Now we can use asoc_substream_to_rtd() macro,
let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87o8ob0yun.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc-xxx are getting rtd from substream by
rtd = substream->private_data;
But, getting data from "private_data" is very unclear.
This patch adds asoc_substream_to_rtd() macro which is
easy to understand that rtd from substream.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wo2z0yve.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
kabylake_ssp_fixup function uses snd_soc_dpcm to identify the
codecs DAIs. The HW parameters are changed based on the codec DAI of the
stream. The earlier approach to get snd_soc_dpcm was using container_of()
macro on snd_pcm_hw_params.
The structures have been modified over time and snd_soc_dpcm does not have
snd_pcm_hw_params as a reference but as a copy. This causes the current
driver to crash when used.
This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime
holds 2 dpcm instances (one for playback and one for capture). 2 codecs
on the SSP are dmic (capture) and speakers (playback). Based on the
stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime.
Tested for all use cases of the driver.
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
Tested-by: Lukasz Majczak <lma@semihalf.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/1595432147-11166-1-git-send-email-harshapriya.n@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The series re-uses mt8183-mt6358-ts3a227-max98357.c to support machine driver
with max98357b.
The 1st patch enables left justified format from mt8183 audio platform.
The 2nd patch adds document for the new proposed compatible string for
max98357b.
The 3rd patch supports machine driver with max98357b and uses left justified
format for it.
Tzung-Bi Shih (3):
ASoC: mediatek: mt8183: support left justified format for I2S
ASoC: dt-bindings: mt8183: add compatible string for using max98357b
ASoC: mediatek: mt8183: support machine driver with max98357b
.../sound/mt8183-mt6358-ts3a227-max98357.txt | 1 +
sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 59 ++++++++++++++++---
.../mt8183/mt8183-mt6358-ts3a227-max98357.c | 22 ++++++-
3 files changed, 73 insertions(+), 9 deletions(-)
--
2.28.0.rc0.105.gf9edc3c819-goog
Daniel Baluta <daniel.baluta@nxp.com>:
From: Daniel Baluta <daniel.baluta@nxp.com>
This patchseries contains a couple of SOF IMX fixes
found during our first IMX SOF release.
Daniel Baluta (7):
ASoC: SOF: define INFO_ flags in dsp_ops for imx8
ASoC: SOF: imx: Use ARRAY_SIZE instead of hardcoded value
ASoC: SOF: imx8: Fix ESAI DAI driver name for i.MX8/iMX8X
ASoC: SOF: imx8m: Fix SAI DAI driver for i.MX8M
ASoC: SOF: imx8: Add SAI dai driver for i.MX/i.MX8X
ASoC: SOF: topology: Update SAI config bclk/fsync rate
ASoC: SOF: pcm: Update rate/channels for SAI/ESAI DAIs
sound/soc/sof/imx/imx8.c | 24 +++++++++++++++++++++---
sound/soc/sof/imx/imx8m.c | 4 ++--
sound/soc/sof/pcm.c | 8 ++++++++
sound/soc/sof/topology.c | 2 ++
4 files changed, 33 insertions(+), 5 deletions(-)
--
2.17.1
Commit 5bd70440cb ("ASoC: soc-dai: revert all changes to DAI
startup/shutdown sequence"), introduced a slight change of semantics
to DAI startup/shutdown. If startup() returns an error, shutdown()
is now called for the DAI.
This causes a deadlock in hdac_hda which issues a call to
snd_hda_codec_pcm_put() in case open fails. Upon error, soc_pcm_open()
will call shutdown(), and pcm_put() ends up getting called twice. Result
is a deadlock on pcm->open_mutex, as snd_device_free() gets called from
within snd_pcm_open(). Typical task backtrace looks like this:
[ 334.244627] snd_pcm_dev_disconnect+0x49/0x340 [snd_pcm]
[ 334.244634] __snd_device_disconnect.part.0+0x2c/0x50 [snd]
[ 334.244640] __snd_device_free+0x7f/0xc0 [snd]
[ 334.244650] snd_hda_codec_pcm_put+0x87/0x120 [snd_hda_codec]
[ 334.244660] soc_pcm_open+0x6a0/0xbe0 [snd_soc_core]
[ 334.244676] ? dpcm_add_paths.isra.0+0x491/0x590 [snd_soc_core]
[ 334.244679] ? kfree+0x9a/0x230
[ 334.244686] dpcm_be_dai_startup+0x255/0x300 [snd_soc_core]
[ 334.244695] dpcm_fe_dai_open+0x20e/0xf30 [snd_soc_core]
[ 334.244701] ? snd_pcm_hw_rule_muldivk+0x110/0x110 [snd_pcm]
[ 334.244709] ? dpcm_be_dai_startup+0x300/0x300 [snd_soc_core]
[ 334.244714] ? snd_pcm_attach_substream+0x3c4/0x540 [snd_pcm]
[ 334.244719] snd_pcm_open_substream+0x69a/0xb60 [snd_pcm]
[ 334.244729] ? snd_pcm_release_substream+0x30/0x30 [snd_pcm]
[ 334.244732] ? __mutex_lock_slowpath+0x10/0x10
[ 334.244736] snd_pcm_open+0x1b3/0x3c0 [snd_pcm]
Fixes: 5bd70440cb ("ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2159
Link: https://lore.kernel.org/r/20200717101950.3885187-3-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The hdac_hda remove implementation fails to free the hda codec
resources, leading to memleaks at module unload. This gap has been there
from the start, commit 6bae5ea949 ("ASoC: hdac_hda: add asoc
extension for legacy HDA codec drivers").
Instead of duplicating the cleanup logic, use the common
snd_hda_codec_cleanup_for_unbind() to free the resources. Remove
existing code in hdac_hda to cleanup "codec.jackpoll_work" and call to
snd_hdac_regmap_exit(), as these are already done in
snd_hda_codec_cleanup_for_unbind().
The cleanup is done in ASoC component remove() callback and not in the
HDAC bus hdev_detach(). This is done to ensure the codec specific
cleanup routines are run before the parent card is freed.
Fixes: 6bae5ea949 ("ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2195
Link: https://lore.kernel.org/r/20200717101950.3885187-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add error handling for patch_ops in hdac_hda_codec_probe().
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200717101950.3885187-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Acked-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200719153822.59788-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Supports machine driver with max98357b
("mt8183-mt6358-ts3a227-max98357b").
The key difference from max98357a: max98357b needs to use left
justified format.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200720012559.906088-4-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
MT8183 audio platform supports EIAJ and I2S formats. The code fixed to
use I2S format in the past.
Supports EIAJ mode via set_fmt ops and preserves to use I2S format as
the default format intentionally.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200720012559.906088-2-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Starting in commit cbc7a6b5a8 ("ASoC: soc-card: add
snd_soc_card_add_dai_link()"), error value from ASoc add_dai_link() is
no longer ignored.
The generic HDA machine driver relied on the old semantics to disable
i915 HDMI/DP audio codec at runtime. If no display codec was present,
add_dai_link() returned an error, but this was ignored and rest of the
card was successfully probed.
Fix the problem by changing the machine driver add_dai_link() to not
return an error in this case.
Fixes: cbc7a6b5a8 ("ASoC: soc-card: add snd_soc_card_add_dai_link()")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2261
Link: https://lore.kernel.org/r/20200714132804.3638221-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixup BE DAI links rate/channels parameters to match any values
from topology.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-8-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
These parameters are read from topology file and sent to DSP.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-7-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With SOF we support 1 ESAI interface and 1 SAI interface.
This patch adds SAI1 interface support existing on i.MX8/i.MX8X
boards.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-6-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This must match DAI name from topology. Also, sai-port
is too generic. Physical DAI port on i.MX8MP is labeled SAI3.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-5-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This must match DAI name from topology. Also, esai-port is too generic
as they are 2 ESAIs on i.MX8/i.MX8X boards.
