| Age | Commit message (Collapse) | Author | Files | Lines |
|---|
|
It turned out that VIA codecs also mute the sound in the lowest mixer
level. Turn on the dac_min_mute flag to indicate the mute-as-minimum
in TLV like already done in Conexant and IDT codecs.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210559
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210114072453.11379-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The Acer Apire E5-575T laptop with codec ALC255 has a terrible
background noise comes from internal mic capture. And the jack
sensing dose not work for headset like some other Acer laptops.
This patch limits the internal mic boost on top of the existing
ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk for Acer Aspire E5-575T.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210114082728.74729-1-chiu@endlessos.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Since the commit 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for
implicit fb sync"), we apply the hw constraints for the implicit
feedback sync to make the secondary open aligned with the already
opened stream setup. This change assumed that the secondary open is
performed after the first stream has been already set up, and adds the
hw constraints to sync with the first stream's parameters only when
the EP setup for the first stream was confirmed at the open time.
However, most of applications handling the full-duplex operations do
open both playback and capture streams at first, then set up both
streams. This results in skipping the additional hw constraints since
the counter-part stream hasn't been set up yet at the open of the
second stream, and it eventually leads to "incompatible EP" error in
the end.
This patch corrects the behavior by always applying the hw constraints
for the implicit fb sync. The hw constraint rules are defined so that
they check the sync EP dynamically at each invocation, instead. This
covers the concurrent stream setups better and lets the hw refine
calls resolving to the right configuration.
Also this patch corrects a minor error that has existed in the debug
print that isn't built as default.
Fixes: 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync")
Link: https://lore.kernel.org/r/20210111081611.12790-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
|
|
Invoke hda_dsp_remove() as the .shutdown() callback. This will help to
perform shutdown of the DSP safely on TGL platforms before shutting down
or rebooting the system.
BugLink: https://github.com/thesofproject/linux/issues/2571
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210113152617.4048541-4-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Add the .shutdown() callback to the sof-pci-dev driver, to help to
handle shutting down specific tasks for SOF PCI platforms.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210113152617.4048541-3-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Add helper snd_sof_device_shutdown() to wrap the platform specific
.shutdown callbacks for SOF platforms.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210113152617.4048541-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Add .shutdown() callback to the struct snd_sof_dsp_ops, for
doing platform specific actions at shutdown.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210113152617.4048541-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The earlier commit to fix runtime PM in case i915 init fails,
introduces a possibility to hit a page fault.
snd_hdac_ext_bus_device_exit() is designed to be called from
dev.release(). Calling it outside device reference counting, is
not safe and may lead to calling the device_exit() function
twice. Additionally, as part of ext_bus_device_init(), the device
is also registered with snd_hdac_device_register(). Thus before
calling device_exit(), the device must be removed from device
hierarchy first.
Fix the issue by rolling back init actions by calling
hdac_device_unregister() and then releasing device with put_device().
This matches with existing code in hdac-ext module.
To complete the fix, add handling for the case where
hda_codec_load_module() returns -ENODEV, and clean up the hdac_ext
resources also in this case.
In future work, hdac-ext interface should be extended to allow clients
more flexibility to handle the life-cycle of individual devices, beyond
just the current snd_hdac_ext_bus_device_remove(), which removes all
devices.
BugLink: https://github.com/thesofproject/linux/issues/2646
Reported-by: Jaroslav Kysela <perex@perex.cz>
Fixes: 6c63c954e1c5 ("ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Link: https://lore.kernel.org/r/20210113150715.3992635-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Add HD Audio PCI ID and HDMI codec vendor ID for Intel AlderLake-P.
Signed-off-by: Kai Vehmanen <kai.vehmanen@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: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210113155629.4097057-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Hi Mark
These are not so important, but for
soc-pcm cleanup patches.
Kuninori Morimoto (6):
ASoC: soc-pcm: move dpcm_set_fe_update_state()
ASoC: soc-pcm: add dpcm_set_be_update_state()
ASoC: soc-pcm: add soc_pcm_set_dai_params()
ASoC: soc-pcm: cleanup soc_pcm_apply_symmetry()
ASoC: soc-pcm: cleanup soc_pcm_params_symmetry()
ASoC: soc-pcm: setup pcm at one place in soc_new_pcm()
sound/soc/soc-pcm.c | 231 +++++++++++++++++---------------------------
1 file changed, 90 insertions(+), 141 deletions(-)
--
2.25.1
Thank you for your help !!
