Replace direct modifications to vma->vm_flags with calls to modifier
functions to be able to track flag changes and to keep vma locking
correctness.
[akpm@linux-foundation.org: fix drivers/misc/open-dice.c, per Hyeonggon Yoo]
Link: https://lkml.kernel.org/r/20230126193752.297968-5-surenb@google.com
Signed-off-by: Suren Baghdasaryan <surenb@google.com>
Acked-by: Michal Hocko <mhocko@suse.com>
Acked-by: Mel Gorman <mgorman@techsingularity.net>
Acked-by: Mike Rapoport (IBM) <rppt@kernel.org>
Acked-by: Sebastian Reichel <sebastian.reichel@collabora.com>
Reviewed-by: Liam R. Howlett <Liam.Howlett@Oracle.com>
Reviewed-by: Hyeonggon Yoo <42.hyeyoo@gmail.com>
Cc: Andy Lutomirski <luto@kernel.org>
Cc: Arjun Roy <arjunroy@google.com>
Cc: Axel Rasmussen <axelrasmussen@google.com>
Cc: David Hildenbrand <david@redhat.com>
Cc: David Howells <dhowells@redhat.com>
Cc: Davidlohr Bueso <dave@stgolabs.net>
Cc: David Rientjes <rientjes@google.com>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Greg Thelen <gthelen@google.com>
Cc: Hugh Dickins <hughd@google.com>
Cc: Ingo Molnar <mingo@redhat.com>
Cc: Jann Horn <jannh@google.com>
Cc: Joel Fernandes <joelaf@google.com>
Cc: Johannes Weiner <hannes@cmpxchg.org>
Cc: Kent Overstreet <kent.overstreet@linux.dev>
Cc: Laurent Dufour <ldufour@linux.ibm.com>
Cc: Lorenzo Stoakes <lstoakes@gmail.com>
Cc: Matthew Wilcox <willy@infradead.org>
Cc: Minchan Kim <minchan@google.com>
Cc: Paul E. McKenney <paulmck@kernel.org>
Cc: Peter Oskolkov <posk@google.com>
Cc: Peter Xu <peterx@redhat.com>
Cc: Peter Zijlstra <peterz@infradead.org>
Cc: Punit Agrawal <punit.agrawal@bytedance.com>
Cc: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Cc: Shakeel Butt <shakeelb@google.com>
Cc: Soheil Hassas Yeganeh <soheil@google.com>
Cc: Song Liu <songliubraving@fb.com>
Cc: Vlastimil Babka <vbabka@suse.cz>
Cc: Will Deacon <will@kernel.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Merge series from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
From q6dsp side issues are around locking of position pointer and handle
From LPASS codec side most of the staiblity issues were around runtime pm,:
While testing X13s audio, we found multiple stablity issues this patchset
fixes these issues.
From q6dsp side issues are around locking of position pointer and handle
multiple prepare cases along with pulse audio timerbased scheduling workaround.
From LPASS codec side most of the staiblity issues were around runtime pm,
hitting various issues as the codec was firstly resetting the soundwire block
for every clk disable/enable which is taking the slaves out of sync and
resulting in re-enumerating. Second issue was around fsgen clk is not
brining up the codec out of suspend as it was not added after
runtime pm enabled. Final issue was with codec mclk rate which should
have been 192KHz same as npl instead of 96KHz. We were getting lucky as
wsa drivers are setting the same clk to 192KHz.
With this patches, x13s audio is pretty stable.
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
In a course of creating complicated topologies where multiple output pins of a
copier is enabled, we have discovered that additional configuration needs to be
sent to the firmware to make the use cases working.
Merge series from Herve Codina <herve.codina@bootlin.com>:
The Infineon PEB2466 codec is a programmable DSP-based four channels
codec with filters capabilities.
It also provides signals as GPIOs.
This is the initial codec driver for rt712 SDCA (Jack+Amp topology).
The host should connect with rt712 SdW1 interface.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20230207090946.20659-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Simple-card/audio-graph-card drivers do not handle MCLK clock when it
is specified in the codec device node. The expectation here is that,
the codec should actually own up the MCLK clock and do necessary setup
in the driver.
This is inspired from,
commit dbf54a9534 ("ASoC: rt5659: Update MCLK rate in set_sysclk()").
Cc: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1675953417-8686-2-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Infineon PEB2466 codec is a programmable DSP-based four channels
codec with filters capabilities.
It also provides signals as GPIOs.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://lore.kernel.org/r/20230206144904.91078-3-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
MOD_INIT_INSTANCE IPC for a copier only contains the sink format for
output pin 0. Any additional output pins that are used need to have their
sink format set using the LARGE_CONFIG_SET IPC message.