SOF integration only uses ESAI0 for now.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-4-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With this change we no longer need to update num_drv when adding
new DAI driver.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-3-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the past, the INFO_ flags such as PAUSE/NO_PERIOD_WAKEUP were
defined in the SOF PCM core, but that was changed since
commit 27e322fabd ("ASoC: SOF: define INFO_ flags in dsp_ops")
Now these flags must be set in DSP ops.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-2-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As expected, this requires the same quirk as the SSL2+ in order for the
clock to sync. This was suggested by, and tested on an SSL2, by Dmitry.
Suggested-by: Dmitry <dpavlushko@gmail.com>
Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200621075005.52mjjfc6dtdjnr3h@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Overview
========
Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio
Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio
pre-processing and post-processing. It also includes a programmable full
crossbar for routing audio data across these accelerators.
This series exposes some of these below mentioned HW devices as ASoC
components for Tegra platforms from Tegra210 onwards.
* ADMAIF : The interface between ADMA and AHUB
* XBAR : Crossbar for routing audio samples across various modules
* I2S : Inter-IC Sound Controller
* DMIC : Digital Microphone
* DSPK : Digital Speaker
Following is the summary of current series.
* Add YAML DT binding documentation for above mentioned modules.
* Helper function for ACIF programming is exposed for Tegra210 and later.
* Add ASoC driver components for each of the above modules.
* Build ACONNECT and ADMA drivers which are essential to realize audio
use case.
* Add DT entries for above components for Tegra210, Tegra186 and
Tegra194.
As per the suggestion in [0] audio graph based sound card support
is pushed in a separate series.
[0] https://lkml.org/lkml/2020/6/27/4
Changelog
=========
v4 -> v5
--------
* Common changes
- simple-card driver changes are dropped. Changes are migrated to audio
graph card and are moved to a separate series as suggested.
- '#sound-dai-cells' property is not needed for planned audio graph card
Hence dropped from documentation and related DT binding of component
drivers.
- CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to
represent DAI links using audio graph card. Similary DAIs are added in
AHUB driver to describe endpoints in audio crossbar. Routing is updated
to reflect the same in drivers.
v3 -> v4
--------
* [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210"
- Removed multiple examples and retained one example per doc
- Fixed as per inputs on the previous series
- Tested bindings with 'make dt_binding_check/dtbs_check'
* [2/23] "ASoC: tegra: Add support for CIF programming"
- No change
* Common changes (for patch [3/10] to [7/10])
- Mixer control overrides, for PCM parameters (rate, channel, bits),
in each driver are dropped.
- Updated routing as per DPCM usage
- Minor changes related to formatting
* New changes (patch [8/23] to [18/23] and patch [23/23])
- Based on discussions in following threads DPCM is used for Tegra Audio.
https://lkml.org/lkml/2020/2/20/91https://lkml.org/lkml/2020/4/30/519
- The simple-card driver is used for Tegra Audio and accordingly
some enhancements are made in simple-card and core drivers.
- Patch [8/23] to [18/23] are related to simple-card and core changes.
- Patch [23/23] adds sound card support to realize complete audio path.
This is based on simple-card driver with proposed enhancements.
- Re-ordered patches depending on above
v2 -> v3
--------
* [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB
- Updated licence
- Removed redundancy w.r.t items/const/enum
- Added constraints wherever needed with "pattern" property
* [2/10] "ASoC: tegra: add support for CIF programming"
- Removed tegra_cif.c
- Instead added inline helper function in tegra_cif.h
* common changes (for patch [3/10] to [7/10])
- Replace LATE system calls with Normal sleep
- Remove explicit RPM suspend in driver remove() call
- Use devm_kzalloc() instead of devm_kcalloc() for single element
- Replace 'ret' with 'err' for better reading
- Consistent error printing style across drivers
- Minor formating fixes
* [8/10] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later"
(New patch)
- Enables ACONNECT and AHUB components. With this AHUB and components are
registered with ASoC core.
v1 -> v2
--------
* [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB"
- no changes
* [2/9] "ASoC: tegra: add support for CIF programming"
- removed CIF programming changes for legacy chips.
- this patch now exposes helper function for CIF programming,
which can be used on Tegra210 later.
- later tegra_cif.c can be extended for legacy chips as well.
- updated commit message accordingly
* [3/9] "ASoC: tegra: add Tegra210 based DMIC driver"
- removed unnecessary initialization of 'ret' in probe()
* [4/9] "ASoC: tegra: add Tegra210 based I2S driver"
- removed unnecessary initialization of 'ret' in probe()
- fixed indentation
- added consistent bracing for if-else clauses
- updated 'rx_fifo_th' type to 'unsigned int'
- used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h
* [5/9] "ASoC: tegra: add Tegra210 based AHUB driver"
- used of_device_get_match_data() to get 'soc_data' and removed
explicit of_match_device()
- used devm_platform_ioremap_resource() and removed explicit
platform_get_resource()
- fixed indentation for devm_snd_soc_register_component()
- updated commit message
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [6/9] "ASoC: tegra: add Tegra186 based DSPK driver"
- removed unnecessary initialization of 'ret' in probe()
- updated 'max_th' to 'unsigned int'
- shortened lengthy macro names to avoid wrapping in
tegra186_dspk_wr_reg() and to be consistent
* [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver"
- used of_device_get_match_data() and removed explicit of_match_device()
- used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [8/9] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* common changes for patch [3/9] to [7/9]
- sorted headers in alphabetical order
- moved MODULE_DEVICE_TABLE() right below *_of_match table
- removed macro DRV_NAME
- removed explicit 'owner' field from platform_driver structure
- added 'const' to snd_soc_dai_ops structure
Sameer Pujar (11):
ASoC: dt-bindings: tegra: Add DT bindings for Tegra210
ASoC: tegra: Add support for CIF programming
ASoC: tegra: Add Tegra210 based DMIC driver
ASoC: tegra: Add Tegra210 based I2S driver
ASoC: tegra: Add Tegra210 based AHUB driver
ASoC: tegra: Add Tegra186 based DSPK driver
ASoC: tegra: Add Tegra210 based ADMAIF driver
arm64: defconfig: Build AHUB component drivers
arm64: defconfig: Build ADMA and ACONNECT driver
arm64: tegra: Enable ACONNECT, ADMA and AGIC on Jetson Nano
arm64: tegra: Add DT binding for AHUB components
.../bindings/sound/nvidia,tegra186-dspk.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-admaif.yaml | 111 +++
.../bindings/sound/nvidia,tegra210-ahub.yaml | 136 ++++
.../bindings/sound/nvidia,tegra210-dmic.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-i2s.yaml | 101 +++
arch/arm64/boot/dts/nvidia/tegra186.dtsi | 217 +++++-
arch/arm64/boot/dts/nvidia/tegra194.dtsi | 225 +++++-
arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 12 +
arch/arm64/boot/dts/nvidia/tegra210.dtsi | 140 ++++
arch/arm64/configs/defconfig | 8 +
sound/soc/tegra/Kconfig | 56 ++
sound/soc/tegra/Makefile | 10 +
sound/soc/tegra/tegra186_dspk.c | 442 +++++++++++
sound/soc/tegra/tegra186_dspk.h | 70 ++
sound/soc/tegra/tegra210_admaif.c | 800 ++++++++++++++++++++
sound/soc/tegra/tegra210_admaif.h | 162 ++++
sound/soc/tegra/tegra210_ahub.c | 676 +++++++++++++++++
sound/soc/tegra/tegra210_ahub.h | 127 ++++
sound/soc/tegra/tegra210_dmic.c | 455 ++++++++++++
sound/soc/tegra/tegra210_dmic.h | 82 +++
sound/soc/tegra/tegra210_i2s.c | 812 +++++++++++++++++++++
sound/soc/tegra/tegra210_i2s.h | 126 ++++
sound/soc/tegra/tegra_cif.h | 65 ++
sound/soc/tegra/tegra_pcm.c | 235 +++++-
sound/soc/tegra/tegra_pcm.h | 21 +-
25 files changed, 5251 insertions(+), 4 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml
create mode 100644 sound/soc/tegra/tegra186_dspk.c
create mode 100644 sound/soc/tegra/tegra186_dspk.h
create mode 100644 sound/soc/tegra/tegra210_admaif.c
create mode 100644 sound/soc/tegra/tegra210_admaif.h
create mode 100644 sound/soc/tegra/tegra210_ahub.c
create mode 100644 sound/soc/tegra/tegra210_ahub.h
create mode 100644 sound/soc/tegra/tegra210_dmic.c
create mode 100644 sound/soc/tegra/tegra210_dmic.h
create mode 100644 sound/soc/tegra/tegra210_i2s.c
create mode 100644 sound/soc/tegra/tegra210_i2s.h
create mode 100644 sound/soc/tegra/tegra_cif.h
--
2.7.4
ADMAIF is the interface between ADMA and AHUB. Each ADMA channel that
sends/receives data to/from AHUB must intreface through an ADMAIF channel.
