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The fifo_depth is 64 on i.MX8QM/i.MX8QXP, 128 on i.MX8MQ, 16 on
i.MX7ULP.
Original FSL_SAI_CR1_RFW_MASK value 0x1F is not suitable for
these platform, the FIFO watermark mask should be updated
according to the fifo_depth.
Fixes: a860fac42097 ("ASoC: fsl_sai: Add support for imx7ulp/imx8mq")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/1596176895-28724-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Commit b73287f0b074 ("ASoC: soc-pcm: dpcm: fix playback/capture checks")
changed the meaning of dpcm_playback/dpcm_capture and now requires the
CPU DAI BE to aligned with those flags.
This broke all Amlogic cards with uni-directional backends (All gx and
most axg cards).
While I'm still confused as to how this change is an improvement, those
cards can't remain broken forever. Hopefully, next time an API change is
done like that, all the users will be updated as part of the change, and
not left to fend for themselves.
Fixes: b73287f0b074 ("ASoC: soc-pcm: dpcm: fix playback/capture checks")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200731120603.2243261-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Previous updates to set dailink capabilities and check dailink
capabilities were based on a flawed assumption that all dais support
the same capabilities as the dailink. This is true for TDM
configurations but existing configurations use an amplifier and a
capture device on the same dailink, and the tests would prevent the
card from probing.
This patch modifies the snd_soc_dai_link_set_capabilities()
helper so that the dpcm_playback (resp. dpcm_capture) dailink
capabilities are set if at least one dai supports playback (resp. capture).
Likewise the checks are modified so that an error is reported only
when dpcm_playback (resp. dpcm_capture) is set but none of the CPU
DAIs support playback (resp. capture).
Fixes: 25612477d20b5 ('ASoC: soc-dai: set dai_link dpcm_ flags with a helper')
Fixes: b73287f0b0745 ('ASoC: soc-pcm: dpcm: fix playback/capture checks')
Suggested-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200723180533.220312-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patcheset is collection of fixes for the TDM input and output the
axg audio architecture. Its fixes:
- slave mode format setting
- g12 and sm1 skew offset
- tdm clock inversion
- standard daifmt props names which don't require a specific prefix
Jerome Brunet (4):
ASoC: meson: axg-tdm-interface: fix link fmt setup
ASoC: meson: axg-tdmin: fix g12a skew
ASoC: meson: axg-tdm-formatters: fix sclk inversion
ASoC: meson: cards: remove DT_PREFIX for standard daifmt properties
sound/soc/meson/axg-tdm-formatter.c | 11 ++++++-----
sound/soc/meson/axg-tdm-formatter.h | 1 -
sound/soc/meson/axg-tdm-interface.c | 26 +++++++++++++++++---------
sound/soc/meson/axg-tdmin.c | 16 +++++++++++++++-
sound/soc/meson/axg-tdmout.c | 3 ---
sound/soc/meson/meson-card-utils.c | 2 +-
6 files changed, 39 insertions(+), 20 deletions(-)
--
2.25.4
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The allocation order of things in soc_new_pcm_runtime was changed to
move the device_register before the allocation of the rtd structure.
This was to allow the rtd allocation to be managed by devm. However
currently the sysfs entries are added by device_register and their
visibility depends on variables within the rtd structure, this causes
the pmdown_time and dapm_widgets sysfs entries to be missing for all
rtds.
Correct this issue by manually calling device_add_groups after the
appropriate information is available.
Fixes: d918a37610b1 ("ASoC: soc-core: tidyup soc_new_pcm_runtime() alloc order")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200730120715.637-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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After carefully checking, it appears that both tdmout and tdmin require the
rising edge of the sclk they get to be synchronized with the frame sync
event (which should be a rising edge of lrclk).
TDMIN was improperly set before this patch. Remove the sclk_invert quirk
which is no longer needed and fix the sclk phase.
Fixes: 1a11d88f499c ("ASoC: meson: add tdm formatter base driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-4-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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After carefully checking the result provided by the TDMIN on the g12a and
sm1 SoC families, the TDMIN skew offset appears to be 3 instead of 2 on the
axg.