Best regards
---
Kuninori Morimoto
|
|
System takes a very long time to suspend after commit 215a22ed31a1
("ALSA: hda: Refactor codec PM to use direct-complete optimization"):
[ 90.065964] PM: suspend entry (s2idle)
[ 90.067337] Filesystems sync: 0.001 seconds
[ 90.185758] Freezing user space processes ... (elapsed 0.002 seconds) done.
[ 90.188713] OOM killer disabled.
[ 90.188714] Freezing remaining freezable tasks ... (elapsed 0.001 seconds) done.
[ 90.190024] printk: Suspending console(s) (use no_console_suspend to debug)
[ 90.904912] intel_pch_thermal 0000:00:12.0: CPU-PCH is cool [49C], continue to suspend
[ 321.262505] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5
[ 328.426919] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5
[ 329.490933] ACPI: EC: interrupt blocked
That commit keeps the codec suspended during the system suspend. However,
mute/micmute LED will clear codec's direct-complete flag by
dpm_clear_superiors_direct_complete().
This doesn't play well with SOF driver. When its runtime resume is
called for system suspend, hda_codec_jack_check() schedules
jackpoll_work which uses snd_hdac_is_power_on() to check whether codec
is suspended. Because the direct-complete path isn't taken,
pm_runtime_disable() isn't called so snd_hdac_is_power_on() returns
false and jackpoll continues to run, and snd_hda_power_up_pm() cannot
power up an already suspended codec in multiple attempts, causes the
long delay on system suspend:
if (dev->power.direct_complete) {
if (pm_runtime_status_suspended(dev)) {
pm_runtime_disable(dev);
if (pm_runtime_status_suspended(dev)) {
pm_dev_dbg(dev, state, "direct-complete ");
goto Complete;
}
pm_runtime_enable(dev);
}
dev->power.direct_complete = false;
}
When direct-complete path is taken, snd_hdac_is_power_on() returns true
and hda_jackpoll_work() is skipped by accident. So this is still not
correct.
If we were to use snd_hdac_is_power_on() in system PM path,
pm_runtime_status_suspended() should be used instead of
pm_runtime_suspended(), otherwise pm_runtime_{enable,disable}() may
change the outcome of snd_hdac_is_power_on().
Because devices suspend in reverse order (i.e. child first), it doesn't
make much sense to resume an already suspended codec from audio
controller. So avoid the issue by making sure jackpoll isn't used in
system PM process.
Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210112181128.1229827-3-kai.heng.feng@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Modify hda_codec_jack_wake_enable() to also support disable WAKEEN.
In addition, this patch also moves the WAKEEN disablement call out of
hda_codec_jack_check() into hda_codec_jack_wake_enable().
This is a preparation for next patch.
No functional change intended.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210112181128.1229827-2-kai.heng.feng@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Instead of queueing jackpoll_work, runtime resume the codec to let it
use different jack detection methods based on jackpoll_interval.
This partially matches SOF driver's behavior with commit a6e7d0a4bdb0
("ALSA: hda: fix jack detection with Realtek codecs when in D3"), the
difference is SOF unconditionally resumes the codec.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210112181128.1229827-1-kai.heng.feng@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Instead of manually managing its DMA buffers using
dma_{alloc,free}_coherent() lets the sound core take care of this using
managed buffers.
On one hand this reduces the amount of boiler plate code, but the main
motivation for the change is to use the shared code where possible. This
makes it easier to argue about correctness and that the code does not
contain subtle bugs like data leakage or similar.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20210106133650.13509-3-lars@metafoo.de
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Instead of manually managing its DMA buffers using
dma_{alloc,free}_coherent() lets the sound core take care of this using
managed buffers.
On one hand this reduces the amount of boiler plate code, but the main
motivation for the change is to use the shared code where possible. This
makes it easier to argue about correctness and that the code does not
contain subtle bugs like data leakage or similar.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20210106133650.13509-2-lars@metafoo.de
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Instead of manually managing its DMA buffers using
dma_{alloc,free}_coherent() lets the sound core take care of this using
managed buffers.