ADMA channel sending data to AHUB pairs with an ADMAIF Tx channel and
similarly ADMA channel receiving data from AHUB pairs with an ADMAIF Rx
channel. Buffer size is configurable for each ADMAIF channel, but currently
SW uses default values.
This patch registers ADMAIF driver with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes ADMAIF interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The ADMAIF device can be enabled in the DT via
"nvidia,tegra210-admaif" compatible binding.
Tegra PCM driver is updated to expose required PCM interfaces and
snd_pcm_ops callbacks.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-8-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the reset property name when allocating the GPIO descriptor.
The gpiod_get_optional appends either the -gpio or -gpios suffix to the
name.
Fixes: 1a476abc72 ("tas2770: add tas2770 smart PA kernel driver")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200720181202.31000-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Partially reverts commit 128f825aea ("ASoC: max98357a: move control
of SD_MODE to DAPM").
In order to have mute control of max98357 from machine drivers, commit
128f825aea ("ASoC: max98357a: move control of SD_MODE to DAPM")
moves the control of SD_MODE from DAI ops to DAPM events. However, pop
noise has been observed on rk3399-gru-kevin boards due to this commit.
The commit 128f825aea caused sequence of DAI clocks and SD_MODE
changed on rk3399-gru-kevin boards.
With the commit 128f825aea:
- SD_MODE will be set to 1 before DAI clocks start.
- SD_MODE will be set to 0 after DAI clocks stop.
As a result, pop noise.
Moves the control of SD_MODE back to DAI ops. In the meantime, uses an
additional flag in DAPM event to provide chance of mute control for
machine drivers.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Tested-By: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Link: https://lore.kernel.org/r/20200721114232.2812254-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This PR became fairly large, containing mostly the collection of
ASoC fixes that slipped from the previous request, so I sent now
a bit earlier than usual. But all changes look small and mostly
device-specific, hence nothing to worry too much.
Majority of changes are for x86 based platforms and their CODEC
drivers, in order to address some issues hit by their recent tests
and fuzzing. The rest are other ASoC device-specific fixes (imx,
qcom, wm8974, amd, rockchip) as well as a trivial fix for a kernel
WARNING hit by syzkaller.
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Merge tag 'sound-5.8-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into master
Pull sound fixes from Takashi Iwai:
"This became fairly large, containing mostly the collection of ASoC
fixes that slipped from the previous request, so I sent now a bit
earlier than usual. But all changes look small and mostly
device-specific, hence nothing to worry too much.
Majority of changes are for x86 based platforms and their CODEC
drivers, in order to address some issues hit by their recent tests and
fuzzing. The rest are other ASoC device-specific fixes (imx, qcom,
wm8974, amd, rockchip) as well as a trivial fix for a kernel WARNING
hit by syzkaller"
* tag 'sound-5.8-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (28 commits)
ALSA: hda/realtek: Fixed ALC298 sound bug by adding quirk for Samsung Notebook Pen S
ALSA: info: Drop WARN_ON() from buffer NULL sanity check
ASoC: rt5682: Report the button event in the headset type only
ASoC: Intel: bytcht_es8316: Add missed put_device()
ASoC: rt5682: Enable Vref2 under using PLL2
ASoC: rt286: fix unexpected interrupt happens
ASoC: wm8974: remove unsupported clock mode
ASoC: wm8974: fix Boost Mixer Aux Switch
ASoC: SOF: core: fix null-ptr-deref bug during device removal
ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend
ASoC: codecs: max98373: Removed superfluous volume control from chip default
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: fix kernel oops on route addition error
ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M
ASoC: Intel: bdw-rt5677: fix non BE conversion
ASoC: soc-dai: set dai_link dpcm_ flags with a helper
MAINTAINERS: Add Shengjiu to reviewer list of sound/soc/fsl
ASoC: core: Remove only the registered component in devm functions
MAINTAINERS: Change Maintainer for some at91 drivers
ASoC: dt-bindings: simple-card: Fix 'make dt_binding_check' warnings
...
In commit d696a61413 ("ASoC: rt1015: Add condition to prevent SoC
providing bclk in ratio of 50 times of sample rate."), PLL input at 50fs
is no longer supported, the new recommended settings at 48Khz rate are:
PLL input SSP bclk
------------------------
64fs 3.073Mhz
100fs 4.8Mhz
(bclk update is reflected in topoplogy.)
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The mc_private->hdmi_pcm_list is populated by elements loaded during
DSP topology load. Valid topologies for this machine driver will always
have PCM nodes for HDMI, but driver should fail gracefully even in the case
this is not true. Add a sanity check to sof_sdw_hdmi_card_late_probe()
for this case. Without the fix, a null pcm handle gets dereferenced.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Extend the generic SOF Soundwire machine driver to support systems where
iDisp HDMI/DP audio codec is disabled for some reason (i915 driver
disabled, HDMI/DP implemented with a discrete GPU, etc). Switch codecs
to SoC dummy in the affected DAI links. This allows to reuse existing
topologies for this case.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The rt711 jack detection properties are set from the machine drivers
during the card probe, as done in other ASoC examples.
KASAN reports a use-after-free error when unbinding drivers due to a
confusing sequence between the ACPI core, the device core and the
SoundWire device cleanups.
Rather than fixing this sequence, follow the recommendation to have
the same caller add and remove properties, add an explicit
device_remove_properties() in the card .remove() callback.
In future patches the use of device_add/remove_properties will be
replaced by a direct handling of a swnode, but the sequence will
remain the same.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We can get codec name from dai link.
Suggested-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Overview
========
Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio
Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio
pre-processing and post-processing. It also includes a programmable full
crossbar for routing audio data across these accelerators.
This series exposes some of these below mentioned HW devices as ASoC
components for Tegra platforms from Tegra210 onwards.
* ADMAIF : The interface between ADMA and AHUB
* XBAR : Crossbar for routing audio samples across various modules
* I2S : Inter-IC Sound Controller
* DMIC : Digital Microphone
* DSPK : Digital Speaker
Following is the summary of current series.
* Add YAML DT binding documentation for above mentioned modules.
* Helper function for ACIF programming is exposed for Tegra210 and later.
* Add ASoC driver components for each of the above modules.
* Build ACONNECT and ADMA drivers which are essential to realize audio
use case.
* Add DT entries for above components for Tegra210, Tegra186 and
Tegra194.
As per the suggestion in [0] audio graph based sound card support
is pushed in a separate series.
[0] https://lkml.org/lkml/2020/6/27/4
Changelog
=========
v4 -> v5
--------
* Common changes
- simple-card driver changes are dropped. Changes are migrated to audio
graph card and are moved to a separate series as suggested.
- '#sound-dai-cells' property is not needed for planned audio graph card
Hence dropped from documentation and related DT binding of component
drivers.
- CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to
represent DAI links using audio graph card. Similary DAIs are added in
AHUB driver to describe endpoints in audio crossbar. Routing is updated
to reflect the same in drivers.
v3 -> v4
--------
* [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210"
- Removed multiple examples and retained one example per doc
- Fixed as per inputs on the previous series
- Tested bindings with 'make dt_binding_check/dtbs_check'
* [2/23] "ASoC: tegra: Add support for CIF programming"
- No change
* Common changes (for patch [3/10] to [7/10])
- Mixer control overrides, for PCM parameters (rate, channel, bits),
in each driver are dropped.
