- 25 Aug, 2020 2 commits
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Mohan Kumar authored
The Tegra HDA codec HW implementation has an issue related to not swapping the 2 channel Audio Sample Packet(ASP) channel mapping. Whatever the FL and FR mapping specified the left channel always comes out of left speaker and right channel on right speaker. So add condition to disallow the swapping of FL,FR during the playback. Signed-off-by:
Mohan Kumar <mkumard@nvidia.com> Acked-by:
Sameer Pujar <spujar@nvidia.com> Link: https://lore.kernel.org/r/20200825052415.20626-2-mkumard@nvidia.comSigned-off-by:
Takashi Iwai <tiwai@suse.de>
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Tong Zhang authored
snd_ca0106_spi_write() returns 1 on error, snd_ca0106_pcm_power_dac() is returning the error code directly, and the caller is expecting an negative error code Signed-off-by:
Tong Zhang <ztong0001@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200824224541.1260307-1-ztong0001@gmail.comSigned-off-by:
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- 24 Aug, 2020 1 commit
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Randy Dunlap authored
Fix documentation build warnings for underline length too short, caused by s/http/https/ and not changing the accompanying underlines. Documentation/sound/cards/audigy-mixer.rst:335: WARNING: Title underline too short. US Patents (https://www.uspto.gov/) ---------------------------------- Documentation/sound/cards/sb-live-mixer.rst:340: WARNING: Title underline too short. US Patents (https://www.uspto.gov/) ---------------------------------- Fixes: 7ed33ea6 ("ALSA: Replace HTTP links with HTTPS ones") Signed-off-by:
Randy Dunlap <rdunlap@infradead.org> Cc: Alexander A. Klimov <grandmaster@al2klimov.de> Link: https://lore.kernel.org/r/357ee576-32a2-6e2b-1db6-78be39253846@infradead.orgSigned-off-by:
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- 23 Aug, 2020 3 commits
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Kai-Heng Feng authored
If USB autosuspend is enabled, both front and rear panel can no longer detect jack insertion. Enable USB remote wakeup, i.e. needs_remote_wakeup = 1, doesn't help either. So disable USB autosuspend to prevent missing jack detection event. Signed-off-by:
Kai-Heng Feng <kai.heng.feng@canonical.com> Link: https://lore.kernel.org/r/20200823105854.26950-1-kai.heng.feng@canonical.comSigned-off-by:
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Takashi Sakamoto authored
Avid Adrenaline is reported that ALSA firewire-digi00x driver is bound to. However, as long as he investigated, the design of this model is hardly similar to the one of Digi 00x family. It's better to exclude the model from modalias of ALSA firewire-digi00x driver. This commit changes device entries so that the model is excluded. $ python3 crpp < ~/git/am-config-rom/misc/avid-adrenaline.img ROM header and bus information block ----------------------------------------------------------------- 400 04203a9c bus_info_length 4, crc_length 32, crc 15004 404 31333934 bus_name "1394" 408 e064a002 irmc 1, cmc 1, isc 1, bmc 0, cyc_clk_acc 100, max_rec 10 (2048) 40c 00a07e01 company_id 00a07e | 410 00085257 device_id 0100085257 | EUI-64 00a07e0100085257 root directory ----------------------------------------------------------------- 414 0005d08c directory_length 5, crc 53388 418 0300a07e vendor 41c 8100000c --> descriptor leaf at 44c 420 0c008380 node capabilities 424 8d000002 --> eui-64 leaf at 42c 428 d1000004 --> unit directory at 438 eui-64 leaf at 42c ----------------------------------------------------------------- 42c 0002410f leaf_length 2, crc 16655 430 00a07e01 company_id 00a07e | 434 00085257 device_id 0100085257 | EUI-64 00a07e0100085257 unit directory at 438 ----------------------------------------------------------------- 438 0004d6c9 directory_length 4, crc 54985 43c 1200a02d specifier id: 1394 TA 440 13014001 version: Vender Unique and AV/C 444 17000001 model 448 81000009 --> descriptor leaf at 46c descriptor leaf at 44c ----------------------------------------------------------------- 44c 00077205 leaf_length 7, crc 29189 450 00000000 textual descriptor 454 00000000 minimal ASCII 458 41766964 "Avid" 45c 20546563 " Tec" 460 686e6f6c "hnol" 464 6f677900 "ogy" 468 00000000 descriptor leaf at 46c ----------------------------------------------------------------- 46c 000599a5 leaf_length 5, crc 39333 470 00000000 textual descriptor 474 00000000 minimal ASCII 478 41647265 "Adre" 47c 6e616c69 "nali" 480 6e650000 "ne" Reported-by:Simon Wood <simon@mungewell.org> Fixes: 9edf723f ("ALSA: firewire-digi00x: add skeleton for Digi 002/003 family") Cc: <stable@vger.kernel.org> # 4.4+ Signed-off-by:
Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20200823075545.56305-1-o-takashi@sakamocchi.jpSigned-off-by:
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Takashi Sakamoto authored
Tascam FE-8 is known to support communication by asynchronous transaction only. The support can be implemented in userspace application and snd-firewire-ctl-services project has the support. However, ALSA firewire-tascam driver is bound to the model. This commit changes device entries so that the model is excluded. In a commit 53b3ffee ("ALSA: firewire-tascam: change device probing processing"), I addressed to the concern that version field in configuration differs depending on installed firmware. However, as long as I checked, the version number is fixed. It's safe to return version number back to modalias. Fixes: 53b3ffee ("ALSA: firewire-tascam: change device probing processing") Cc: <stable@vger.kernel.org> # 4.4+ Signed-off-by:
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- 19 Aug, 2020 3 commits
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Sameer Pujar authored
By default 'sdo_limit' is initialized with a default value of '8' as per spec. This is overridden in cases where a different value is required. However this is getting reset when snd_hdac_bus_init_chip() is called again, which happens during runtime PM cycle. Avoid this reset by moving 'sdo_limit' setup to 'snd_hdac_bus_init()' function which would be called only once. Fixes: 67ae482a ("ALSA: hda: add member to store ratio for stripe control") Cc: <stable@vger.kernel.org> Signed-off-by:
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Mike Pozulp authored
The Galaxy Book Ion uses the same ALC298 codec as other Samsung laptops which have the no headphone sound bug, like my Samsung Notebook. The Galaxy Book owner confirmed that this patch fixes the bug. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423Signed-off-by:
Mike Pozulp <pozulp.kernel@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200818165446.499821-1-pozulp.kernel@gmail.comSigned-off-by:
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Takashi Iwai authored
Merge tag 'asoc-fix-v5.9-rc1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.9 A bunch of fixes that came in during the merge window, mostly for issues that were uncovered by the changes to report errors on invalid register access plus one important fix in that code itself.
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- 17 Aug, 2020 7 commits
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Tom Yan authored
Some devices have broken extension unit where getting current value doesn't work. Attempt that once when creating mixer control for it. If it fails, just ignore it, so that it won't cripple the device entirely (and/or make the error floods). Signed-off-by:
Tom Yan <tom.ty89@gmail.com> Link: https://lore.kernel.org/r/5f3abc52.1c69fb81.9cf2.fe91@mx.google.comSigned-off-by:
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Dinghao Liu authored
When power_up_sst() fails, stream needs to be freed just like when try_module_get() fails. However, current code is returning directly and ends up leaking memory. Fixes: 0121327c ("ASoC: Intel: mfld-pcm: add control for powering up/down dsp") Signed-off-by:
Dinghao Liu <dinghao.liu@zju.edu.cn> Acked-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200813084112.26205-1-dinghao.liu@zju.edu.cnSigned-off-by:
Mark Brown <broonie@kernel.org>
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Sylwester Nawrocki authored
The driver supports WM1811, WM8994, WM8958 devices but according to documentation and the regmap definitions the WM8958_DSP2_* registers are only available on WM8958. In current code these registers are being accessed as if they were available on all the three chips. When starting playback on WM1811 CODEC multiple errors like: "wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5" can be seen, which is caused by attempts to read an unavailable WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent commit "e2329eeb ASoC: soc-component: add soc_component_err()". This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling is only done for WM8958. Signed-off-by:
Sylwester Nawrocki <s.nawrocki@samsung.com> Acked-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200731173834.23832-1-s.nawrocki@samsung.comSigned-off-by:
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Srinivas Kandagatla authored
For some reason interrupt set and clear register offsets are not set correctly. This patch corrects them! Fixes: 585e881e ("ASoC: codecs: Add msm8916-wcd analog codec") Signed-off-by:
Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Tested-by:
Stephan Gerhold <stephan@gerhold.net> Reviewed-by:
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Sylwester Nawrocki authored
The ADC2 and DAC2 are not available on WM1811 device. This patch moves the ADC2, DAC2 VU bitfields to a separate array so we can skip accessing them and avoid unreadable register access on WM1811. This allows to get rid of warnings during boot like: wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5 Signed-off-by:
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Mike Pozulp authored
The very quiet and distorted headphone output bug that afflicted my Samsung Notebook 9 is appearing in many other Samsung laptops. Expose the quirk which fixed my laptop as a model so other users can try it. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423Signed-off-by:
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Hector Martin authored
As the recent fix addressed the channel swap problem more properly, update the comment as well. Fixes: 1b7ecc24 ("ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109") Signed-off-by:
Hector Martin <marcan@marcan.st> Link: https://lore.kernel.org/r/20200816084431.102151-1-marcan@marcan.stSigned-off-by:
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- 16 Aug, 2020 1 commit
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Liang Wang authored
Fix spelling mistakes in the comments: initailise ==> initialise tranfer ==> transfer Initialse ==> Initialise Signed-off-by:
Liang Wang <wangliang101@huawei.com> Link: https://lore.kernel.org/r/20200816071309.121461-1-wangliang101@huawei.comSigned-off-by:
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- 15 Aug, 2020 1 commit
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Alexander Tsoy authored
Capture and playback endpoints on Saffire 6 (USB 1.1) resides on the same interface. This was not supported by the composite quirk back in the day when initial support for this device was added, thus only playback was enabled until now. Fixes: 11e424e8 ("ALSA: usb-audio: Add support for Focusrite Saffire 6 USB") Signed-off-by:
Alexander Tsoy <alexander@tsoy.me> Cc: <stable.vger.kernel.org> Link: https://lore.kernel.org/r/20200815002103.29247-1-alexander@tsoy.meSigned-off-by:
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- 14 Aug, 2020 1 commit
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Mike Pozulp authored
The Flex Book uses the same ALC298 codec as other Samsung laptops which have the no headphone sound bug, like my Samsung Notebook. The Flex Book owner used Early Patching to confirm that this quirk fixes the bug. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423Signed-off-by:
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- 13 Aug, 2020 1 commit
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Dinghao Liu authored
Freeing chip on error may lead to an Oops at the next time the system goes to resume. Fix this by removing all snd_echo_free() calls on error. Fixes: 47b5d028 ("ALSA: Echoaudio - Add suspend support #2") Signed-off-by:
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- 12 Aug, 2020 2 commits
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Kai-Heng Feng authored
There's another HP desktop has buggy BIOS which flags the Port Connectivity bit as no connection. Apply force connectivity quirk to enable DP/HDMI audio. Signed-off-by:
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Takashi Iwai authored
The previous fix forgot to remove the unused variable that triggers a compile warning now: sound/pci/hda/patch_realtek.c: In function 'alc285_fixup_hp_gpio_led': sound/pci/hda/patch_realtek.c:4163:19: warning: unused variable 'spec' [-Wunused-variable] Fix it. Fixes: 40469064 ("ALSA: hda - reverse the setting value in the micmute_led_set") Reported-by:
Stephen Rothwell <sfr@canb.auug.org.au> Link: https://lore.kernel.org/r/20200812070256.32145-1-tiwai@suse.deSigned-off-by:
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- 11 Aug, 2020 5 commits
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Srinivas Kandagatla authored
Most of the DAPM widgets for DSP ASoC components reuse reg field of the widgets for its internal calculations, however these are not real registers. So read/writes to these numbers are not really valid. However ASoC core will read these registers to get default state during startup. With recent changes to ASoC core, every register read/write failures are reported very verbosely. Prior to this fails to reads are totally ignored, so we never saw any error messages. To fix this add dummy read/write function to return default value. Fixes: e3a33673 ("ASoC: qdsp6: q6routing: Add q6routing driver") Reported-by:
John Stultz <john.stultz@linaro.org> Signed-off-by:
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Srinivas Kandagatla authored
Looks like the q6afe-dai dapm widget registers are set as "0", which is a not correct. As this registers will be read by ASoC core during startup which will throw up errors, Fix this by making the registers as SND_SOC_NOPM as these should be never used. With recent changes to ASoC core, every register read/write failures are reported very verbosely. Prior to this fails to reads are totally ignored, so we never saw any error messages. Fixes: 24c4cbcf ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Reported-by:
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Takashi Iwai authored
Along with the recent unification of snd_soc_component_read*() functions, the behavior of snd_soc_component_read() was changed slightly; namely it returns the register read value directly, and even if an error happens, it returns zero (but it prints an error message). That said, the caller side can't know whether it's an error or not any longer. Ideally this shouldn't matter much, but in practice this seems causing a regression, as John reported. And, grepping the tree revealed that there are still plenty of callers that do check the error code, so we'll need to deal with them in anyway. As a quick band-aid over the regression, this patch changes the return value of snd_soc_component_read() again to the negative error code. It can't work, obviously, for 32bit register values, but it should be enough for the known regressions, so far. Fixes: cf6e26c7 ("ASoC: soc-component: merge snd_soc_component_read() and snd_soc_component_read32()") Reported-by:
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Hui Wang authored
Before the micmute_led_set() is introduced, the function of alc_gpio_micmute_update() will set the gpio value with the !micmute_led.led_value, and the machines have the correct micmute led status. After the micmute_led_set() is introduced, it sets the gpio value with !!micmute_led.led_value, so the led status is not correct anymore, we need to set micmute_led_polarity = 1 to workaround it. Now we fix the micmute_led_set() and remove micmute_led_polarity = 1. Fixes: 87dc3648 ("ALSA: hda/realtek - Add LED class support for micmute LED") Reported-and-suggested-by:
Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20200811122430.6546-1-hui.wang@canonical.comSigned-off-by:
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Takashi Iwai authored
The dsp_registers field of struct echoaduio has the volatile modifier, but it's basically superfluous; the field is accessed only for the base pointer of readl() and writel(), hence marking with __iomem alone should suffice. OTOH, having the volatile prefix causes a compile warning like: sound/pci/echoaudio/echoaudio.c:1878:14: warning: passing argument 1 of 'iounmap' discards 'volatile' qualifier from pointer target type [-Wdiscarded-qualifiers] So it's better to drop this superfluous modifier. Link: https://lore.kernel.org/r/20200803143958.24324-1-tiwai@suse.deSigned-off-by:
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- 10 Aug, 2020 6 commits
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Ravulapati Vishnu vardhan rao authored
Replacing string compare with "codec_dai->name" instead of comparing with "codec_dai->component->name" in hw_params because, Here the component name for codec RT1015 is "i2c-10EC5682:00" and will never be "rt1015-aif1" as it is codec-dai->name. So, strcmp() always compares and fails to set the sysclk,pll,bratio for expected codec-dai="rt1015-aif1". Signed-off-by:
Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com> Link: https://lore.kernel.org/r/20200807161046.17932-1-Vishnuvardhanrao.Ravulapati@amd.comSigned-off-by:
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Kai-Heng Feng authored
The USB device (0x17aa:0x1046) that support Lenovo P620 rear panel line-in claim to support volume control, but it doens't seem to have an AMP, so when line-in volume lowers below 80, nothing gets recorded anymore. Disable the volume control to workaround the issue. Fixes: f8c11eb7 ("ALSA: usb-audio: Add support for Lenovo ThinkStation P620") Signed-off-by:
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Hector Martin authored
This is just another Pioneer device with fixed endpoints. Input is dummy but used as feedback (it always returns silence). Cc: stable@vger.kernel.org Signed-off-by:
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Hector Martin authored