Fixes: f01bc67f58fd ("ASoC: meson: axg-tdm-formatter: rework quirks settings")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The .set_fmt() callback of the axg tdm interface incorrectly
test the content of SND_SOC_DAIFMT_MASTER_MASK as if it was a
bitfield, which it is not.
Implement the test correctly.
Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Global EN register guide to off before AMP_EN register
when amp disable sequence.
- remove AMP_EN control before max98390_dac_event call
Signed-off-by: Steve Lee <steves.lee@maximintegrated.com>
Link: https://lore.kernel.org/r/20200724060058.19201-1-steves.lee@maximintegrated.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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kabylake_ssp_fixup function uses snd_soc_dpcm to identify the
codecs DAIs. The HW parameters are changed based on the codec DAI of the
stream. The earlier approach to get snd_soc_dpcm was using container_of()
macro on snd_pcm_hw_params.
The structures have been modified over time and snd_soc_dpcm does not have
snd_pcm_hw_params as a reference but as a copy. This causes the current
driver to crash when used.
This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime
holds 2 dpcm instances (one for playback and one for capture). 2 codecs
on the SSP are dmic (capture) and speakers (playback). Based on the
stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime.
Tested for all use cases of the driver.
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
Tested-by: Lukasz Majczak <lma@semihalf.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/1595432147-11166-1-git-send-email-harshapriya.n@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Starting in commit cbc7a6b5a87a ("ASoC: soc-card: add
snd_soc_card_add_dai_link()"), error value from ASoc add_dai_link() is
no longer ignored.
The generic HDA machine driver relied on the old semantics to disable
i915 HDMI/DP audio codec at runtime. If no display codec was present,
add_dai_link() returned an error, but this was ignored and rest of the
card was successfully probed.
Fix the problem by changing the machine driver add_dai_link() to not
return an error in this case.
Fixes: cbc7a6b5a87a ("ASoC: soc-card: add snd_soc_card_add_dai_link()")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2261
Link: https://lore.kernel.org/r/20200714132804.3638221-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Partially reverts commit 128f825aeab7 ("ASoC: max98357a: move control
of SD_MODE to DAPM").
In order to have mute control of max98357 from machine drivers, commit
128f825aeab7 ("ASoC: max98357a: move control of SD_MODE to DAPM")
moves the control of SD_MODE from DAI ops to DAPM events. However, pop
noise has been observed on rk3399-gru-kevin boards due to this commit.
The commit 128f825aeab7 caused sequence of DAI clocks and SD_MODE
changed on rk3399-gru-kevin boards.
With the commit 128f825aeab7:
- SD_MODE will be set to 1 before DAI clocks start.
- SD_MODE will be set to 0 after DAI clocks stop.
As a result, pop noise.
Moves the control of SD_MODE back to DAI ops. In the meantime, uses an
additional flag in DAPM event to provide chance of mute control for
machine drivers.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Tested-By: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Link: https://lore.kernel.org/r/20200721114232.2812254-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into master
Pull sound fixes from Takashi Iwai:
"This became fairly large, containing mostly the collection of ASoC
fixes that slipped from the previous request, so I sent now a bit
earlier than usual. But all changes look small and mostly
device-specific, hence nothing to worry too much.
Majority of changes are for x86 based platforms and their CODEC
drivers, in order to address some issues hit by their recent tests and
fuzzing. The rest are other ASoC device-specific fixes (imx, qcom,
wm8974, amd, rockchip) as well as a trivial fix for a kernel WARNING
hit by syzkaller"
* tag 'sound-5.8-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (28 commits)
ALSA: hda/realtek: Fixed ALC298 sound bug by adding quirk for Samsung Notebook Pen S
ALSA: info: Drop WARN_ON() from buffer NULL sanity check
ASoC: rt5682: Report the button event in the headset type only
ASoC: Intel: bytcht_es8316: Add missed put_device()
ASoC: rt5682: Enable Vref2 under using PLL2
ASoC: rt286: fix unexpected interrupt happens
ASoC: wm8974: remove unsupported clock mode
ASoC: wm8974: fix Boost Mixer Aux Switch
ASoC: SOF: core: fix null-ptr-deref bug during device removal
ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend
ASoC: codecs: max98373: Removed superfluous volume control from chip default
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: fix kernel oops on route addition error
ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M
ASoC: Intel: bdw-rt5677: fix non BE conversion
ASoC: soc-dai: set dai_link dpcm_ flags with a helper
MAINTAINERS: Add Shengjiu to reviewer list of sound/soc/fsl
ASoC: core: Remove only the registered component in devm functions
MAINTAINERS: Change Maintainer for some at91 drivers
ASoC: dt-bindings: simple-card: Fix 'make dt_binding_check' warnings
...