On one hand this reduces the amount of boiler plate code, but the main
motivation for the change is to use the shared code where possible. This
makes it easier to argue about correctness and that the code does not
contain subtle bugs like data leakage or similar.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20210106133650.13509-1-lars@metafoo.de
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
At the moment it is necessary to set up the DAPM routes between
front-end AIF<->DAI explicitly in the device tree, e.g. using
audio-routing =
"MM_DL1", "MultiMedia1 Playback",
"MM_DL3", "MultiMedia3 Playback",
"MM_DL4", "MultiMedia4 Playback",
"MultiMedia2 Capture", "MM_UL2";
This is prone to mistakes and (sadly) there is no clear error if one
of these routes is missing. :(
Actually, this should not be necessary because the ASoC core normally
automatically links AIF<->DAI within snd_soc_dapm_link_dai_widgets().
This is done using the "stname" parameter of SND_SOC_DAPM_AIF_IN/OUT.
For SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
it should create the route from above: MM_DL1 <-> MultiMedia1 Playback.
This does not work at the moment because the AIF widget (MM_DL1)
and the DAI widget (MultiMedia1 Playback) belong to different
DAPM contexts (q6routing / q6asm-dai).
Fix this by declaring the AIF widgets in the same driver as the DAIs
(q6asm-dai). Now the routes above are created automatically
and no longer need to be specified in the device tree.
This is also more consistent with the back-end AIFs which are already
declared in q6afe-dais instead of q6routing. q6routing should only link
the components together using mixers.
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20201211203255.148246-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
There are two issues with this code. The first error path forgot to set
the error code and instead returns success. The second error path
doesn't clean up.
Fixes: 272b5edd3b8f ("ASoC: Add support for CS42L56 CODEC")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/X9NE/9nK9/TuxuL+@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
soc_pcm_apply_symmetry() want to call snd_pcm_hw_constraint_single()
for rate/channel/sample_bits, but, it needs many condition check.
These are very similar but different, thus, it needs to have very
verbose code.
This patch use macro for it and make code more simple.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wnxo7uyq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Getting rate/channels/sample_bits from param needs fixed method.
This patch adds new soc_pcm_set_dai_params() and replace existing code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y2i47uyw.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
soc-pcm has dpcm_set_fe_update_state() to update FE's runtime_update
(except dpcm_fe_dai_do_trigger() which needs to update it without it).
OTOH, it doesn't have BE's update function.
O: dpcm_set_fe_update_state()
X: dpcm_set_be_update_state()
This patch add BE's dpcm_set_fe_update_state()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zh2k7uz1.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This patch moves dpcm_set_fe_update_state() to top side.
This is prepare for cleanup soc-pcm.c
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/871rfw99jn.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Hi All,
This series adds support for devices with only a headphone jack
(no speakers/internal mic). Specifically this adds support for the
Mele PCG03 Mini PC. But the new no-speakers and no-internal-mic quirks
will likely be useful on other devices too.
Regards,
Hans
|
|
There could be more than one thread read/write the dsp_power_state
simultaneously (e.g. hda_dsp_d0i3_work and sof_ipc_tx_message), add a
mutex power_state_access to make sure the access to it is mutually
exclusive.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210105155640.3725238-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
We are able to power down the GPU and audio via the GPU driver
so flag these asics as supporting runtime pm.
Reviewed-by: Evan Quan <evan.quan@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Link: https://lore.kernel.org/r/20210105175245.963451-1-alexander.deucher@amd.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
sound/soc/soc-core.c: soc_remove_component() unconditionally calls
snd_soc_component_set_jack(component, NULL, NULL); on any components
being removed.
This means that on machines where the machine-driver does not provide
a jack through snd_soc_component_set_jack() es8316_disable_jack_detect()
will still get called and at this time es8316->jack will be NULL and
the es8316->jack->status check in es8316_disable_jack_detect() will
lead to a NULL pointer deref.
Fix this by checking for es8316->jack bein NULL at the start of
es8316_disable_jack_detect() and turn the function into a no-op in
that case.
Cc: russianneuromancer <russianneuromancer@ya.ru>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210112101725.44200-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Because clk_disable_unprepare() already checked NULL clock parameter,
so the additional check is unnecessary, just remove it.
Signed-off-by: Xu Wang <vulab@iscas.ac.cn>
Link: https://lore.kernel.org/r/20210108084456.6603-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Move the snd_soc_dai_set_tdm_slot() call from cht_codec_init() to
cht_codec_fixup(). There are 2 reasons for doing this:
1. This aligns the cht_bsw_nau8824 with all the other BYT/CHT machine
drivers which also do this from their codec_fixup function.
