazing the links, ASoC treats the codec-to-codec links of this card type as a DPCM backend. This error eventually leads to the Oops. Most of the card driver code is shared between DPCM backends and codec-to-codec links. The property "no_pcm" marking DCPM BE was left set on codec-to-codec links, leading to this problem. This commit fixes that. Fixes: 0a8f1117a680 ("ASoC: meson: axg-card: add basic codec-to-codec link support") Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20200420114511.450560-2-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-20ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobosTakashi Iwai
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>
2020-04-19ALSA: hda: Remove ASUS ROG Zenith from the blacklistTakashi Iwai
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>
2020-04-18ALSA: hda/realtek - Fix unexpected init_amp overrideTakashi Iwai
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>
2020-04-18ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devicesAlexander Tsoy
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>
2020-04-17ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/BroadwellPierre-Louis Bossart
Major regressions were detected by SOF CI on CherryTrail and Broadwell: [ 25.705750] SSP2-Codec: ASoC: no backend playback stream [ 27.923378] SSP2-Codec: ASoC: no users playback at close - state This is root-caused to the introduction of the DAI capability checks with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a requirement for all DAIs to report at least a non-zero min_channels field. For some reason the SSP structures used for SKL+ did provide this information but legacy platforms didn't. Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200417172014.11760-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-17ASoC: stm32: sai: fix sai probeOlivier Moysan
pcm config must be set before snd_dmaengine_pcm_register() call. Fixes: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") Signed-off-by: Olivier Moysan <olivier.moysan@st.com> Link: https://lore.kernel.org/r/20200417142122.10212-1-olivier.moysan@st.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-17Merge tag 'sound-5.7-rc2' of ↵Linus Torvalds
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
2020-04-17ALSA: hda/hdmi: Add module option to disable audio component bindingTakashi Iwai
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>
2020-04-16Merge series "ASoC: rsnd: Fixes for multichannel HDMI audio output" from ↵Mark Brown
Matthias Blankertz <matthias.blankertz@cetitec.com>: This fixes two issues in the snd-soc-rcar driver blocking multichannel HDMI audio out: The parent SSI in a multi-SSI configuration is not correctly set up and started, and the SSI->HDMI channel mapping is wrong. With these patches, the following device tree snippet can be used on an r8a7795-based platform (Salvator-X) to enable multichannel HDMI audio on HDMI0: rsnd_port1: port@1 { rsnd_endpoint1: endpoint { remote-endpoint = <&dw_hdmi0_snd_in>; dai-format = "i2s"; bitclock-master = <&rsnd_endpoint1>; frame-master = <&rsnd_endpoint1>; playback = <&ssi0 &ssi1 &ssi2 &ssi9>; }; }; With a capable receiver attached, all of 2ch (stereo), 6ch (e.g. 5.1) and 8ch audio output should work. Matthias Blankertz (2): ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode sound/soc/sh/rcar/ssi.c | 8 ++++---- sound/soc/sh/rcar/ssiu.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) base-commit: 7111951b8d4973bda27ff663f2cf18b663d15b48 -- 2.26.0
2020-04-16ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entryAmadeusz Sławiński
If we don't find any pcm, pcm will point at address at an offset from the the list head and not a meaningful structure. Fix this by returning correct pcm if found and NULL if not. Found with coccinelle. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20200415162849.308-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-16ASoC: rsnd: Fix HDMI channel mapping for multi-SSI modeMatthias Blankertz
The HDMI?_SEL register maps up to four stereo SSI data lanes onto the sdata[0..3] inputs of the HDMI output block. The upper half of the register contains four blocks of 4 bits, with the most significant controlling the sdata3 line and the least significant the sdata0 line. The shift calculation has an off-by-one error, causing the parent SSI to be mapped to sdata3, the first multi-SSI child to sdata0 and so forth. As the parent SSI transmits the stereo L/R channels, and the HDMI core expects it on the sdata0 line, this causes no audio to be output when playing stereo audio on a multichannel capable HDMI out, and multichannel audio has permutated channels. Fix the shift calculation to map the parent SSI to sdata0, the first child to sdata1 etc. Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20200415141017.384017-3-matthias.blankertz@cetitec.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-16ASoC: rsnd: Fix parent SSI start/stop in multi-SSI modeMatthias Blankertz
The parent SSI of a multi-SSI setup must be fully setup, started and stopped since it is also part of the playback/capture setup. So only skip the SSI (as per commit 203cdf51f288 ("ASoC: rsnd: SSI parent cares SWSP bit") and commit 597b046f0d99 ("ASoC: rsnd: control SSICR::EN correctly")) if the SSI is parent outside of a multi-SSI setup. Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20200415141017.384017-2-matthias.blankertz@cetitec.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-15ASoC: soc-dai: revert all changes to DAI startup/shutdown sequencePierre-Louis Bossart
On Baytrail/Cherrytrail, the Atom/SST driver fails miserably: [ 9.741953] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01 [ 9.832992] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.833019] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.833028] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.833033] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.833037] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed [ 9.853942] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.853974] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.853984] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.853990] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.853994] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed Commit b56be800f1292 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once") was the initial problematic commit. Commit 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence") was an attempt to fix things but it does not work on Baytrail, reverting all changes seems necessary for now. Fixes: 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Tested-by: Hans de Goede <hdegoede@redhat.com> Link: https://lore.kernel.org/r/20200415030437.23803-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-15ASoC: topology: Fix endianness issueAmadeusz Sławiński
