Random noise could be heard when playing audio to the HDMI output.
This is due to the IEC conversion is invoked in the external loop.
As a result, this additional loop takes up a lot of the processing
cycle.
hdmi_reformat_iec958() process the conversion using an internal loop,
it is safe to move it out from the external loop to avoid unnecessary
processing cycle been spent. Furthermore, ALSA IEC958 plugin works in
32bit format only.
Signed-off-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210421005546.7534-1-jee.heng.sia@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The last two patches in this series fix a longstanding issue that prevented
the ALC3263 codec from using a headset mic. This codec can be found on Dell
systems including the Latitude 13 7350, Venue 11 Pro 7140, and XPS 13 9343.
In fact, there is an ACPI quirk for the XPS 13 9343, which forces it to use
legacy HD Audio just to avoid this issue:
https://lore.kernel.org/alsa-devel/CAPeXnHv07HkvcHrYFmZMr8OTp7U7F=k_k=LPYnUtp89iPn2d2Q@mail.gmail.com/
This may allow that ACPI quirk to be removed. Either way, the other systems
mentioned above do not support this quirk and already use the ASoC driver,
so this fix is necessary for headset mic support on those systems.
Note: there is likely other handling for this codec that only exists in the
HDA driver, but which also belongs in the ASoC driver. Commit 394c97f824
("ALSA: hda/realtek - Change EAPD to verb control") describes an issue that
does not seem to be resolved in the ASoC driver, to give an example.
Other patches in this series are not specific to the ALC3263. These patches
set the correct combo jack configuration when headphones are inserted, and
fix a misaligned value set in the DMIC2 Configuration Default register.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=114171
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961
Signed-off-by: David Ward <david.ward@gatech.edu>
David Ward (5):
ASoC: rt286: Fix upper byte in DMIC2 configuration
ASoC: rt286: Configure combo jack for headphones
ASoC: rt298: Configure combo jack for headphones
ASoC: rt286: Make RT286_SET_GPIO_* readable and writable
ASoC: rt286: Generalize support for ALC3263 codec
sound/soc/codecs/rt286.c | 34 +++++++++++++++++++++-------------
sound/soc/codecs/rt298.c | 9 +++++++--
2 files changed, 28 insertions(+), 15 deletions(-)
--
2.31.1
base-commit: a38fd87484
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.
Based on similar fix in kbl_rt5663_rt5514_max98927.c
from Harsha Priya <harshapriya.n@intel.com> and
Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
Cc: <stable@vger.kernel.org> # 5.4+
Signed-off-by: Lukasz Majczak <lma@semihalf.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210415124347.475432-1-lma@semihalf.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When using the driver in I2S TDM mode, the fsl_esai_startup()
function rewrites the number of slots previously set by the
fsl_esai_set_dai_tdm_slot() function to 2.
To fix this, let's use the saved slot count value or, if TDM
is not used and the number of slots is not set, the driver will use
the default value (2), which is set by fsl_esai_probe().
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
card->owner is a required property and since commit 81033c6b58 ("ALSA:
core: Warn on empty module") a warning is issued if it is empty. Add it.
This fixes following warning observed on Lamobo R1:
WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd]
Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ...
CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1
Hardware name: Allwinner sun7i (A20) Family
Call trace:
(snd_card_new [snd])
(snd_soc_bind_card [snd_soc_core])
(snd_soc_register_card [snd_soc_core])
(sun4i_codec_probe [sun4i_codec])
Fixes: 45fb6b6f2a ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs")
Related: commit 3c27ea23ff ("ASoC: qcom: Set card->owner to avoid warnings")
Related: commit ec653df2a0 ("drm/vc4/vc4_hdmi: fill ASoC card owner")
Cc: linux-arm-kernel@lists.infradead.org
Cc: alsa-devel@alsa-project.org
Signed-off-by: Bastian Germann <bage@linutronix.de>
Link: https://lore.kernel.org/r/20210331151843.30583-1-bage@linutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
0x20FF(amp global enable) register was defined as non-volatile,
but it is not. Overheating, overcurrent can cause amp shutdown
in hardware.
'regmap_write' compare register readback value before writing
to avoid same value writing. 'regmap_read' just read cache
not actual hardware value for the non-volatile register.
When amp is internally shutdown by some reason, next 'AMP ON'
command can be ignored because regmap think amp is already ON.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Link: https://lore.kernel.org/r/20210325033555.29377-1-ryans.lee@maximintegrated.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SST firmware's media and deep-buffer inputs are hardcoded to
S16LE, the corresponding DAIs don't have a hw_params callback and
their prepare callback also does not take the format into account.
So far the advertising of non working S24LE support has not caused
issues because pulseaudio defaults to S16LE, but changing pulse-audio's
config to use S24LE will result in broken sound.
Pipewire is replacing pulse now and pipewire prefers S24LE over S16LE
when available, causing the problem of the broken S24LE support to
come to the surface now.
Cc: stable@vger.kernel.org
BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/866
Fixes: 098c2cd281 ("ASoC: Intel: Atom: add 24-bit support for media playback and capture")
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210324132711.216152-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The input MCLK is 12.288MHz, the desired output sysclk is 11.2896MHz
and sample rate is 44100Hz, with the configuration pllprescale=2,
postscale=sysclkdiv=1, some chip may have wrong bclk
and lrclk output with pll enabled in master mode, but with the
configuration pllprescale=1, postscale=2, the output clock is correct.