2. When using the SOF driver, things like the TDM info is set from the
topology file. Moving the call to the codec_fixup function, which gets
skipped when using the SOF driver avoids the call interfering with the
settings when using the SOF driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210107115324.11602-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Add a quirk for the Mele PCG03 Mini PC, being a Mini PC this device
has no speakers and no internal microphone.
To make matters worse the speaker output pins are shorted (to gnd or
to each other?) and SPKVDD is provided. So trying to output sound on the
speakers leads to shorting SPKVDD, this leads to a power dip after
which the codec is an unknown state. Sometimes it drops of the i2c
bus, sometimes it does still respond to i2c transfers, but is otherwise
not functional. TL;DR: trying to use the speaker outputs on this model
is BAD.
Besides not having speakers / an internal mic, this is a Bay Trail CR
device without a CHAN package in ACPI, so we default to SSP0-AIF2 as
codec connection. But the device is actually using SSP0-AIF1, so we
need to quirk that too.
Cc: Rasmus Porsager <rasmus@beat.dk>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210109210119.159032-4-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Some devices, like mini PCs/media/top-set boxes do not have an internal
microphone at all, an example of the is the Mele PCG03 Mini PC.
Add a new BYT_RT5640_NO_INTERNAL_MIC_MAP input-mapping for this,
which does not add any internal-mic routes and modifies the components
and the (optional) long_name strings to reflect this.
Cc: Rasmus Porsager <rasmus@beat.dk>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210109210119.159032-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Some devices, like mini PCs/media/top-set boxes do not have any speakers
at all, an example of the is the Mele PCG03 Mini PC.
Add a new BYT_RT5640_NO_SPEAKERS quirk-flag which when sets does not add
speaker routes and modifies the components and the (optional) long_name
strings to reflect that there are no speakers.
Cc: Rasmus Porsager <rasmus@beat.dk>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210109210119.159032-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
As snd_fw_async_midi_port.consume_bytes is unsigned int, and
NSEC_PER_SEC is 1000000000L, the second multiplication in
port->consume_bytes * 8 * NSEC_PER_SEC / 31250
always overflows on 32-bit platforms, truncating the result. Fix this
by precalculating "NSEC_PER_SEC / 31250", which is an integer constant.
Note that this assumes port->consume_bytes <= 16777.
Fixes: 531f471834227d03 ("ALSA: firewire-lib/firewire-tascam: localize async midi port")
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20210111130251.361335-3-geert+renesas@glider.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
As snd_ff.rx_bytes[] is unsigned int, and NSEC_PER_SEC is 1000000000L,
the second multiplication in
ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250
always overflows on 32-bit platforms, truncating the result. Fix this
by precalculating "NSEC_PER_SEC / 31250", which is an integer constant.
Note that this assumes ff->rx_bytes[port] <= 16777.
Fixes: 19174295788de77d ("ALSA: fireface: add transaction support")
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20210111130251.361335-2-geert+renesas@glider.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Currently hda on tegra30 fails to open a stream with an input/output error.
For example:
speaker-test -Dhw:0,3 -c 2
speaker-test 1.2.2
Playback device is hw:0,3
Stream parameters are 48000Hz, S16_LE, 2 channels
Using 16 octaves of pink noise
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 64 to 16384
Period size range from 32 to 8192
Using max buffer size 16384
Periods = 4
was set period_size = 4096
was set buffer_size = 16384
0 - Front Left
Write error: -5,Input/output error
xrun_recovery failed: -5,Input/output error
Transfer failed: Input/output error
The tegra-hda device was introduced in tegra30 but only utilized in
tegra124 until recent chips. Tegra210/186 work only due to a hardware
change. For this reason it is unknown when this issue first manifested.
Discussions with the hardware team show this applies to all current tegra
chips. It has been resolved in the tegra234, which does not have hda
support at this time.
The explanation from the hardware team is this:
Below is the striping formula referenced from HD audio spec.
{ ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
The current issue is seen because Tegra HW has a problem with boundary
condition (= 8) for striping. The reason why it is not seen on
Tegra210/Tegra186 is because it uses max 2SDO lines. Max SDO lines is
read from GCAP register.