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This configuration is for EHL with the RT5660 codec. RT5660
should use "10EC5660" ID instead of "INTC1027".
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Notebook Pen S
Fixed no headphone sound bug on laptop Samsung Notebook Pen S
(950SBE-951SBE), by using existing patch in Linus' tree, commit
14425f1f521f (ALSA: hda/realtek: Add quirk for Samsung Notebook).
This laptop uses the same ALC298 but different subsystem id 0x144dc812.
I added SND_PCI_QUIRK at sound/pci/hda/patch_realtek.c
Signed-off-by: Joonho Wohn <doomsheart@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAHcbMh291aWDKiWSZoxXB4-Eru6OYRwGA4AVEdCZeYmVLo5ZxQ@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.8
An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them. For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into master
Pull sound fixes from Takashi Iwai:
"No surprise here, just a few device-specific small fixes: two fixes
for USB LINE6 and one for USB-audio drivers wrt syzkaller fuzzer
issues, while the rest are all HD-audio Realtek quirks"
* tag 'sound-5.8-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - fixup for yet another Intel reference board
ALSA: hda/realtek - Enable Speaker for ASUS UX563
ALSA: hda/realtek - Enable Speaker for ASUS UX533 and UX534
ALSA: hda/realtek: Enable headset mic of Acer TravelMate B311R-31 with ALC256
ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289
ALSA: hda/realtek - change to suitable link model for ASUS platform
ALSA: usb-audio: Fix race against the error recovery URB submission
ALSA: line6: Sync the pending work cancel at disconnection
ALSA: line6: Perform sanity check for each URB creation
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axg_card_add_tdm_loopback() misses to call kfree() in an error path. We
can use devm_kasprintf() to fix the issue, also improve maintainability.
So use it instead.
Fixes: c84836d7f650 ("ASoC: meson: axg-card: use modern dai_link style")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200717082242.130627-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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snd_info_get_line() has a sanity check of NULL buffer -- both buffer
itself being NULL and buffer->buffer being NULL. Basically both
checks are valid and necessary, but the problem is that it's with
snd_BUG_ON() macro that triggers WARN_ON(). The latter condition
(NULL buffer->buffer) can be met arbitrarily by user since the buffer
is allocated at the first write, so it means that user can trigger
WARN_ON() at will.
This patch addresses it by simply moving buffer->buffer NULL check out
of snd_BUG_ON() so that spurious WARNING is no longer triggered.
Reported-by: syzbot+e42d0746c3c3699b6061@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200717084023.5928-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The irq work will be manipulated by resume function, and it will report
the wrong jack type while the jack type is headphone in the button event.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200716030123.27122-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error
path. Add the missed function call to fix it.
Fixes: ba49cf6f8e4a ("ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200714080918.148196-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add headset_jack for the intel reference board support with
10ec:1230.
Signed-off-by: PeiSen Hou <pshou@realtek.com.tw>
Link: https://lore.kernel.org/r/20200716090134.9811-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASUS UX563 speaker can't output.
Add quirk to link suitable model will enable it.
This model also could enable headset Mic.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/96dee3ab01a04c28a7b44061e88009dd@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASUS UX533 and UX534 speaker still can't output.
End User feedback speaker didn't have output.
Add this COEF value will enable it.
Fixes: 4e051106730d ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/80334402a93b48e385f8f4841b59ae09@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Enable Vref2 under long term using PLL2 to avoid clock unstable.