- Updated routing as per DPCM usage
- Minor changes related to formatting
* New changes (patch [8/23] to [18/23] and patch [23/23])
- Based on discussions in following threads DPCM is used for Tegra Audio.
https://lkml.org/lkml/2020/2/20/91https://lkml.org/lkml/2020/4/30/519
- The simple-card driver is used for Tegra Audio and accordingly
some enhancements are made in simple-card and core drivers.
- Patch [8/23] to [18/23] are related to simple-card and core changes.
- Patch [23/23] adds sound card support to realize complete audio path.
This is based on simple-card driver with proposed enhancements.
- Re-ordered patches depending on above
v2 -> v3
--------
* [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB
- Updated licence
- Removed redundancy w.r.t items/const/enum
- Added constraints wherever needed with "pattern" property
* [2/10] "ASoC: tegra: add support for CIF programming"
- Removed tegra_cif.c
- Instead added inline helper function in tegra_cif.h
* common changes (for patch [3/10] to [7/10])
- Replace LATE system calls with Normal sleep
- Remove explicit RPM suspend in driver remove() call
- Use devm_kzalloc() instead of devm_kcalloc() for single element
- Replace 'ret' with 'err' for better reading
- Consistent error printing style across drivers
- Minor formating fixes
* [8/10] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later"
(New patch)
- Enables ACONNECT and AHUB components. With this AHUB and components are
registered with ASoC core.
v1 -> v2
--------
* [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB"
- no changes
* [2/9] "ASoC: tegra: add support for CIF programming"
- removed CIF programming changes for legacy chips.
- this patch now exposes helper function for CIF programming,
which can be used on Tegra210 later.
- later tegra_cif.c can be extended for legacy chips as well.
- updated commit message accordingly
* [3/9] "ASoC: tegra: add Tegra210 based DMIC driver"
- removed unnecessary initialization of 'ret' in probe()
* [4/9] "ASoC: tegra: add Tegra210 based I2S driver"
- removed unnecessary initialization of 'ret' in probe()
- fixed indentation
- added consistent bracing for if-else clauses
- updated 'rx_fifo_th' type to 'unsigned int'
- used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h
* [5/9] "ASoC: tegra: add Tegra210 based AHUB driver"
- used of_device_get_match_data() to get 'soc_data' and removed
explicit of_match_device()
- used devm_platform_ioremap_resource() and removed explicit
platform_get_resource()
- fixed indentation for devm_snd_soc_register_component()
- updated commit message
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [6/9] "ASoC: tegra: add Tegra186 based DSPK driver"
- removed unnecessary initialization of 'ret' in probe()
- updated 'max_th' to 'unsigned int'
- shortened lengthy macro names to avoid wrapping in
tegra186_dspk_wr_reg() and to be consistent
* [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver"
- used of_device_get_match_data() and removed explicit of_match_device()
- used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [8/9] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* common changes for patch [3/9] to [7/9]
- sorted headers in alphabetical order
- moved MODULE_DEVICE_TABLE() right below *_of_match table
- removed macro DRV_NAME
- removed explicit 'owner' field from platform_driver structure
- added 'const' to snd_soc_dai_ops structure
Sameer Pujar (11):
ASoC: dt-bindings: tegra: Add DT bindings for Tegra210
ASoC: tegra: Add support for CIF programming
ASoC: tegra: Add Tegra210 based DMIC driver
ASoC: tegra: Add Tegra210 based I2S driver
ASoC: tegra: Add Tegra210 based AHUB driver
ASoC: tegra: Add Tegra186 based DSPK driver
ASoC: tegra: Add Tegra210 based ADMAIF driver
arm64: defconfig: Build AHUB component drivers
arm64: defconfig: Build ADMA and ACONNECT driver
arm64: tegra: Enable ACONNECT, ADMA and AGIC on Jetson Nano
arm64: tegra: Add DT binding for AHUB components
.../bindings/sound/nvidia,tegra186-dspk.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-admaif.yaml | 111 +++
.../bindings/sound/nvidia,tegra210-ahub.yaml | 136 ++++
.../bindings/sound/nvidia,tegra210-dmic.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-i2s.yaml | 101 +++
arch/arm64/boot/dts/nvidia/tegra186.dtsi | 217 +++++-
arch/arm64/boot/dts/nvidia/tegra194.dtsi | 225 +++++-
arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 12 +
arch/arm64/boot/dts/nvidia/tegra210.dtsi | 140 ++++
arch/arm64/configs/defconfig | 8 +
sound/soc/tegra/Kconfig | 56 ++
sound/soc/tegra/Makefile | 10 +
sound/soc/tegra/tegra186_dspk.c | 442 +++++++++++
sound/soc/tegra/tegra186_dspk.h | 70 ++
sound/soc/tegra/tegra210_admaif.c | 800 ++++++++++++++++++++
sound/soc/tegra/tegra210_admaif.h | 162 ++++
sound/soc/tegra/tegra210_ahub.c | 676 +++++++++++++++++
sound/soc/tegra/tegra210_ahub.h | 127 ++++
sound/soc/tegra/tegra210_dmic.c | 455 ++++++++++++
sound/soc/tegra/tegra210_dmic.h | 82 +++
sound/soc/tegra/tegra210_i2s.c | 812 +++++++++++++++++++++
sound/soc/tegra/tegra210_i2s.h | 126 ++++
sound/soc/tegra/tegra_cif.h | 65 ++
sound/soc/tegra/tegra_pcm.c | 235 +++++-
sound/soc/tegra/tegra_pcm.h | 21 +-
25 files changed, 5251 insertions(+), 4 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml
create mode 100644 sound/soc/tegra/tegra186_dspk.c
create mode 100644 sound/soc/tegra/tegra186_dspk.h
create mode 100644 sound/soc/tegra/tegra210_admaif.c
create mode 100644 sound/soc/tegra/tegra210_admaif.h
create mode 100644 sound/soc/tegra/tegra210_ahub.c
create mode 100644 sound/soc/tegra/tegra210_ahub.h
create mode 100644 sound/soc/tegra/tegra210_dmic.c
create mode 100644 sound/soc/tegra/tegra210_dmic.h
create mode 100644 sound/soc/tegra/tegra210_i2s.c
create mode 100644 sound/soc/tegra/tegra210_i2s.h
create mode 100644 sound/soc/tegra/tegra_cif.h
--
2.7.4
The Digital Speaker Controller (DSPK) converts the multi-bit Pulse Code
Modulation (PCM) audio input to oversampled 1-bit Pulse Density Modulation
(PDM) output. From the signal flow perpsective, the DSPK can be viewed as
a PDM transmitter that up-samples the input to the desired sampling rate
by interpolation then converts the oversampled PCM input to the desired
1-bit output via Delta Sigma Modulation (DSM).
This patch registers DSPK component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes DSPK interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The DSPK devices can be enabled in the DT via
"nvidia,tegra186-dspk" compatible binding. This driver can be used
on Tegra194 chip as well.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-7-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Audio Hub (AHUB) comprises a collection of hardware accelerators for
audio pre/post-processing and a programmable full crossbar (XBAR) for
routing audio data across these accelerators in time and in parallel.
AHUB supports multiple interfaces to I2S, DSPK, DMIC etc., XBAR is a
switch used to configure or modify audio routing between HW accelerators
present inside AHUB.
This patch registers AHUB component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes AHUB interfaces, which can be used to connect different
components in the ASoC layer. Currently the driver takes care of XBAR
programming to allow audio data flow through various clients of the AHUB.
Makefile and Kconfig support is added to allow to build the driver. The
AHUB component can be enabled in the DT via below compatible bindings.
- "nvidia,tegra210-ahub" for Tegra210
- "nvidia,tegra186-ahub" for Tegra186 and Tegra194
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-6-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Inter-IC Sound (I2S) controller implements full-duplex, bi-directional
and single direction point to point serial interface. It can interface
with I2S compatible devices. Tegra I2S controller can operate as both
master and slave.
This patch registers I2S controller with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes I2S interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The I2S devices can be enabled in the DT via
"nvidia,tegra210-i2s" compatible binding.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-5-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Digital MIC (DMIC) Controller is used to interface with Pulse Density
Modulation (PDM) input devices. The DMIC controller implements a converter
to convert PDM signals to Pulse Code Modulation (PCM) signals. From signal
flow perspective, the DMIC can be viewed as a PDM receiver.