Further investigation of the L-R swap problem on the MS2109 reveals that the problem isn't that the channels are swapped, but rather that they are swapped and also out of phase by one sample. In other words, the issue is actually that the very first frame that comes from the hardware is a half-frame containing only the right channel, and after that everything becomes offset. So introduce a new quirk field to drop the very first 2 bytes that come in after the format is configured and a capture stream starts. This puts the channels in phase and in the correct order. Cc: stable@vger.kernel.org Signed-off-by:
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Hui Wang authored
After installing the Ubuntu Linux, the micmute led status is not correct. Users expect that the led is on if the capture is disabled, but with the current kernel, the led is off with the capture disabled. We tried the old linux kernel like linux-4.15, there is no this issue. It looks like we introduced this issue when switching to the led_cdev. Cc: <stable@vger.kernel.org> Signed-off-by:
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Hector Martin authored
Matching by device matches all interfaces, which breaks the video/HID portions of the device depending on module load order. Fixes: e337bf19 ("ALSA: usb-audio: add quirk for MacroSilicon MS2109") Cc: stable@vger.kernel.org Signed-off-by:
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- 07 Aug, 2020 1 commit
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Kai-Heng Feng authored
The jack on Intel NUC 8 Rugged rear panel doesn't work. The spec [1] states that the jack supports both headphone and microphone, so override a Pin Complex which has both Amp-In and Amp-Out to make the jack work. Node 0x1b fits the requirement, and user confirmed the jack now works with new pin config. [1] https://www.intel.com/content/dam/support/us/en/documents/mini-pcs/NUC8CCH_TechProdSpec.pdf BugLink: https://bugs.launchpad.net/bugs/1875199Signed-off-by:
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- 06 Aug, 2020 4 commits
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Mirko Dietrich authored
Adds an entry for Creative USB X-Fi to the rc_config array in mixer_quirks.c to allow use of volume knob on the device. Adds support for newer X-Fi Pro card, known as "Model No. SB1095" with USB ID "041e:3263" Signed-off-by:
Mirko Dietrich <buzz@l4m1.de> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200806124850.20334-1-buzz@l4m1.deSigned-off-by:
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Colin Ian King authored
There is a spelling mistake in a usb_audio_dbg debug message. Also replace "param" with "parameter". Fix these. Signed-off-by:
Colin Ian King <colin.king@canonical.com> Link: https://lore.kernel.org/r/20200806105134.46447-1-colin.king@canonical.comSigned-off-by:
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Randy Dunlap authored
Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: -
Randy Dunlap authored
Drop duplicated words in sound/isa/. {be, bit} Signed-off-by:
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- 05 Aug, 2020 2 commits
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Mohan Kumar authored
Tegra HDA has audio data buffer for upto tens of frames, this buffer can help to avoid underflow. HW will keep issuing new data fetch request when buffers are not full and current BDL is not done. When SW disable DMA RUN bit for a stream, HW can't cancel the already issued data fetch request and hence it can't stop DMA. HW has to wait for all issued data fetch request get data returned before it stops DMA. This HW behavior is not in sync with HDA spec which says DMA RUN bit should be cleared within 1 audio frame. For Tegra, DMA RUN bit was active for more than one audio frame, due to this the timeout in snd_hdac_stream_sync function is not helping. When Stream reset set and clear happens during DMA RUN bit active state it results in Memory Decode error. Unfortunately, there is no way to detect when these data accesses have completed, but testing has shown that a 100us delay between Stream reset set and clear operation for Tegra avoids the memory decode error. Therefore, adding a 100us dma stop delay. Signed-off-by:
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Mohan Kumar authored
A variable dma_stop_delay is added as a new member in hdac_bus structure to avoid memory decode error incase DMA RUN bit is not disabled in the given timeout from snd_hdac_stream_sync function and followed by stream reset which results in memory decode error between reset set and clear operation. Signed-off-by:
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