Signed-off-by: derek.fang <derek.fang@realtek.com>
Link: https://lore.kernel.org/r/1594721600-29994-1-git-send-email-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The Acer TravelMate B311R-31 laptop's audio (1025:1430) with ALC256
cannot detect the headset microphone until
ALC256_FIXUP_ACER_MIC_NO_PRESENCE quirk maps the NID 0x19 as the headset
mic pin.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200713060421.62435-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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with ALC289
This patch adds support for headset mic to the ASUS ROG Zephyrus
G14(GA401) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200711110557.18681-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASUS platform couldn't need to use Headset Mode model.
It changes to the suitable model.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/d05bcff170784ec7bb35023407148161@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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USB MIDI driver has an error recovery mechanism to resubmit the URB in
the delayed timer handler, and this may race with the standard start /
stop operations. Although both start and stop operations themselves
don't race with each other due to the umidi->mutex protection, but
this isn't applied to the timer handler.
For fixing this potential race, the following changes are applied:
- Since the timer handler can't use the mutex, we apply the
umidi->disc_lock protection at each input stream URB submission;
this also needs to change the GFP flag to GFP_ATOMIC
- Add a check of the URB refcount and skip if already submitted
- Move the timer cancel call at disconnection to the beginning of the
procedure; this assures the in-flight timer handler is gone properly
before killing all pending URBs
Reported-by: syzbot+0f4ecfe6a2c322c81728@syzkaller.appspotmail.com
Reported-by: syzbot+5f1d24c49c1d2c427497@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200710160656.16819-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Recently syzkaller reported a UAF in LINE6 driver, and it's likely
because we call cancel_delayed_work() at the disconnect callback
instead of cancel_delayed_work_sync(). Let's use the correct one
instead.
Reported-by: syzbot+145012a46658ac00fc9e@syzkaller.appspotmail.com
Suggested-by: Alan Stern <stern@rowland.harvard.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hlfjr4gio.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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LINE6 drivers create stream URBs with a fixed pipe without checking
its validity, and this may lead to a kernel WARNING at the submission
when a malformed USB descriptor is passed.
For avoiding the kernel warning, perform the similar sanity checks for
each pipe type at creating a URB.
Reported-by: syzbot+c190f6858a04ea7fbc52@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hv9iv4hq8.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HV/VREF should not turn off if the headphone jack plug-in.
This patch could solve the unexpected interrupt issue in some devices.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200709101345.11449-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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In DSP_A mode, BIT7 of IFACE should bit 0 according to datasheet (ie.
inverted frame clock is not support in this mode).
Signed-off-by: Puyou Lu <puyou.lu@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1593657056-4989-1-git-send-email-puyou.lu@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Clear BIT6 of INPPGA means not muted (Switch On).
Signed-off-by: Puyou Lu <puyou.lu@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1593657025-4903-1-git-send-email-puyou.lu@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small, mostly device-specific fixes.
The significant one is the regression fix for USB-audio implicit
feedback devices due to the incorrect frame size calculation, which
landed in 5.8 and stable trees.
In addition, a few usual HD-audio and USB-audio quirks, Intel HDMI
fixes, ASoC fsl and rt5682 fixes, as well as the fix in
compress-offload partial drain operation"
* tag 'sound-5.8-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: compress: fix partial_drain completion state
ALSA: usb-audio: Add implicit feedback quirk for RTX6001
ALSA: usb-audio: add quirk for MacroSilicon MS2109
ALSA: hda/realtek: Enable headset mic of Acer Veriton N4660G with ALC269VC
ALSA: hda/realtek: Enable headset mic of Acer C20-820 with ALC269VC
ALSA: hda/realtek - Enable audio jacks of Acer vCopperbox with ALC269VC
ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id
ALSA: hda/hdmi: improve debug traces for stream lookups
ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
ALSA: opl3: fix infoleak in opl3
ALSA: usb-audio: Replace s/frame/packet/ where appropriate
ALSA: usb-audio: Fix packet size calculation
AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module
ALSA: hda - let hs_mic be picked ahead of hp_mic
ASoC: rt5682: fix the pop noise while OMTP type headset plugin
ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable
ASoC: fsl_mqs: Don't check clock is NULL before calling clk API
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<pierre-louis.bossart@linux.intel.com>:
This is hopefully the last set of fixes to avoid probe errors due to
stricter checks of DAI capabilities introduced late in the 5.8 cycle.