This patch registers DMIC component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes DMIC interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The DMIC devices can be enabled in the DT via
"nvidia,tegra210-dmic" compatible string. This driver can be used for
Tegra186 and Tegra194 chips as well.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-4-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Audio Client Interface (CIF) is a proprietary interface employed to route
audio samples through Audio Hub (AHUB) components by inter connecting the
various modules.
This patch exports an inline function tegra_set_cif() which can be used,
for now, to program CIF on Tegra210 and later Tegra generations. Later it
can be extended to include helpers for legacy chips as well.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Reviewed-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/1595134890-16470-3-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This configuration is for EHL with the RT5660 codec. RT5660
should use "10EC5660" ID instead of "INTC1027".
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
All drivers are now using .mute_stream.
Let's remove .digital_mute.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87h7u72dqz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Follow the recent inclusive terminology guidelines and replace the
word "slave" in vmaster API. I chose the word "follower" at this time
since it seems fitting for the purpose.
Note that the word "master" is kept in API, since it refers rather to
audio master volume control.
Also, while we're at it, a typo in comments is corrected, too.
Link: https://lore.kernel.org/r/20200717154517.27599-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://lore.kernel.org/r/20200719151705.59624-1-grandmaster@al2klimov.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed no headphone sound bug on laptop Samsung Notebook Pen S
(950SBE-951SBE), by using existing patch in Linus' tree, commit
14425f1f52 (ALSA: hda/realtek: Add quirk for Samsung Notebook).
This laptop uses the same ALC298 but different subsystem id 0x144dc812.
I added SND_PCI_QUIRK at sound/pci/hda/patch_realtek.c
Signed-off-by: Joonho Wohn <doomsheart@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAHcbMh291aWDKiWSZoxXB4-Eru6OYRwGA4AVEdCZeYmVLo5ZxQ@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them. For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.
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Merge tag 'asoc-fix-v5.8-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.8
An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them. For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.
No surprise here, just a few device-specific small fixes: two fixes
for USB LINE6 and one for USB-audio drivers wrt syzkaller fuzzer
issues, while the rest are all HD-audio Realtek quirks.
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Merge tag 'sound-5.8-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into master
Pull sound fixes from Takashi Iwai:
"No surprise here, just a few device-specific small fixes: two fixes
for USB LINE6 and one for USB-audio drivers wrt syzkaller fuzzer
issues, while the rest are all HD-audio Realtek quirks"
* tag 'sound-5.8-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - fixup for yet another Intel reference board
ALSA: hda/realtek - Enable Speaker for ASUS UX563
ALSA: hda/realtek - Enable Speaker for ASUS UX533 and UX534
ALSA: hda/realtek: Enable headset mic of Acer TravelMate B311R-31 with ALC256
ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289
ALSA: hda/realtek - change to suitable link model for ASUS platform
ALSA: usb-audio: Fix race against the error recovery URB submission
ALSA: line6: Sync the pending work cancel at disconnection
ALSA: line6: Perform sanity check for each URB creation
Some settings should set to default value after the calibration.
This patch also disables the 25MHz and 1MHz clock power when the jack unplugged.
The JD is triggered by JDH, therefore this patch removes JDL setting.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200717070228.28660-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the function q6adm_open(), q6adm_alloc_copp() doesn't return
NULL. Thus use IS_ERR() to validate the returned value instead
of IS_ERR_OR_NULL(). And delete the extra line.
Signed-off-by: Zhang Shengju <zhangshengju@cmss.chinamobile.com>
Signed-off-by: Tang Bin <tangbin@cmss.chinamobile.com>
Link: https://lore.kernel.org/r/20200714112744.20560-1-tangbin@cmss.chinamobile.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The pin status of the widget was connected after the sound card registered.
The rt5682_headset_detect function will use the pin status of these two widgets
to decide the certain register setting on/off.
Therefore this patch disables the pin of these two widgets in the codec probe.
This patch could avoid the misjudgment.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200717070256.28712-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is used for both CPU and Codec.
For example, soc_pcm_prepare() / soc_pcm_hw_free() are caring
both CPU and Codec.
But soc_resume_deferred() / snd_soc_suspend() are not.
This patch cares it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87ft9r2dqr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
-
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Link: https://lore.kernel.org/r/87eepb2dnq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
axg_card_add_tdm_loopback() misses to call kfree() in an error path. We
can use devm_kasprintf() to fix the issue, also improve maintainability.
So use it instead.
Fixes: c84836d7f6 ("ASoC: meson: axg-card: use modern dai_link style")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200717082242.130627-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_info_get_line() has a sanity check of NULL buffer -- both buffer
itself being NULL and buffer->buffer being NULL. Basically both
checks are valid and necessary, but the problem is that it's with
snd_BUG_ON() macro that triggers WARN_ON(). The latter condition
(NULL buffer->buffer) can be met arbitrarily by user since the buffer
is allocated at the first write, so it means that user can trigger
WARN_ON() at will.
This patch addresses it by simply moving buffer->buffer NULL check out
of snd_BUG_ON() so that spurious WARNING is no longer triggered.
Reported-by: syzbot+e42d0746c3c3699b6061@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200717084023.5928-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 3ad796cbc3 ("ALSA: pcm: Use SG-buffer only when
direct DMA is available") also the modification commit 467fd0e82b
("ALSA: pcm: Fix build error on m68k and others").
Poking the DMA internal helper is a layer violation, so we should
avoid that. Meanwhile the actual bug has been addressed by the
Kconfig fix in commit dbed452a07 ("dma-pool: decouple DMA_REMAP from
DMA_COHERENT_POOL"), so we can live without this hack.
Link: https://lore.kernel.org/r/20200717064130.22957-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hi,
this small series is preparation for a set of bugfix ASoC patches
addressing a memleak at module unload for the HDA codec wrapper.
Instead of duplicating HDA code in ASoC tree, I chose to export
more functionality from hda_codec.c so it can be (re)used in ASoC's
hdac_hda.c.
Full series:
https://github.com/thesofproject/linux/pull/2252
Takashi and Mark, feedback is welcome on how to best handle this
kind of series where I have dependent patches both in sound/pci/hda
and in ASoC. For this series, I'm sending the patches separately
and when/if first set is merged by Takashi, I'll route the ASoC
patches via our usually SOF set to Mark.
Kai Vehmanen (2):
ALSA: hda: export snd_hda_codec_cleanup_for_unbind()
ALSA: hda: fix snd_hda_codec_cleanup() documentation
include/sound/hda_codec.h | 2 ++
sound/pci/hda/hda_codec.c | 3 ++-
2 files changed, 4 insertions(+), 1 deletion(-)
--
2.27.0
Support hp and mic detection.
Add a parameter for asoc_simple_init_jack.
Shengjiu Wang (3):
ASoC: simple-card-utils: Support configure pin_name for
asoc_simple_init_jack
ASoC: bindings: fsl-asoc-card: Support hp-det-gpio and mic-det-gpio
ASoC: fsl-asoc-card: Support Headphone and Microphone Jack detection
changes in v2:
- Add more comments in third commit
- Add Acked-by Nicolin.
.../bindings/sound/fsl-asoc-card.txt | 3 +
include/sound/simple_card_utils.h | 6 +-
sound/soc/fsl/Kconfig | 1 +
sound/soc/fsl/fsl-asoc-card.c | 77 ++++++++++++++++++-
sound/soc/generic/simple-card-utils.c | 7 +-
5 files changed, 86 insertions(+), 8 deletions(-)
--
2.27.0
Add missed return for calling soc_component_ret, otherwise the return
value is wrong.
Fixes: e2329eeba4 ("ASoC: soc-component: add soc_component_err()")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/1594876028-1845-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use asoc_simple_init_jack function from simple card to implement
the Headphone and Microphone detection.
Register notifier to disable Speaker when Headphone is plugged in
and enable Speaker when Headphone is unplugged.
Register notifier to disable Digital Microphone when Analog Microphone
is plugged in and enable DMIC when Analog Microphone is unplugged.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1594822179-1849-4-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the pin_name is fixed in asoc_simple_init_jack, but some driver
may use a different pin_name. So add a new parameter in
asoc_simple_init_jack for configuring pin_name.