Otherwise, firmware will report error or crash due to NULL format is used.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230209142123.17193-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Print the queue ID's during bind/unbind errors as well to make it easier
to see what failed exactly.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230209142123.17193-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
resetting soundwire block will put the slaves out of sync and result
in re-enumeration during fsgen disable/enable path this is totally
unnecessary and resulting fifo overflows.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230209122806.18923-8-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason we ended up with incorrect mclk rate which should be
1920000 instead of 96000, So far we were getting lucky as the same clk
is set to 192000 by wsa and va macro. This issue is discovered when there
is no wsa macro active and only rx or tx path is tested.
Fix this by setting correct rate.
Fixes: c39667ddcf ("ASoC: codecs: lpass-tx-macro: add support for lpass tx macro")
Fixes: af3d54b997 ("ASoC: codecs: lpass-rx-macro: add support for lpass rx macro")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230209122806.18923-7-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
currently q6apm_is_adsp_ready() will only return the cached value of
previous result. If we are unlucky and previous result is not-ready
then the caller will always get not-ready flag.
This is not correct, we should query the dsp of its current state in
irrespective of previous reported state.
Fixes: 47bc8cf60e ("ASoC: qdsp6: audioreach: Add ADSP ready check")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230209122806.18923-5-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment, playing audio with PulseAudio with the qdsp6 driver
results in distorted sound. It seems like its timer-based scheduling
does not work properly with qdsp6 since setting tsched=0 in
the PulseAudio configuration avoids the issue.
Apparently this happens when the pointer() callback is not accurate
enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop
PulseAudio from using timer-based scheduling by default.
According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html:
The flag is being used in the sense explained in the previous audio
meeting -- the data transfer granularity isn't fine enough but aligned
to the period size (or less).
q6apm-dai reports the position as multiple of
prtd->pcm_count = snd_pcm_lib_period_bytes(substream)
so it indeed just a multiple of the period size.
Therefore adding the flag here seems appropriate and makes audio
work out of the box.
Comment log inspired by Stephan Gerhold sent for q6asm-dai.c few years back.
Fixes: 9b4fe0f1cd ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230209122806.18923-4-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
It is noticed that the position pointer value seems to get a get corrupted
due to missing locking between updating and reading.
Fix this by adding a spinlock around the position pointer.
Fixes: 9b4fe0f1cd ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230209122806.18923-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
prepare callback can be called multiple times, so unprepare the stream
if its already prepared.
Without this DSP is not happy to setting the params on a already
prepared graph.
Fixes: 9b4fe0f1cd ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230209122806.18923-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
All callers from other files ignore the return value of this function.
And it can only ever return a non-zero value if the parameter card is NULL.
This cannot happen in snd_card_free() as card was dereferenced just before
snd_card_free_when_closed() is called. So the error handling can be dropped
there.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Geoff Levand <geoff@infradead.org>
Acked-by: Thierry Reding <treding@nvidia.com>
Link: https://lore.kernel.org/r/20230207191907.467756-3-u.kleine-koenig@pengutronix.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All callers from other files ignore the return value of this function.
And it can only ever return a non-zero value if the parameter card is NULL.
Move the check for card being NULL into snd_card_free_when_closed() to keep
the previous behaviour. Note this isn't necessary for
snd_card_disconnect_sync() because if card was NULL in there the dereference
of card for dev_err() would oops the kernel. Replace this by an oops
triggered by the dereference of card for spin_lock_irq().
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Geoff Levand <geoff@infradead.org>
Acked-by: Thierry Reding <treding@nvidia.com>
Link: https://lore.kernel.org/r/20230207191907.467756-2-u.kleine-koenig@pengutronix.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SOF driver calls snd_sof_dsp_update8 with parameters mask and value but
the snd_sof_dsp_update8 declares these two parameters in reverse order.
This causes some issues such as d0i3 register can't be set correctly
Now change function definition according to common SOF usage.
Fixes: c28a36b012 ("ASoC: SOF: ops: add snd_sof_dsp_updateb() helper")
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230208104404.20554-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A few more fixes for v6.2, all driver specific and small. It's larger
than is ideal but we can't really control when people find problems.
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Merge tag 'asoc-fix-v6.2-rc7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.2
A few more fixes for v6.2, all driver specific and small. It's larger
than is ideal but we can't really control when people find problems.
When handling error path, ret needs to be set to correct value.
Reported-by: kernel test robot <lkp@intel.com>
Reported-by: Dan Carpenter <error27@gmail.com>
Fixes: d29d41e28e ("ASoC: topology: Add support for multiple kcontrol types to a widget")
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230207210428.2076354-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_emux_xg_control() can be called with an argument 'param' greater
than size of 'control' array. It may lead to accessing 'control'
array at a wrong index.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Signed-off-by: Artemii Karasev <karasev@ispras.ru>
Fixes: 1da177e4c3 ("Linux-2.6.12-rc2")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230207132026.2870-1-karasev@ispras.ru
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The version information is at the bit31 ~ bit16 in the VERID
register, so need to right shift 16bit to get it, otherwise
the result of comparison "sai->verid.version >= 0x0301" is
wrong.