Daniel Baluta (1):
ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M
Pierre-Louis Bossart (2):
ASoC: soc-dai: set dai_link dpcm_ flags with a helper
ASoC: Intel: bdw-rt5677: fix non BE conversion
include/sound/soc-dai.h | 1 +
sound/soc/generic/audio-graph-card.c | 4 +--
sound/soc/generic/simple-card.c | 4 +--
sound/soc/intel/boards/bdw-rt5677.c | 1 +
sound/soc/soc-dai.c | 38 ++++++++++++++++++++++++++++
sound/soc/sof/imx/imx8.c | 8 ++++++
sound/soc/sof/imx/imx8m.c | 8 ++++++
7 files changed, 60 insertions(+), 4 deletions(-)
base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759
--
2.25.1
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Bossart <pierre-louis.bossart@linux.intel.com>:
While experimenting and introducing errors in Baytrail topology files
until I got them right, I encountered multiple kernel oopses and
memory leaks. This is a first batch to harden the code, but we should
probably think of a tool to fuzz the topology...
Pierre-Louis Bossart (5):
ASoC: topology: fix kernel oops on route addition error
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: use break on errors, not continue
ASoC: topology: factor kfree(se) in error handling
ASoC: topology: add more logs when topology load fails.
sound/soc/soc-topology.c | 97 ++++++++++++++++++++++++----------------
1 file changed, 58 insertions(+), 39 deletions(-)
base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759
--
2.25.1
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Bossart <pierre-louis.bossart@linux.intel.com>:
V2 with a number of cleanups:
split between I2C and SoundWire modes, as done for rt5682, and updated
Kconfigs.
removed useless initializations common to both modes
removed idle_bias on
fixed register classified as volatile in error
fixed SPDX comments
Pierre-Louis Bossart (2):
ASoC: codecs: max98373: split I2C and common parts
ASoC: Intel: sof-sdw: add MAX98373 I2C dependencies
Ryan Lee (2):
ASoC: codecs: max98373: Removed superfluous volume control from chip
default
ASoC: codecs: max98373: add SoundWire support
randerwang (2):
ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend
ASoC: Intel: sdw_max98373: add card_late_probe support
sound/soc/codecs/Kconfig | 20 +-
sound/soc/codecs/Makefile | 4 +
sound/soc/codecs/max98373-i2c.c | 612 +++++++++++++++
sound/soc/codecs/max98373-sdw.c | 887 ++++++++++++++++++++++
sound/soc/codecs/max98373-sdw.h | 72 ++
sound/soc/codecs/max98373.c | 619 +--------------
sound/soc/codecs/max98373.h | 17 +-
sound/soc/intel/boards/Kconfig | 7 +-
sound/soc/intel/boards/sof_sdw.c | 19 +-
sound/soc/intel/boards/sof_sdw_common.h | 6 +
sound/soc/intel/boards/sof_sdw_max98373.c | 12 +
11 files changed, 1668 insertions(+), 607 deletions(-)
create mode 100644 sound/soc/codecs/max98373-i2c.c
create mode 100644 sound/soc/codecs/max98373-sdw.c
create mode 100644 sound/soc/codecs/max98373-sdw.h
base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759
--
2.25.1
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The DSP should be notified for device removal only if the
probe was successful. Fixes the following KASAN bug:
BUG: KASAN: null-ptr-deref in sof_ipc_tx_message+0x80/0x160 [snd_sof]
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707204027.114169-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Idle_bias_on is used to decide bias on/off in standby state by dapm.
When Idle_bias_on is set to one, dapm will keep max98373 active at
idle time. Max98373 is doing nothing in this state, so remove
idle_bias_on setting to let max98373 get suspended when it is idle.