For the given stream (channels = 2, bps = 16);
ratio = (channels * bps) / NSDO = 32 / NSDO;
On Tegra30, ratio = 32/4 = 8 (FAIL)
On Tegra210/186, ratio = 32/2 = 16 (PASS)
On Tegra194, ratio = 32/4 = 8 (FAIL) ==> Earlier workaround was
applied for it
If Tegra210/186 is forced to use 4SDO, it fails there as well. So the
behavior is consistent across all these chips.
Applying the fix in [1] universally resolves this issue on tegra30-hda.
Tested on the Ouya game console and the tf201 tablet.
[1] commit 60019d8c650d ("ALSA: hda/tegra: workaround playback failure on
Tegra194")
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Tested-by: Ion Agorria <ion@agorria.com>
Reviewed-by: Sameer Pujar <spujar@nvidia.com>
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Peter Geis <pgwipeout@gmail.com>
Link: https://lore.kernel.org/r/20210108135913.2421585-3-pgwipeout@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
These patches are trying to fix the jack detection and internal
microphone problems on ECS EF20 series laptops which are empowered
by Intel Atom x5-Z8350 CPU (CherryTrail) with Realtek rt5645 audio
codec.
---
v2 -> v3:
Restore the accidentally removed terminator of the
dmi_platform_data[].
v1 -> v2:
Invoke callback() of the DMI quirk if it exists, because
the dmi_first_match() doesn't.
---
Chris Chiu (4):
ASoC: rt5645: Introduce mapping for ACPI-defined GPIO
ASoC: rt5645: Add ACPI-defined GPIO for ECS EF20 series
ASoC: rt5645: add inv_hp_det flag
ASoC: rt5645: Enable internal microphone and JD on ECS EF20
include/sound/rt5645.h | 2 ++
sound/soc/codecs/rt5645.c | 45 +++++++++++++++++++++++++++++++++++++++
2 files changed, 47 insertions(+)
--
2.20.1
|
|
v2: Update patch to align with latest kernel release.
v1: Initial patch version, to enable DMA mode on Intel Keembay platform.
Michael Sit Wei Hong (2):
dt-bindings: sound: intel, keembay-i2s: Add info for device to use DMA
ASoC: Intel: KMB: Enable DMA transfer mode
.../bindings/sound/intel,keembay-i2s.yaml | 14 ++
sound/soc/intel/Kconfig | 2 +
sound/soc/intel/keembay/kmb_platform.c | 157 ++++++++++++++++--
sound/soc/intel/keembay/kmb_platform.h | 9 +
4 files changed, 167 insertions(+), 15 deletions(-)
--
2.17.1
|
|
The error path here doesn't set "ret" so it returns uninitialized data
instead of a negative error code.
Fixes: 2c1382840c19 ("ASoC: soc-pcm: disconnect BEs if the FE is not ready")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/X/wfXQFxeMLvpO+1@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
wm_adsp_read_data_word() used if (ret) to check for an error from
wm_adsp_read_raw_data_block(). While this is perfectly valid,
wm_adsp_read_raw_data_block() itself uses if (ret < 0) and three
calls to wm_adsp_read_data_word() also use if (ret < 0).
This creates an error check chain like this:
1st) if (ret < 0) return ret;
2nd) if (ret) return ret;
3rd) if (ret < 0) ...
This can confuse the compiler into thinking that there are possible
returns > 0 from the middle if() that are not handled by the final
if(). If this was true it would lead to using uninitialized variables
later in the outer function.
Fix this by changing the test in wm_adsp_read_data_word() to be
if (ret < 0).
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210111133825.8758-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Kernel test robot throws below error ->
sound/soc/soc-pcm.c:2523 dpcm_run_update_startup() error: uninitialized
symbol 'ret'.
Initializing ret = 0 and returning correct -ERRNO in failure path.
Reported-by: kernel test robot <lkp@intel.com>
Signed-off-by: Souptick Joarder <jrdr.linux@gmail.com>
Link: https://lore.kernel.org/r/1610163901-5523-1-git-send-email-jrdr.linux@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
To avoid the pop noise, this patch enables the feature of fast discharge.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20210111092544.9064-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This patch will be the workaround to fix getting the wrong device ID on the rare chance.
It seems like something unstable when the system resumes. e.g. the bus clock
This patch tries to read the device ID to check several times.