Fixes: 99c1e74f25 ("ASoC: fsl_sai: store full version instead of major/minor")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Iuliana Prodan <iuliana.prodan@nxp.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/1675760664-25193-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a HP platform needs ALC236_FIXUP_HP_GPIO_LED quirk to
make mic-mute/audio-mute working.
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230207083011.100189-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Daniel Beer <daniel.beer@igorinstitute.com>:
This pair of patches fixes two issues which crept in while revising the
original submission, at a time when I no longer had access to test
hardware.
The fixes here have been tested and verified on hardware.
This Asus Zenbook laptop use Realtek HDA codec combined with
2xCS35L41 Amplifiers using I2C with External Boost.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230206150019.3825120-1-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The val is defined as unsigned int type, if(val<0) is redundant, so
delete it.
sound/soc/codecs/idt821034.c:449 idt821034_kctrl_gain_put() warn: unsigned 'val' is never less than zero.
Reported-by: Abaci Robot <abaci@linux.alibaba.com>
Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=3947
Signed-off-by: Jiapeng Chong <jiapeng.chong@linux.alibaba.com>
Acked-by: Herve Codina <herve.codina@bootlin.com>
Link: https://lore.kernel.org/r/20230206075518.84169-1-jiapeng.chong@linux.alibaba.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck reports
sound/soc/codecs/aw88395/aw88395_lib.c:789:6: error: Uninitialized variable: cur_scene_id [uninitvar]
if (cur_scene_id == 0) {
^
Passing a garbage value to aw_dev_parse_data_by_sec_type_v1() will cause a crash
when the value is used as an array index. This check assumes cur_scene_id is
initialized to 0, so initialize it to 0.
Fixes: 4345865b00 ("ASoC: codecs: ACF bin parsing and check library file for aw88395")
Signed-off-by: Tom Rix <trix@redhat.com>
Link: https://lore.kernel.org/r/20230205015733.1721009-1-trix@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Seems like properties parsing and reading was copy-pasted,
so "everest,interrupt-src" and "everest,interrupt-clk" are saved into
the es8326->jack_pol variable. This might lead to wrong settings
being saved into the reg 57 (ES8326_HP_DET).
Fix this by using proper variables while reading properties.
Signed-off-by: Alexey Firago <a.firago@yadro.com>
Reviewed-by: Yang Yingliang <yangyingliang@huawei.com
Link: https://lore.kernel.org/r/20230204195106.46539-1-a.firago@yadro.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In tas5805m_refresh, we switch pages to update the DSP volume control,
but we need to switch back to page 0 before trying to alter the
soft-mute control. This latter page-switch was missing.
Fixes: ec45268467 ("ASoC: add support for TAS5805M digital amplifier")
Signed-off-by: Daniel Beer <daniel.beer@igorinstitute.com>
Link: https://lore.kernel.org/r/1fea38a71ea6ab0225d19ab28d1fa12828d762d0.1675497326.git.daniel.beer@igorinstitute.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There's some setup we need to do in order to get the DSP initialized,
and this can't be done until a bit-clock is ready. In an earlier version
of this driver, this work was done in a DAPM callback.
The DAPM callback doesn't guarantee that the bit-clock is running, so
the work was moved instead to the trigger callback. Unfortunately this
callback runs in atomic context, and the setup code needs to do I2C
transactions.
Here we use a work_struct to kick off the setup in a thread instead.
Fixes: ec45268467 ("ASoC: add support for TAS5805M digital amplifier")
Signed-off-by: Daniel Beer <daniel.beer@igorinstitute.com>
Link: https://lore.kernel.org/r/85d8ba405cb009a7a3249b556dc8f3bdb1754fdf.1675497326.git.daniel.beer@igorinstitute.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The HP Elitebook 645 G9 laptop (with motherboard model 89D2) uses the
ALC236 codec and requires the alc236_fixup_hp_mute_led_micmute_vref
fixup in order to enable mute/micmute LEDs.
Note: the alc236_fixup_hp_gpio_led fixup, which is used by the Elitebook
640 G9, does not work with the 645 G9.
[ rearranged the entry in SSID order -- tiwai ]
Signed-off-by: Elvis Angelaccio <elvis.angelaccio@kde.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/4055cb48-e228-8a13-524d-afbb7aaafebe@kde.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current structure includes no field to express the number of messages
copied to user space, thus user space application needs to information
out of the structure to parse the content of structure.
This commit adds a field to express the number of messages copied to user
space since It is more preferable to use self-contained structure.
Kees Cook proposed an idea of annotation for bound of flexible arrays
in his future improvement for flexible-length array in kernel. The
additional field for message count is suitable to the idea as well.