>From Datasheet, the PLL performs best when f2 is between
90MHz and 100MHz when the desired sysclk output is 11.2896MHz
or 12.288MHz, so sysclkdiv = 2 (f2/8) is the best choice.
So search available sysclk_divs from 2 to 1 other than from 1 to 2.
Fixes: 84fdc00d51 ("ASoC: codec: wm9860: Refactor PLL out freq search")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1616150926-22892-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With commit 1e30f642cf ("ASoC: simple-card-utils: Fix device module clock")
simple-card-utils can control MCLK clock for rate updates or enable/disable.
But this is breaking some platforms where it is expected that codec drivers
would actually handle the MCLK clock. One such example is following platform.
- "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts"
In above case codec, wm8904, is using internal PLL and configures sysclk
based on fixed MCLK input. In such cases it is expected that, required PLL
output or sysclk, is just passed via set_sysclk() callback and card driver
need not actually update MCLK rate. Instead, codec can take ownership of
this clock and do the necessary configuration.
So the original commit is reverted and codec driver for rt5659 is updated
to fix my board which has this codec.
Sameer Pujar (2):
ASoC: simple-card-utils: Do not handle device clock
ASoC: rt5659: Update MCLK rate in set_sysclk()
sound/soc/codecs/rt5659.c | 5 +++++
sound/soc/generic/simple-card-utils.c | 13 +++++++------
2 files changed, 12 insertions(+), 6 deletions(-)
--
2.7.4
This reverts commit 1e30f642cf ("ASoC: simple-card-utils: Fix device
module clock"). The original patch ended up breaking following platform,
which depends on set_sysclk() to configure internal PLL on wm8904 codec
and expects simple-card-utils to not update the MCLK rate.
- "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts"
It would be best if codec takes care of setting MCLK clock via DAI
set_sysclk() callback.
Reported-by: Michael Walle <michael@walle.cc>
Suggested-by: Mark Brown <broonie@kernel.org>
Suggested-by: Michael Walle <michael@walle.cc>
Fixes: 1e30f642cf ("ASoC: simple-card-utils: Fix device module clock")
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Tested-by: Michael Walle <michael@walle.cc>
Link: https://lore.kernel.org/r/1615829492-8972-2-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During testing John Stultz and Amit reported few array our bound issues
after enabling bound sanitizer
This patch series attempts to fix those!
changes since v1:
- make sure the wcd is not de-referenced without intialization
Srinivas Kandagatla (3):
ASoC: qcom: sdm845: Fix array out of bounds access
ASoC: qcom: sdm845: Fix array out of range on rx slim channels
ASoC: codecs: wcd934x: add a sanity check in set channel map
sound/soc/codecs/wcd934x.c | 6 ++++++
sound/soc/qcom/sdm845.c | 6 +++---
2 files changed, 9 insertions(+), 3 deletions(-)
--
2.21.0
Hi All,
Here is a patch series for reporting to user space jack and button events and
add the support for Capture. With some cleanups and fixes along the way.
Regards,
Lucas Tanure
Lucas Tanure (12):
ASoC: cs42l42: Fix Bitclock polarity inversion
ASoC: cs42l42: Fix channel width support
ASoC: cs42l42: Fix mixer volume control
ASoC: cs42l42: Don't enable/disable regulator at Bias Level
ASoC: cs42l42: Always wait at least 3ms after reset
ASoC: cs42l42: Remove power if the driver is being removed
ASoC: cs42l42: Disable regulators if probe fails
ASoC: cs42l42: Provide finer control on playback path
ASoC: cs42l42: Set clock source for both ways of stream
ASoC: cs42l42: Add Capture Support
ASoC: cs42l42: Report jack and button detection
ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called
Richard Fitzgerald (3):
ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT
ASoC: cs42l42: Only start PLL if it is needed
ASoC: cs42l42: Wait for PLL to lock before switching to it
sound/soc/codecs/cs42l42.c | 437 +++++++++++++++++++++----------------
sound/soc/codecs/cs42l42.h | 41 +++-
2 files changed, 282 insertions(+), 196 deletions(-)
--
2.30.1
Attempting to use the RX MIX path at 48kHz plays at 96kHz, because these
controls are incorrectly toggling the first bit of the register, which
is part of the FS_RATE field.
Fix the problem by using the same method used by the "WSA RX_MIX EC0_MUX"
control, which is to use SND_SOC_NOPM as the register and use an enum in
the shift field instead.
Fixes: 2c4066e5d4 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route")
Signed-off-by: Jonathan Marek <jonathan@marek.ca>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210305005049.24726-1-jonathan@marek.ca
Signed-off-by: Mark Brown <broonie@kernel.org>
An interface can have multiple decimators enabled, so loop over all active
decimators. Otherwise only one channel will be unmuted, and other channels
will be zero. This fixes recording from dual DMIC as a single two channel
stream.
Also remove the now unused "active_decimator" field.
Fixes: 908e6b1df2 ("ASoC: codecs: lpass-va-macro: Add support to VA Macro")
Signed-off-by: Jonathan Marek <jonathan@marek.ca>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210304215646.17956-1-jonathan@marek.ca
Signed-off-by: Mark Brown <broonie@kernel.org>