If this parameter is NULL, then the default pin_name is used.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1594822179-1849-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87pn95wiwa.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87r1tlwiwe.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/87sge1wiwi.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87tuyhwiwm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/87v9ixwiwr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87wo3dwiwv.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87y2ntwix0.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87zh89wix5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/871rllxxhp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/873661xxhu.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/874kqhxxhz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/875zaxxxi4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/878sftxxie.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87a709xxij.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87blkpxxip.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
For hdmi-codec, we need to update struct hdmi_codec_ops,
and all its users in the same time.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87d055xxj2.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling "direction".
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
To prepare merging mute_stream()/digital_mute(),
this patch adds .no_capture_mute support to emulate .digital_mute().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87eeplxxj7.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() will return -ENOTSUPP if driver doesn't
support mute.
In hdmi-codec case, hdmi_codec_digital_mute() will be used for it,
and each driver has .digital_mute() callback.
hdmi_codec_digital_mute() want to return -ENOTSUPP to follow it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87fta1xxjc.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To avoid duplicated code for cleanup, and match the already exported
snd_hda_codec_pcm_new(), also export snd_hda_codec_cleanup_for_unbind().
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200715174551.3730165-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Using uninitialized_var() is dangerous as it papers over real bugs[1]
(or can in the future), and suppresses unrelated compiler warnings
(e.g. "unused variable"). If the compiler thinks it is uninitialized,
either simply initialize the variable or make compiler changes.
In preparation for removing[2] the[3] macro[4], remove all remaining
needless uses with the following script:
git grep '\buninitialized_var\b' | cut -d: -f1 | sort -u | \
xargs perl -pi -e \
's/\buninitialized_var\(([^\)]+)\)/\1/g;
s:\s*/\* (GCC be quiet|to make compiler happy) \*/$::g;'
drivers/video/fbdev/riva/riva_hw.c was manually tweaked to avoid
pathological white-space.
No outstanding warnings were found building allmodconfig with GCC 9.3.0
for x86_64, i386, arm64, arm, powerpc, powerpc64le, s390x, mips, sparc64,
alpha, and m68k.
[1] https://lore.kernel.org/lkml/20200603174714.192027-1-glider@google.com/
[2] https://lore.kernel.org/lkml/CA+55aFw+Vbj0i=1TGqCR5vQkCzWJ0QxK6CernOU6eedsudAixw@mail.gmail.com/
[3] https://lore.kernel.org/lkml/CA+55aFwgbgqhbp1fkxvRKEpzyR5J8n1vKT1VZdz9knmPuXhOeg@mail.gmail.com/
[4] https://lore.kernel.org/lkml/CA+55aFz2500WfbKXAx8s67wrm9=yVJu65TpLgN_ybYNv0VEOKA@mail.gmail.com/
Reviewed-by: Leon Romanovsky <leonro@mellanox.com> # drivers/infiniband and mlx4/mlx5
Acked-by: Jason Gunthorpe <jgg@mellanox.com> # IB
Acked-by: Kalle Valo <kvalo@codeaurora.org> # wireless drivers
Reviewed-by: Chao Yu <yuchao0@huawei.com> # erofs
Signed-off-by: Kees Cook <keescook@chromium.org>
The irq work will be manipulated by resume function, and it will report
the wrong jack type while the jack type is headphone in the button event.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200716030123.27122-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error
path. Add the missed function call to fix it.
Fixes: ba49cf6f8e ("ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200714080918.148196-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that there's a function that calculates the SHA-256 digest of a
buffer in one step, use it instead of sha256_init() + sha256_update() +
sha256_final().
Also simplify the code by inlining calculate_sha256() into its caller
and switching a debug log statement to use %*phN instead of bin2hex().
Acked-by: Tzung-Bi Shih <tzungbi@google.com>
Reviewed-by: Ard Biesheuvel <ardb@kernel.org>
Cc: alsa-devel@alsa-project.org
Cc: Ard Biesheuvel <ardb@kernel.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Cc: Guenter Roeck <groeck@chromium.org>
Cc: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Eric Biggers <ebiggers@google.com>
Signed-off-by: Herbert Xu <herbert@gondor.apana.org.au>
ASUS UX563 speaker can't output.
Add quirk to link suitable model will enable it.
This model also could enable headset Mic.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/96dee3ab01a04c28a7b44061e88009dd@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS UX533 and UX534 speaker still can't output.
End User feedback speaker didn't have output.
Add this COEF value will enable it.
Fixes: 4e05110673 ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/80334402a93b48e385f8f4841b59ae09@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the recent inclusive terminology guidelines and replace the
word "blacklist" appropriately.
Only a comment fix, no functional changes.
Link: https://lore.kernel.org/r/20200714172631.25371-11-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the recent inclusive terminology guidelines and replace the
words "whitelist" appropriately.
Only comment or variable renames, no functional changes.
Link: https://lore.kernel.org/r/20200714172631.25371-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the recent inclusive terminology guidelines and replace the
words "whitelist" and "blacklist" appropriately.
Only comment or enum/variable renames, no functional changes.
Link: https://lore.kernel.org/r/20200714172631.25371-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the recent inclusive terminology guidelines and replace the
words "whitelist" and "blacklist" appropriately.
Correcting only comments, or error/module messages, no functional
changes.
Link: https://lore.kernel.org/r/20200714172631.25371-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the recent inclusive terminology guidelines and replace the
words "whitelist" and "blacklist" appropriately.
Only comment or variable renames, no functional changes.
Note that pm_blacklist module option is still kept as was, so that
users can still keep the old option.
Link: https://lore.kernel.org/r/20200714172631.25371-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the recent inclusive terminology guidelines and replace the
words "whitelist" and "blacklist" appropriately.
Only comment or variable renames, no functional changes.
Link: https://lore.kernel.org/r/20200714172631.25371-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the recent inclusive terminology guidelines and replace the
word "blacklist" appropriately.
Only correcting the error message, no functional changes.
Link: https://lore.kernel.org/r/20200714172631.25371-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the recent inclusive terminology guidelines and replace the
words "whitelist" and "blacklist" appropriately.
Only comment or variable renames, no functional changes.
Link: https://lore.kernel.org/r/20200714172631.25371-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the recent inclusive terminology guidelines and replace the
words "whitelist" and "blacklist" appropriately.
Only comment or function/variable renames, no functional changes.
Link: https://lore.kernel.org/r/20200714172631.25371-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit f34a4c9dd4 ("ALSA: hda: Enable sync-write operation as default
for all controllers") enabled sync-write for all controllers and this is
causing audio playback on the Tegra186 HDA device to fail. For now,
disable sync-write support for Tegra to fix this.
Fixes: f34a4c9dd4 ("ALSA: hda: Enable sync-write operation as default for all controllers")
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20200714160841.2293-1-jonathanh@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
siu is using discriminatory terms for function parameter.
This patch changes it to "secondary"
One note here is that it do nothing to DMA related naming
for now, because it needs framework level modification.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87d04z3qqg.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd is using discriminatory terms for function names.
This patch changes it to "secondary"
One note here is that it do nothing to DMA related naming
for now, because it needs framework level modification.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h7ub3qra.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This should be spin_unlock_irq() instead of spin_lock().
Fixes: 6c33125448 ("ALSA: echoaudio: Prevent races in calls to set_audio_format()")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20200713105324.GB251988@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add hw monitor volume control for POD HD500. The same change may
work for HD500X but I don't have it to test.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Link: https://lore.kernel.org/r/20200713152852.65832-1-anarsoul@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer TravelMate B311R-31 laptop's audio (1025:1430) with ALC256
cannot detect the headset microphone until
ALC256_FIXUP_ACER_MIC_NO_PRESENCE quirk maps the NID 0x19 as the headset
mic pin.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200713060421.62435-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for headset mic to the ASUS ROG Zephyrus
G14(GA401) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200711110557.18681-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS platform couldn't need to use Headset Mode model.