Reference: https://people.kernel.org/kees/bounded-flexible-arrays-in-c
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20230202133708.163936-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some motherboards have multiple HDA codecs connected to the serial bus.
The current code may create multiple mixer controls with the almost
identical identification.
The current code use id.device field from the control element structure
to store the codec address to avoid such clashes for multiple codecs.
Unfortunately, the user space do not handle this correctly. For mixer
controls, only name and index are used for the identifiers.
This patch fixes this problem to compose the index using the codec
address as an offset in case, when the control already exists. It is
really unlikely that one codec will create 10 similar controls.
This patch adds new kernel module parameter 'ctl_dev_id' to allow
select the old behaviour, too. The CONFIG_SND_HDA_CTL_DEV_ID Kconfig
option sets the default value.
BugLink: https://github.com/alsa-project/alsa-lib/issues/294
BugLink: https://github.com/alsa-project/alsa-lib/issues/205
Fixes: 54d1740315 ("[ALSA] hda-codec - Fix connection list parsing")
Fixes: 1afe206ab6 ("ALSA: hda - Try to find an empty control index when it's occupied")
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230202092013.4066998-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
The following series adds support for the PCM delay reporting in SOF core level
and implements the needed infrastructure with IPC4 to finally enable it for MTL.
Currently this is only supported on MTL (and via IPC4), but with the
infrastructure in place it will be possible to support other platforms with
DeepBuffer.
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
Today I came across two regressions in next with SOF:
The topology would not load with a failure of creating playback DAI
the first patch is fixing this which was caused by a missing 'else' in the patch
After fixing the topology loading, the module unloading caused kernel panic.
The second patch is correcting that which is I likely caused by copy-paste to
set wrong unload callback for the graph element.
With these patches applied SOF is working on next and modules can be unloaded
My randconfig build setup ran into a rare build failure with
CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE=y
CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE=m
CONFIG_SOUNDWIRE=y
CONFIG_SOUNDWIRE_INTEL=m
CONFIG_SND_SOC_SOF_HDA=y
CONFIG_SND_SOC_SOF_INTEL_TGL=y
x86_64-linux-ld: sound/soc/sof/intel/hda.o: in function `hda_init_caps':
hda.c:(.text+0x691): undefined reference to `sdw_intel_cnl_hw_ops'
x86_64-linux-ld: hda.c:(.text+0x6f2): undefined reference to `sdw_intel_probe'
x86_64-linux-ld: sound/soc/sof/intel/hda.o: in function `hda_sdw_startup':
hda.c:(.text+0x1c40): undefined reference to `sdw_intel_startup'
x86_64-linux-ld: sound/soc/sof/intel/hda.o: in function `hda_sdw_process_wakeen':
hda.c:(.text+0x1cb6): undefined reference to `sdw_intel_process_wakeen_event'
x86_64-linux-ld: sound/soc/sof/intel/hda.o: in function `hda_dsp_interrupt_thread':
hda.c:(.text+0x1d67): undefined reference to `sdw_intel_thread'
x86_64-linux-ld: sound/soc/sof/intel/hda.o: in function `hda_dsp_remove':
hda.c:(.text+0x2655): undefined reference to `sdw_intel_exit'
My best understanding is that the definition of
SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE was intended to avoid this
problem, but got it wrong for the SND_SOC_SOF_INTEL_SOUNDWIRE=m case,
where the 'select' is meant to set SOUNDWIRE_INTEL to the value of
SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE rather than the intersection of
SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE and SND_SOC_SOF_INTEL_SOUNDWIRE.
Change the condition to check for SND_SOC_SOF_INTEL_SOUNDWIRE to be a
boolean rather than a tristate expression in order to propagate this
as intended.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20230202102247.806749-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Some Smatch static checker warning like below was found.
sound/soc/mediatek/mt8188/mt8188-dai-etdm.c:2487
mt8188_dai_etdm_parse_of()
warn: 'ret' returned from snprintf() might be larger than 48
2479 for (i = 0; i < MT8188_AFE_IO_ETDM_NUM; i++) {
2480 dai_id = ETDM_TO_DAI_ID(i);
2481 etdm_data = afe_priv->dai_priv[dai_id];
2482
2483 ret = snprintf(prop, sizeof(prop),
2484 "mediatek,%s-multi-pin-mode",
2485 of_afe_etdms[i].name);
2486 if (ret < 0) {
--> 2487 dev_err(afe->dev, "%s snprintf
err=%d\n",
2488
In linux kernel, snprintf() never returns negatives. On the other hand,
the format string like "mediatek,%s-multi-pin-mode" must be smaller
than sizeof(prop)=48.
After discussing in the mail thread[1], I remove the dead code to fix
the Smatch warnings.