Signed-off-by: randerwang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ryan Lee <ryans.lee@maximintegrated.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707205740.114927-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Volume control in probe function is not necessary.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707205740.114927-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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we need to free all allocated tlvs, not just the one allocated in
the loop before releasing kcontrols - other the tlvs references will
leak.
Fixes: 9f90af3a995298 ('ASoC: topology: Consolidate and fix asoc_tplg_dapm_widget_*_create flow')
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/20200707203749.113883-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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When errors happens while loading graph components, the kernel oopses
while trying to remove all topology components. This can be
root-caused to a list pointing to memory that was already freed on
error.
remove_route() is already called on errors and will perform the
required cleanups so there's no need to free the route memory in
soc_tplg_dapm_graph_elems_load() if the route was added to the
list. We do however want to free the routes allocated but not added to
the list.
Fixes: 7df04ea7a31ea ('ASoC: topology: modify dapm route loading routine and add dapm route unloading')
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/20200707203749.113883-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This is identical with change for Intel platforms done with
commit 8c05246c0b58 ("ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell")
and fixes a regression on i.MX8/i.MX8M:
[ 25.705750] esai-Codec: ASoC: no backend playback stream
[ 27.923378] esai-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.
Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200707210439.115300-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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When SOF is used, the normal links are converted into DPCM ones. This
generates an error
[ 58.276668] bdw-rt5677 bdw-rt5677: CPU DAI spi-RT5677AA:00 for rtd
Wake on Voice does not support playback
[ 58.276676] bdw-rt5677 bdw-rt5677: ASoC: can't create pcm Wake on
Voice :-22
Fix by forcing the capture direction.
Fixes: b73287f0b0745 ('ASoC: soc-pcm: dpcm: fix playback/capture checks')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Curtis Malainey <curtis@malainey.com>
Link: https://lore.kernel.org/r/20200707210439.115300-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add a helper to walk through all the DAIs and set dpcm_playback and
dpcm_capture flags based on the DAIs capabilities, and use this helper
to avoid setting these flags arbitrarily in generic cards.
The commit referenced in the Fixes tag did not introduce the
configuration issue but will prevent the card from probing when
detecting invalid configurations.
Fixes: b73287f0b0745 ('ASoC: soc-pcm: dpcm: fix playback/capture checks')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200707210439.115300-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The ASoC devm_ functions that register a component
(devm_snd_soc_register_component and devm_snd_dmaengine_pcm_register) will
clean their component by running snd_soc_unregister_component.
snd_soc_unregister_component will then remove all the components for the
device that was used to register the component in the first place.
However, some drivers register several components (such as a DAI and a
dmaengine PCM) on the same device, and if the dmaengine PCM is registered
first, then the DAI will be cleaned up first and
snd_dmaengine_pcm_unregister will be called next.
snd_dmaengine_pcm_unregister will then lookup the dmaengine PCM component
on the device, and if there's one unregister that component and release its
dmaengine channels. That doesn't happen in practice though since the first
call to snd_soc_unregister_component removed all the components, so we
never get the chance to release the dmaengine channels.
In order to fix this, instead of removing all the components for a given
device, we can simply remove the component that was registered in the first
place. We should have the same number of component registration than we
have components, so it should work just fine.
Signed-off-by: Maxime Ripard <maxime@cerno.tech>
Link: https://lore.kernel.org/r/20200707074237.287171-1-maxime@cerno.tech
Signed-off-by: Mark Brown <broonie@kernel.org>
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On partial_drain completion we should be in SNDRV_PCM_STATE_RUNNING
state, so set that for partially draining streams in
snd_compr_drain_notify() and use a flag for partially draining streams
While at it, add locks for stream state change in
snd_compr_drain_notify() as well.
Fixes: f44f2a5417b2 ("ALSA: compress: fix drain calls blocking other compress functions (v6)")
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Tested-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200629134737.105993-4-vkoul@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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USB Audio analyzer RTX6001 uses the same implicit feedback quirk
as other XMOS-based devices.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/822f0f20-1886-6884-a6b2-d11c685cbafa@ivitera.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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