It changes to the suitable model.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/d05bcff170784ec7bb35023407148161@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB MIDI driver has an error recovery mechanism to resubmit the URB in
the delayed timer handler, and this may race with the standard start /
stop operations. Although both start and stop operations themselves
don't race with each other due to the umidi->mutex protection, but
this isn't applied to the timer handler.
For fixing this potential race, the following changes are applied:
- Since the timer handler can't use the mutex, we apply the
umidi->disc_lock protection at each input stream URB submission;
this also needs to change the GFP flag to GFP_ATOMIC
- Add a check of the URB refcount and skip if already submitted
- Move the timer cancel call at disconnection to the beginning of the
procedure; this assures the in-flight timer handler is gone properly
before killing all pending URBs
Reported-by: syzbot+0f4ecfe6a2c322c81728@syzkaller.appspotmail.com
Reported-by: syzbot+5f1d24c49c1d2c427497@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200710160656.16819-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This series is a follow up for a long time ago series
(https://patchwork.kernel.org/cover/11204303/).
The old series bound too much on the patches of DRM bridge and ASoC
machine driver. And unluckily, the dependencies
(https://lore.kernel.org/patchwork/patch/1126819/) have not applied.
Revewing the ASoC patches in the old series, I found that they could be
decoupled from the DRM bridge patches. And they are harmless as it is
an optional attribute ("hdmi-codec") in DTS.
This series arranges and rebases the harmless ASoC patches for
mt8183-mt6358-ts3a227-max98357 and mt8183-da7219-max98357.
The 1st and 4th patch add an optional DT property. The 1st patch was
acked long time ago (https://patchwork.kernel.org/patch/11204321/).
The 2nd and 5th patch add DAI link for using hdmi-codec.
The 3rd and 6th patch support the HDMI jack reporting.
Tzung-Bi Shih (6):
ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec"
ASoC: mediatek: mt8183: use hdmi-codec
ASoC: mediatek: mt8183: support HDMI jack reporting
ASoC: dt-bindings: mt8183-da7219: add a property "mediatek,hdmi-codec"
ASoC: mediatek: mt8183-da7219: use hdmi-codec
ASoC: mediatek: mt8183-da7219: support HDMI jack reporting
.../bindings/sound/mt8183-da7219-max98357.txt | 4 +++
.../sound/mt8183-mt6358-ts3a227-max98357.txt | 2 ++
sound/soc/mediatek/Kconfig | 2 ++
.../mediatek/mt8183/mt8183-da7219-max98357.c | 29 +++++++++++++++++--
.../mt8183/mt8183-mt6358-ts3a227-max98357.c | 29 +++++++++++++++++--
5 files changed, 60 insertions(+), 6 deletions(-)
--
2.27.0.383.g050319c2ae-goog
Both Lee Jones and I submitted separate series, this is the second
part of the merged result, for which no feedback was provided.
I picked Lee's patches for rt5659 and ak4458 and added the pxa and
ux500 that I didn't fix. The rest is largely identical between our
respective series, with the exception of the sunxi which I documented
and Lee removed. I don't have any specific preference and will go with
the flow on this.
Changes since v3:
Improved commit subjects from 'fix kernel-doc' as suggested by Lee
Jones. In a couple of cases I just reverted to Lee's patches when the
code was identical.
Added a couple of CC: tags from Lee's patches.
Added Arnaud Pouliquen's Acked-by tag in first patch.
Lee Jones (6):
ASoC: sunxi: sun4i-spdif: Fix misspelling of 'reg_dac_txdata' in
kernel-doc
ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header
ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and
'reg_val_TSTDR'
ASoC: codecs: rt5659: Remove many unused const variables
ASoC: codecs: tlv320aic26: Demote seemingly unintentional kerneldoc
header
ASoC: codecs: ak4458: Remove set but never checked variable 'ret'
Pierre-Louis Bossart (4):
ASoC: sti: uniperif: fix 'defined by not used' warning
ASoC: qcom: qdsp6: q6asm: Provide documentation for 'codec_profile'
ASoC: sunxi: sun4i-i2s: add missing clock and format arguments in
kernel-doc
ASoC: codecs: rt5631: fix spurious kernel-doc start and missing
arguments
sound/soc/codecs/ak4458.c | 6 +++---
sound/soc/codecs/rt5631.c | 8 +++++--
sound/soc/codecs/rt5659.c | 37 ---------------------------------
sound/soc/codecs/tlv320aic26.c | 2 +-
sound/soc/pxa/pxa-ssp.c | 2 +-
sound/soc/qcom/qdsp6/q6asm.c | 2 +-
sound/soc/sti/uniperif.h | 2 +-
sound/soc/sunxi/sun4i-i2s.c | 10 ++++++++-
sound/soc/sunxi/sun4i-spdif.c | 2 +-
sound/soc/ux500/ux500_msp_i2s.c | 8 +++----
10 files changed, 27 insertions(+), 52 deletions(-)
base-commit: 6940701c71
--
2.25.1
Clear the validity bit for TX
Add kctl for configuring TX validity bit
Shengjiu Wang (2):
ASoC: fsl_spdif: Clear the validity bit for TX
ASoC: fsl_spdif: Add kctl for configuring TX validity bit
sound/soc/fsl/fsl_spdif.c | 51 ++++++++++++++++++++++++++++++++++++---
1 file changed, 47 insertions(+), 4 deletions(-)
--
2.21.0
As Pierre-Louis Bossart pointed out, saying that the default mode for the
SSP is TDM 4 slot is not entirely accurate.
There really are 2 default modes:
The default mode for the SSP configuration is TDM 4 slot for the
cpu-dai (hard-coded in DSP firmware),
The default mode for the SSP configuration is I2S for the codec-dai
(hard-coded in the 'SSP2-Codec" .dai_fmt masks, so far unused).
This commit updates the comment in cht_codec_fixup() to properly reflect
this.
Suggested-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/20200703103840.333732-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add one kctl for configuring TX validity bit from user
space.
The type of this kctl is boolean:
on - Outgoing validity always set
off - Outgoing validity always clear
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1594112066-31297-3-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In IEC958 spec, "The validity bit is logical "0" if the
information in the main data field is reliable, and it
is logical "1" if it is not".
The default value of "ValCtrl" is zero, which means
"Outgoing Validity always set", then all the data is not
reliable, then some spdif sink device will drop the data.
So set "ValCtrl" to 1, that is to clear "Outgoing Validity"
in default.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1594112066-31297-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Recently syzkaller reported a UAF in LINE6 driver, and it's likely
because we call cancel_delayed_work() at the disconnect callback
instead of cancel_delayed_work_sync(). Let's use the correct one
instead.
Reported-by: syzbot+145012a46658ac00fc9e@syzkaller.appspotmail.com
Suggested-by: Alan Stern <stern@rowland.harvard.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hlfjr4gio.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
LINE6 drivers create stream URBs with a fixed pipe without checking
its validity, and this may lead to a kernel WARNING at the submission
when a malformed USB descriptor is passed.
For avoiding the kernel warning, perform the similar sanity checks for
each pipe type at creating a URB.
Reported-by: syzbot+c190f6858a04ea7fbc52@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hv9iv4hq8.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Looks as though the result of snd_soc_update_bits() has never been checked.
Fixes the following W=1 kernel build warning(s):
sound/soc/codecs/ak4458.c: In function ‘ak4458_set_dai_mute’:
sound/soc/codecs/ak4458.c:408:16: warning: variable ‘ret’ set but not
used [-Wunused-but-set-variable]
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Junichi Wakasugi <wakasugi.jb@om.asahi-kasei.co.jp>
Cc: Mihai Serban <mihai.serban@nxp.com>
Link: https://lore.kernel.org/r/20200709162328.259586-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This is the only use of kerneldoc in the sourcefile and no
descriptions are provided.
Fixes the following W=1 kernel build warning(s):
sound/soc/codecs/tlv320aic26.c:138: warning: Function parameter or
member 'dai' not described in 'aic26_mute'
sound/soc/codecs/tlv320aic26.c:138: warning: Function parameter or
member 'mute' not described in 'aic26_mute'
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Grant Likely <grant.likely@secretlab.ca>
Link: https://lore.kernel.org/r/20200709162328.259586-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks as though they've never been used.