[1]: https://lore.kernel.org/all/Y9EdBg641tJDDrt%2F@kili/
Signed-off-by: Trevor Wu <trevor.wu@mediatek.com>
Link: https://lore.kernel.org/r/20230202103704.15626-1-trevor.wu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The delay function is used to calculate the difference
between hw_ptr and dai dma position. I2S, DMIC and SDW will
use dai dma position in shared SRAM window to calculate the
delay. HDaudio will retrieve dai dma position from host mmio memory
space since it doesn't support LLP counter reported by firmware.
In two cases dai dma position is inaccurate for delay calculation
(1) dai pipeline is started before host pipeline
(2) multiple streams mixed into one. Each stream has the same dai
dma position
Firmware calculates correct stream_start_offset for all cases including
above two. Driver subtracts stream_start_offset from dai dma position to
get accurate one.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230202132954.26773-10-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Get HDaudio link position for current stream delay calculation
from hda registers.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230202132954.26773-9-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
PCM delay depends on stream position based on hardware
counter to calculate stream delay so add this ops to get
stream position according to hardware counter.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230202132954.26773-8-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Sof framework will call specific delay function for
different IPC version.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230202132954.26773-7-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the hw_params to init time info for ipc4 delay calculation.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230202132954.26773-6-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Allocate time info when pcm is loaded by topology
and free it when pcm is unloaded by topology.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230202132954.26773-5-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Start_stream_offset is used to strip invalid sample count in dai
for some cases like dai is started before host. llp_offset is used
to get current dai position from memory windows.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230202132954.26773-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
FW can share some information with host driver, .e.g fw status, pipeline
status and volume status. On ipc4 platform it is located in memory
windows 0 with size of struct sof_ipc4_fw_reg.
With this patch, ipc4 driver can find fw information in fw_info_box
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230202132954.26773-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Ipc4_fw_reg defines the content of memory window 0 shared by fw.
Host driver can get fw information by data structure defined in
this file.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230202132954.26773-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Clang warns:
../sound/soc/atmel/mchp-spdifrx.c:455:3: error: variable 'mr' is uninitialized when used here [-Werror,-Wuninitialized]
mr |= SPDIFRX_MR_ENDIAN_BIG;
^~
../sound/soc/atmel/mchp-spdifrx.c:432:8: note: initialize the variable 'mr' to silence this warning
u32 mr;
^
= 0
1 error generated.
Zero initialize mr so that these bitwise OR and assignment operation
works unconditionally.
Fixes: fa09fa6038 ("ASoC: mchp-spdifrx: fix controls which rely on rsr register")
Link: https://github.com/ClangBuiltLinux/linux/issues/1797
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Reviewed-by: Claudiu Beznea <claudiu.beznea@microchip.com>
Link: https://lore.kernel.org/r/20230202-mchp-spdifrx-fix-uninit-mr-v1-1-629a045d7a2f@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
As interrupts are Level-triggered,unless and until we deassert the register
the interrupts are generated which causes spurious interrupts unhandled.
Now we deasserted the interrupt at top half which solved the below
"nobody cared" warning.
warning reported in dmesg:
irq 80: nobody cared (try booting with the "irqpoll" option)
CPU: 5 PID: 2735 Comm: irq/80-AudioDSP
Not tainted 5.15.86-15817-g4c19f3e06d49 #1 1bd3fd932cf58caacc95b0504d6ea1e3eab22289
Hardware name: Google Skyrim/Skyrim, BIOS Google_Skyrim.15303.0.0 01/03/2023
Call Trace:
<IRQ>
dump_stack_lvl+0x69/0x97
__report_bad_irq+0x3a/0xae
note_interrupt+0x1a9/0x1e3
handle_irq_event_percpu+0x4b/0x6e
handle_irq_event+0x36/0x5b
handle_fasteoi_irq+0xae/0x171
__common_interrupt+0x48/0xc4
</IRQ>
handlers:
acp_irq_handler [snd_sof_amd_acp] threaded [<000000007e089f34>] acp_irq_thread [snd_sof_amd_acp]
Disabling IRQ #80
Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy@amd.com>
Link: https://lore.kernel.org/r/20230203123254.1898794-1-Vsujithkumar.Reddy@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Mario Limonciello <mario.limonciello@amd.com>:
It's been reported that a number of laptops have a low volume
level from the digital microphone compared to Windows.
AMD offers a register that can adjust the gain for PDM which is not
configured at maximum gain by default.
To fix this change the default for all 3 drivers to raise the gain
but also offer a module parameter. The module parameter can be used
for debugging if the gain is too high on a given laptop.
This is intentionally split into multiple patches for default and
parameter so that if the default really does behave better universally
we can bring it back to stable too later.
In case of regressions for any users that the new pdm_gain value is
too high and for additional debugging, introduce a module parameter
that would let them configure it.