After the test, the driver will get the correct device ID the second time.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20210111092740.9128-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
On ECS EF20 series laptops, the internal mic is on DMIC2/IN2P.
And they need the inv_hp_det to make jack detection to work as
exoected.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Link: https://lore.kernel.org/r/20210111054141.4668-5-chiu@endlessos.org
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The ECS EF20EA laptop use gpio for jack detection instead of rt5645
rt5645 JD. However, the GPIO polarity is inverse for hp-detect based
on the _DSD property of the RTK2 device.
Name (_DSD, Package () {
ToUUID("daffd814-6eba-4d8c-8a91-bc9bbf4aa301"),
Package () {
Package () {"hp-detect-gpio", Package() {^RTK2, 0, 0, 1 }},
}
})
This flag will invert the hp-detect gpio polarity.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Link: https://lore.kernel.org/r/20210111054141.4668-4-chiu@endlessos.org
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Add the hp-detect gpio for ECS EF20 series laptops based on the
_CRS defined in DSDT table.
Method (_CRS, 0, NotSerialized)
{
Name (SBUF, ResourceTemplate ()
{
I2cSerialBusV2 (0x001A, ControllerInitiated, 0x00061A80,
AddressingMode7Bit, "\\_SB.PCI0.I2C2",
0x00, ResourceConsumer, , Exclusive,
)
GpioInt (Edge, ActiveBoth, SharedAndWake, PullNone, 0x0000,
"\\_SB.GPO3", 0x00, ResourceConsumer, ,
)
{ // Pin list
0x004F
}
GpioIo (Shared, PullDefault, 0x0000, 0x0000, IoRestrictionInputOnly,
"\\_SB.GPO3", 0x00, ResourceConsumer, ,
)
{ // Pin list
0x004F
}
})
Return (SBUF) /* \_SB_.PCI0.I2C2.RTK2._CRS.SBUF */
}
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Link: https://lore.kernel.org/r/20210111054141.4668-3-chiu@endlessos.org
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
On at least one laptop (ECS EF20EA) the 'hp-detect' GPIO is defined in
the DSDT table by the ACPI GpioIo resources in _CRS. The GPIO related
information should be mapped to the rt5645 driver to enable the jack
detection also on non-DT platforms.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Link: https://lore.kernel.org/r/20210111054141.4668-2-chiu@endlessos.org
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Pioneer devices have both playback and capture streams sharing the
same iface/altsetting, and those need to be paired as implicit
feedback. Instead of a half-baked (and broken) static quirk entry,
set up more generically for those devices by checking the number of
endpoints and the attribute of the secondary EP.
Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Reported-by: František Kučera <konference@frantovo.cz>
Link: https://lore.kernel.org/r/20210108075219.21463-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
There are devices that have multiple endpoints sharing the same
iface/altset not only for sync but also for the actual streams, and
the audioformat for such an endpoint needs to be handled with the
proper endpoint index; otherwise it confuses the endpoint management.
This patch extends the audioformat to annotate the endpoint index, and
put the proper ep_idx=1 to Pioneer device quirk entries accordingly.
Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The current endpoint handling assumed (more or less) a unique 1:1
relation between the endpoint and the iface/altset. The exception was
the sync EP without the implicit feedback which has usually the
secondary EP of the same altset. This works fine for most devices,
but it turned out that some unusual devices like Pinoeer's ones have
both playback and capture endpoints in the same iface/altsetting and
use both for the implicit feedback mode. For handling such a case, we
need to extend the endpoint management to take the shared interface
into account.
This patch does that: it adds a new object snd_usb_iface_ref for
managing the reference counts of the each USB interface that is used
by each endpoint. The interface setup is performed only once for the
(sharing) endpoints, and the doubly initialization is avoided.
Along with this, the resource release of endpoints and interface
refcounts are put into a single function, snd_usb_endpoint_free_all()
instead of looping in the caller side.
Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
Takashi Iwai
Chris Chiu
Mark Brown
Keyon Jie
Kai Vehmanen
Kai-Heng Feng
Lars-Peter Clausen
Stephan Gerhold
Dan Carpenter
Kuninori Morimoto
Alex Deucher
Hans de Goede
Xu Wang
Geert Uytterhoeven
Peter Geis
Richard Fitzgerald
Souptick Joarder
Shuming Fan