As done in already existing cases, we should use le32_to_cpu macro while accessing hdr->magic. Found with sparse. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20200415162435.31859-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-15ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIsStephan Gerhold
For some reason, the MI2S DAIs do not have channels_min/max defined. This means that snd_soc_dai_stream_valid() returns false, i.e. the DAIs have neither valid playback nor capture stream. It's quite surprising that this ever worked correctly, but in 5.7-rc1 this is now failing badly: :) Commit 0e9cf4c452ad ("ASoC: pcm: check if cpu-dai supports a given stream") introduced a check for snd_soc_dai_stream_valid() before calling hw_params(), which means that the q6i2s_hw_params() function was never called, eventually resulting in: qcom-q6afe aprsvc:q6afe:4:4: no line is assigned ... even though "qcom,sd-lines" is set in the device tree. Commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") now even avoids creating PCM devices if the stream is not supported, which means that it is failing even earlier with e.g.: Primary MI2S: ASoC: no backend playback stream Avoid all that trouble by adding channels_min/max for the MI2S DAIs. Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe 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/20200415150050.616392-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-15ASoC: soc-pcm: dpcm: Only allow playback/capture if supportedStephan Gerhold
At the moment, PCM devices for DPCM are only created based on the dpcm_playback/capture parameters of the DAI link, without considering if the CPU/FE DAI is actually capable of playback/capture. Normally the dpcm_playback/capture parameter should match the capabilities of the CPU DAI. However, there is no way to set that parameter from the device tree (e.g. with simple-audio-card or qcom sound cards). dpcm_playback/capture are always both set to 1. This causes problems when the CPU DAI does only support playback or capture. Attemting to open that PCM device with an unsupported stream type then results in a null pointer dereference: Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 Internal error: Oops: 96000044 [#1] PREEMPT SMP CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 pc : invalidate_paths_ep+0x30/0xe0 lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 Call trace: invalidate_paths_ep+0x30/0xe0 snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 dpcm_path_get+0x38/0xd0 dpcm_fe_dai_open+0x70/0x920 snd_pcm_open_substream+0x564/0x840 snd_pcm_open+0xfc/0x228 snd_pcm_capture_open+0x4c/0x78 snd_open+0xac/0x1a8 ... ... because the DAI playback/capture_widget is not set in that case. We could add checks there to fix the problem (maybe we should anyway), but much easier is to not expose the device as playback/capture in the first place. Attemting to use that device would always fail later anyway. Add checks for snd_soc_dai_stream_valid() to the DPCM case to avoid exposing playback/capture if it is not supported. Signed-off-by: Stephan Gerhold <stephan@gerhold.net> Link: https://lore.kernel.org/r/20200415104928.86091-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-15ASoC: sgtl5000: Fix VAG power-on handlingSebastian Reichel
As mentioned slightly out of patch context in the code, there is no reset routine for the chip. On boards where the chip is supplied by a fixed regulator, it might not even be resetted during (e.g. watchdog) reboot and can be in any state. If the device is probed with VAG enabled, the driver's probe routine will generate a loud pop sound when ANA_POWER is being programmed. Avoid this by properly disabling just the VAG bit and waiting the required power down time. Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.com> Reviewed-by: Fabio Estevam <festivem@gmail.com> Link: https://lore.kernel.org/r/20200414181140.145825-1-sebastian.reichel@collabora.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-14ALSA: hda: call runtime_allow() for all hda controllersHui Wang
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>
2020-04-14ASoC: madera: Remove a couple of stray blank linesCharles Keepax
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200409181311.30247-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-14ASoC: wsa881x: mark read_only_wordlength flagSrinivas Kandagatla
WSA881x works in PDM mode so the wordlength is fixed, which also makes the only field "WordLength" in DPN_BlockCtrl1 register a read-only. Writing to this register will throw up errors with Qualcomm Controller. So use ro_blockctrl1_reg flag to mark this field as read-only so that core will not write to this register. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20200414110347.23829-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-14ASoC: qcom: common: Silence duplicate parse error messagesBjorn Andersson
All error paths in qcom_snd_parse_of() prints more specific error messages, so silence the one in apq8096_platform_probe() and sdm845_snd_platform_probe() to avoid spamming the kernel log. Signed-off-by: Bjorn Andersson <bjorn.andersson@linaro.org> Link: https://lore.kernel.org/r/20200406003229.2354631-1-bjorn.andersson@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-14soc/stm/stm32_sub_sai: Add missing '\n' in log messagesSebastian Fricke
Message logged by 'dev_xxx()' or 'pr_xxx()' should end with a '\n'. Fixes: 3e086ed("ASoC: stm32: add SAI drivers") Signed-off-by: Sebastian Fricke <sebastian.fricke.linux@gmail.com> Link: https://lore.kernel.org/r/20200413042952.7675-1-sebastian.fricke.linux@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-14ASoC: dapm: Remove dapm_connect_dai_link_widgets helperCharles Keepax
This helper is adding very little both it and is one caller are very small functions simply combine the two. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200409181209.30130-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>