This parameter should be removed in the future:
* If it's determined that the parameter is not needed, just hardcode
the correct value as before
* If users do end up using it to debug and report different values
we should introduce a config knob that can have policy set by ucm.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-7-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
No issues have been reported yet for DMIC audio level on ps platforms,
but as problems were found both on YC (Rembrandt) and Renoir based
designs it's very likely they happen on ps too.
Increase the PDM gain to solve this problem.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-6-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of regressions for any users that the new pdm_gain value is
too high and for additional debugging, introduce a module parameter
that would let them configure it.
This parameter should be removed in the future:
* If it's determined that the parameter is not needed, just hardcode
the correct value as before
* If users do end up using it to debug and report different values
we should introduce a config knob that can have policy set by ucm.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-5-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A similar issue that was reported on Rembrandt based laptops with
low DMIC volume is also being reported for Barcelo based laptops
that use renoir acp3x.
Increase the PDM gain to overcome this problem.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-4-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of regressions for any users that the new pdm_gain value is
too high and for additional debugging, introduce a module parameter
that would let them configure it.
This parameter should be removed in the future:
* If it's determined that the parameter is not needed, just hardcode
the correct value as before
* If users do end up using it to debug and report different values
we should introduce a config knob that can have policy set by ucm.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-3-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A number of users for Lenovo Rembrandt based laptops are
reporting that the microphone is too quiet relative to
Windows with a dual boot.
Increase the PDM gain to overcome this problem.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-2-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Using the control_unload for graph type of elem will lead surprises on
module unload.
The correct callback to use is the dapm_route_unload.
Fixes: 31e9273912 ("ASoC: topology: Use unload() op directly")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230201112846.27707-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The conversion to use generic helpers missed the else for the dai
direction check which leads to failure when loading playback widgets
Fixes: 323f09a61d ("ASoC: sof: use helper function")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20230201112846.27707-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A bit higher volume of changes than wished, but each change is
relatively small and the fix targets are mostly device-specific,
so those should be safe as a late stage merge.
The most significant LoC is about the memalloc helper fix, which
is applied only to Xen PV. The other major parts are ASoC Intel
SOF and AVS fixes that are scattered as various small code
changes. The rest are device-specific fixes and quirks for HD-
and USB-audio, FireWire and ASoC AMD / HDMI.
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Merge tag 'sound-6.2-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A bit higher volume of changes than wished, but each change is
relatively small and the fix targets are mostly device-specific, so
those should be safe as a late stage merge.
The most significant LoC is about the memalloc helper fix, which is
applied only to Xen PV. The other major parts are ASoC Intel SOF and
AVS fixes that are scattered as various small code changes. The rest
are device-specific fixes and quirks for HD- and USB-audio, FireWire
and ASoC AMD / HDMI"
* tag 'sound-6.2-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (30 commits)
ALSA: firewire-motu: fix unreleased lock warning in hwdep device
ALSA: memalloc: Workaround for Xen PV
ASoC: cs42l56: fix DT probe
ASoC: codecs: wsa883x: correct playback min/max rates
ALSA: hda/realtek: Add Acer Predator PH315-54
ASoC: amd: yc: Add Xiaomi Redmi Book Pro 15 2022 into DMI table
ALSA: hda: Do not unset preset when cleaning up codec
ASoC: SOF: sof-audio: prepare_widgets: Check swidget for NULL on sink failure
ASoC: hdmi-codec: zero clear HDMI pdata
ASoC: SOF: ipc4-mtrace: prevent underflow in sof_ipc4_priority_mask_dfs_write()
ASoC: Intel: sof_ssp_amp: always set dpcm_capture for amplifiers
ASoC: Intel: sof_nau8825: always set dpcm_capture for amplifiers
ASoC: Intel: sof_cs42l42: always set dpcm_capture for amplifiers
ASoC: Intel: sof_rt5682: always set dpcm_capture for amplifiers
ALSA: hda/via: Avoid potential array out-of-bound in add_secret_dac_path()
ALSA: usb-audio: Add FIXED_RATE quirk for JBL Quantum610 Wireless
ALSA: hda/realtek: fix mute/micmute LEDs, speaker don't work for a HP platform
ASoC: SOF: keep prepare/unprepare widgets in sink path
ASoC: SOF: sof-audio: skip prepare/unprepare if swidget is NULL
ASoC: SOF: sof-audio: unprepare when swidget->use_count > 0
...
The ucb1400 MFD driver and its gpio and touchscreen child
drivers were only used on a few PXA machines that were unused
for a while and are now removed.
Removing these leaves the AC97 support as ALSA specific,
no other drivers are now connected through this interface.
Cc: Linus Walleij <linus.walleij@linaro.org>
Cc: Bartosz Golaszewski <brgl@bgdev.pl>
Cc: Dmitry Torokhov <dmitry.torokhov@gmail.com>
Cc: Lee Jones <lee@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Marek Vasut <marex@denx.de>
Cc: linux-kernel@vger.kernel.org
Cc: linux-gpio@vger.kernel.org
Cc: linux-input@vger.kernel.org
Cc: alsa-devel@alsa-project.org
Acked-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Most PXA/MMP boards were removed, so the board specific ASoC
support is no longer needed, leaving only support for DT
based ones, as well as the "gumstix" and "spitz" machines
that may get converted to DT later.
Cc: Ian Molton <spyro@f2s.com>
Cc: Ken McGuire <kenm@desertweyr.com>
Cc: Marek Vasut <marek.vasut@gmail.com>
Cc: Mike Rapoport <rppt@kernel.org>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Acked-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Both the DAC and ADC have digital gain controls available
for their mixers, which go from -31 to 0db by step of 1db.
Signed-off-by: Christophe Branchereau <cbranchereau@gmail.com>
Link: https://lore.kernel.org/r/20230122210703.2552384-1-cbranchereau@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Samsung Galaxy Book2 Pro 360 (13" 2022 NP930QED-KA1FR) with codec SSID
144d:ca03 requires the same workaround for enabling the speaker amp
like other Samsung models with ALC298 codec.
Cc: <stable@vger.kernel.org>
Signed-off-by: Guillaume Pinot <texitoi@texitoi.eu>
Link: https://lore.kernel.org/r/20230129171338.17249-1-texitoi@texitoi.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This loop accidentally reuses the "i" iterator for both the inside and
the outside loop. The value of MAX_STREAM_BUFFER is 5. I believe that
chip->rmh.stat_len is in the 2-12 range. If the value of .stat_len is
4 or more then it will loop exactly one time, but if it's less then it
is a forever loop.
It looks like it was supposed to combined into one loop where
conditions are checked.
Fixes: 8e6320064c ("ALSA: lx_core: Remove useless #if 0 .. #endif")
Signed-off-by: Dan Carpenter <error27@gmail.com>
Link: https://lore.kernel.org/r/Y9jnJTis/mRFJAQp@kili
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is tested on V4H White Hawk + ARD-AUDIO-DA7212
Signed-off-by: Linh Phung <linh.phung.jy@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o7qe5ej5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch tidyups rsnd_dma_probe(), but there is no effect.
This is prepare for Gen4 support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r0va5elq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ADG need to know output rate of 44.1kHz/48kHz.
It is using single variable for each, but this patch changes
it to array. Nothing is changed by this patch.
This is prepare for R-Car Gen4 support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tu065em3.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current adg.c is assuming number of clkin/clkout are fixed, but it is
not correct on Gen4. This patch uses clkin/out_size to handling it.
This is prepare for R-Car Gen4 support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v8km5em7.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch moves clkout_name to top of the file to handling both
clkin/clkout in the same place.
This is prepare for R-Car Gen4 support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wn525emc.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current adg.c is usig "clk" as clock IN, but is using "clkout" for
clock OUT. This patch arranges "clk" to "clkin".
This is prepare for R-Car Gen4 support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y1pi5emh.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The flag LRCLK_ASYNC / AUDIO_OUT_48 had been added to handling
special case of Salvator-X board, but it is not used on upstream.
It makes code complex today, let's remove these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zg9y5emm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some SoC can't handle all requested hw rule. In such case, it will indicate
like below, but it is unclear why it didn't work to user.
This patch indicates warning in such case once, because player will try to
similar rule many times.
# aplay sound.wav
Playing WAVE 'sound.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
aplay: aplay.c: 1359: set_params: Assertion `err >= 0' failed.
Aborted by signal Aborted...
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87357q6t7b.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_ssi_master_clk_start() indicates error message if it couldn't
handle requested clock/rate, but it is not caring all cases.
This patch cares it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874js66t7g.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit b43b8ae87c ("ASoC: rsnd: protect mod->status") removed
RSND_DEBUG_NO_DAI_CALL and rsnd_dbg_dai_call(), but these are still
exist on rsnd.h. This patch removes it.
Fixes: b43b8ae87c ("ASoC: rsnd: protect mod->status")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875ycm6t7l.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 1f9c82b5ab ("ASoC: rsnd: add debugfs support") added
CONFIG_DEBUG_FS related definitions on rsnd.h, but it should be
added inside of RSND_H. This patch fixup it.
Fixes: 1f9c82b5ab ("ASoC: rsnd: add debugfs support")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877cx26t7r.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd sets "channels_min" which is used from
snd_soc_dai_stream_valid() without checking DT playback/capture property.
Thus, "aplay -l" or "arecord -l" will indicate un-exising device.
This patch checks DT proerty and do nothing playback/capture settings if
not exist.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/878rhi6t7x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Set driver name to allow matching different UCM2 configurations
for the multiple devices sharing the same APQ8096 ASoC.
Signed-off-by: Yassine Oudjana <y.oudjana@protonmail.com>
Link: https://lore.kernel.org/r/20220622061106.35071-1-y.oudjana@protonmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Stefan Binding <sbinding@opensource.cirrus.com>:
The CS42L42 has a SoundWire interface for control and audio. This
chain of patches adds support for this.
Patches #1 .. #5 split out various changes to the existing code that
are needed for adding Soundwire. These are mostly around clocking and
supporting the separate probe and enumeration stages in SoundWire.
Patches #6 .. #8 actually adds the SoundWire handling.
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
struct snd_soc_dai need to have info for playback/capture,
but it is using "playback/capture_xxx" or "tx/tx_xxx" or array.
This kind of random definition is very difficult to read.
This patch-set add helper functions and each driver use it.
And cleanup the definition.
Merge series from Claudiu Beznea <claudiu.beznea@microchip.com>:
This series adds runtime PM support for Microchip SPDIFRX driver.
Along with it I added few fixes identified while going though the code
and playing with Microchip SPDIFRX controller.
Merge series from wangweidong.a@awinic.com:
The Awinic AW88395 is an I2S/TDM input, high efficiency
digital Smart K audio amplifier with an integrated 10.25V
smart boost converter.
Add a DT schema for describing Awinic AW88395 audio amplifiers. They are
controlled using I2C
Merge series from Herve Codina <herve.codina@bootlin.com>:
The Renesas IDT821034 codec is four channel PCM codec with on-chip
filters and programmable gain setting. It also provides SLIC
(Subscriber Line Interface Circuit) signals as GPIOs.
Since clock stop causes bus reset on Intel controllers, we need
to wait for the debounce interval on resume, to ensure all the
interrupt status registers are set correctly.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-9-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
idle_bias_on was set because cs42l42 has a "VMID" type pseudo-midrail
supply (named FILT+), and these typically take a long time to charge.
But the driver never enabled pm_runtime so it would never have powered-
down the cs42l42 anyway.
In fact, FILT+ can charge to operating voltage within 12.5 milliseconds
of enabling HP or ADC. This time is already covered by the startup
delay of the HP/ADC.
The datasheet warning about FILT+ taking up to 1 second to charge only
applies in the special cases that either the PLL is started or
DETECT_MODE set to non-zero while both HP and ADC are off. The driver
never does either of these.
Removing idle_bias_on allows the Soundwire host controller to suspend
if there isn't a snd_soc_jack handler registered.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-8-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds support for using CS42L42 as a SoundWire device.
SoundWire-specifics are kept separate from the I2S implementation as
much as possible, aiming to limit the risk of breaking the I2C+I2S
support.
There are some important differences in the silicon behaviour between
I2S and SoundWire mode that are reflected in the implementation:
- ASP (I2S) most not be used in SoundWire mode because the two interfaces
share pins.
- The SoundWire capture (record) port only supports 1 channel. It does
not have left-to-right duplication like the ASP.
- DP2 can only be prepared if the HP has powered-up. DP1 can only be
prepared if the ADC has powered-up. (This ordering restriction does
not exist for ASPs.) The SoundWire core port-prepare step is
triggered by the DAI-link prepare(). This happens before the
codec DAI prepare() or the DAPM sequence so these cannot be used
to enable HP/ADC. Instead the HP/ADC enable/disable are done during
the port_prep callback.
- The SRCs are an integral part of the audio chain but in silicon their
power control is linked to the ASP. There is no equivalent power link
to SoundWire DPs so the driver must take "manual" control of SRC power.
- The SoundWire control registers occupy the lower part of the SoundWire
address space so cs42l42 registers are offset by 0x8000 (non-paged) in
SoundWire mode.
- Register addresses are 8-bit paged in I2C mode but 16-bit unpaged in
SoundWire.
- Special procedures are needed on register read/writes to (a) ensure
that the previous internal bus transaction has completed, and
(b) handle delayed read results, when the read value could not be
returned within the SoundWire read command.
There are also some differences in driver implementation between I2S
and SoundWire operation:
- CS42L42 I2S does not runtime_suspend, but runtime_suspend/resume support
has been added into the driver in SoundWire mode as the most convenient
way to power-up the bus manager and to handle the unattach_request
condition, though the CS42L42 chip does not itself suspend or resume.
- Intel SoundWire host controllers have a low-power clock-stop mode that
requires resetting all peripherals when resuming. This means that the
interrupt registers will be reset in between the interrupt being
generated and the interrupt being handled, and since the interrupt
status is debounced, these values may not be accurate immediately,
and may cause spurious unplug events before settling.
- As in I2S mode, the PLL is only used while audio is active because
of clocking quirks in the silicon. For SoundWire the cs42l42_pll_config()
is deferred until the DAI prepare(), to allow the cs42l42_bus_config()
callback to set the SCLK.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-7-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Export functions that will be needed by a SoundWire module.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-6-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