Fixes the following W=1 kernel build warning(s):
In file included from sound/soc/codecs/rt5659.c:25:
In file included from sound/soc/codecs/rt5659.c:25:
sound/soc/codecs/rt5659.c:1232:2: warning: ‘rt5659_ad_monor_asrc_enum’ defined but not used [-Wunused-const-variable=]
1232 | rt5659_ad_monor_asrc_enum, RT5659_ASRC_3, RT5659_AD_MONO_R_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1231:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1231 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1228:2: warning: ‘rt5659_ad_monol_asrc_enum’ defined but not used [-Wunused-const-variable=]
1228 | rt5659_ad_monol_asrc_enum, RT5659_ASRC_3, RT5659_AD_MONO_L_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1227:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1227 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1224:2: warning: ‘rt5659_ad_sto2_asrc_enum’ defined but not used [-Wunused-const-variable=]
1224 | rt5659_ad_sto2_asrc_enum, RT5659_ASRC_3, RT5659_AD_STO2_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1223:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1223 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1220:2: warning: ‘rt5659_ad_sto1_asrc_enum’ defined but not used [-Wunused-const-variable=]
1220 | rt5659_ad_sto1_asrc_enum, RT5659_ASRC_2, RT5659_AD_STO1_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1219:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1219 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1216:2: warning: ‘rt5659_da_monor_asrc_enum’ defined but not used [-Wunused-const-variable=]
1216 | rt5659_da_monor_asrc_enum, RT5659_ASRC_2, RT5659_DA_MONO_R_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1215:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1215 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1212:2: warning: ‘rt5659_da_monol_asrc_enum’ defined but not used [-Wunused-const-variable=]
1212 | rt5659_da_monol_asrc_enum, RT5659_ASRC_2, RT5659_DA_MONO_L_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1211:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1211 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5659.c:1208:2: warning: ‘rt5659_da_sto_asrc_enum’ defined but not used [-Wunused-const-variable=]
1208 | rt5659_da_sto_asrc_enum, RT5659_ASRC_2, RT5659_DA_STO_T_SFT, 0x7,
| ^~~~~~~~~~~~~~~~~~~~~~~
include/sound/soc.h:359:24: note: in definition of macro ‘SOC_VALUE_ENUM_DOUBLE_DECL’
359 | const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, | ^~~~
sound/soc/codecs/rt5659.c:1207:8: note: in expansion of macro ‘SOC_VALUE_ENUM_SINGLE_DECL’
1207 | static SOC_VALUE_ENUM_SINGLE_DECL(
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200709162328.259586-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following W=1 kernel build warning(s):
sound/soc/codecs/rt5631.c:72: warning: Function parameter or member
'component' not described in 'rt5631_write_index'
sound/soc/codecs/rt5631.c:72: warning: Function parameter or member
'reg' not described in 'rt5631_write_index'
sound/soc/codecs/rt5631.c:72: warning: Function parameter or member
'value' not described in 'rt5631_write_index'
sound/soc/codecs/rt5631.c:82: warning: Function parameter or member
'component' not described in 'rt5631_read_index'
sound/soc/codecs/rt5631.c:82: warning: Function parameter or member
'reg' not described in 'rt5631_read_index'
sound/soc/codecs/rt5631.c:367: warning: Function parameter or member
'component' not described in 'onebit_depop_power_stage'
sound/soc/codecs/rt5631.c:405: warning: Function parameter or member
'component' not described in 'onebit_depop_mute_stage'
sound/soc/codecs/rt5631.c:443: warning: Function parameter or member
'component' not described in 'depop_seq_power_stage'
sound/soc/codecs/rt5631.c:515: warning: Function parameter or member
'component' not described in 'depop_seq_mute_stage'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200709162328.259586-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like these have been unchecked since the driver's inception in 2012.
Fixes the following W=1 kernel build warning(s):
sound/soc/ux500/ux500_msp_i2s.c: In function ‘flush_fifo_rx’:
sound/soc/ux500/ux500_msp_i2s.c:398:6: warning: variable ‘reg_val_DR’
set but not used [-Wunused-but-set-variable]
sound/soc/ux500/ux500_msp_i2s.c: In function ‘flush_fifo_tx’:
sound/soc/ux500/ux500_msp_i2s.c:415:6: warning: variable
‘reg_val_TSTDR’ set but not used [-Wunused-but-set-variable]
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: zhong jiang <zhongjiang@huawei.com>
Cc: Ola Lilja <ola.o.lilja@stericsson.com>
Cc: Roger Nilsson <roger.xr.nilsson@stericsson.com>
Cc: Sandeep Kaushik <sandeep.kaushik@st.com>
Link: https://lore.kernel.org/r/20200709162328.259586-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This is the only use of kerneldoc in the sourcefile and full
descriptions are not provided.
Fixes the following W=1 kernel build warning(s):
sound/soc/pxa/pxa-ssp.c:186: warning: Function parameter or member
'ssp' not described in 'pxa_ssp_set_scr'
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Daniel Mack <daniel@zonque.org>
Cc: Haojian Zhuang <haojian.zhuang@gmail.com>
Cc: Robert Jarzmik <robert.jarzmik@free.fr>
Link: https://lore.kernel.org/r/20200709162328.259586-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Property name descriptions need to match exactly.
Fixes the following W=1 kernel build warning(s):
sound/soc/sunxi/sun4i-spdif.c:178: warning: Function parameter or
member 'reg_dac_txdata' not described in 'sun4i_spdif_quirks'
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Maxime Ripard <mripard@kernel.org>
Cc: Chen-Yu Tsai <wens@csie.org>
Cc: Philipp Zabel <p.zabel@pengutronix.de>
Cc: Andrea Venturi <be17068@iperbole.bo.it>
Cc: Marcus Cooper <codekipper@gmail.com>
Link: https://lore.kernel.org/r/20200709162328.259586-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warnings - missing fields in description
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or
member 'bclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'num_bclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'mclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'num_mclk_dividers' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'get_bclk_parent_rate' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'get_sr' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'get_wss' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'set_chan_cfg' not described in 'sun4i_i2s_quirks'
sound/soc/sunxi/sun4i-i2s.c:160: warning: Function parameter or member
'set_fmt' not described in 'sun4i_i2s_quirks'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Maxime Ripard <mripard@kernel.org>
Cc: Chen-Yu Tsai <wens@csie.org>
Cc: Philipp Zabel <p.zabel@pengutronix.de>
Cc: Andrea Venturi <be17068@iperbole.bo.it>
Link: https://lore.kernel.org/r/20200709162328.259586-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following W=1 kernel build warning(s):
sound/soc/qcom/qdsp6/q6asm.c:924: warning: Function parameter or
member 'codec_profile' not described in 'q6asm_open_write'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200709162328.259586-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning. The table uni_tdm_hw is declared in a header included
by multiple C file. This isn't really a good practice but for now
using __maybe_unused makes the following warning go away.
sound/soc/sti/sti_uniperif.c:12:
sound/soc/sti/uniperif.h:1351:38: warning: ‘uni_tdm_hw’ defined but
not used [-Wunused-const-variable=]
1351 | static const struct snd_pcm_hardware uni_tdm_hw = {
| ^~~~~~~~~~
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Link: https://lore.kernel.org/r/20200709162328.259586-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This series tries to reuse mt8183-da7219-max98357.c for supporting machine
driver with rt1015 speaker amplifier.
The first 3 patches refactor the code for easier to change for subsequent
patches.
The 4th patch adds document for the new proposed compatible string.
The 5th patch changes the machine driver to support either "MAX98357A" or
"RT1015" codecs.
Tzung-Bi Shih (5):
ASoC: mediatek: mt8183-da7219: sort header inclusions in alphabetical
ASoC: mediatek: mt8183-da7219: remove forward declaration of
headset_init
ASoC: mediatek: mt8183-da7219: extract codec and DAI names
ASoC: mediatek: mt8183-da7219: add compatible string for using rt1015
ASoC: mediatek: mt8183-da7219: support machine driver with rt1015
.../bindings/sound/mt8183-da7219-max98357.txt | 5 +-
sound/soc/mediatek/Kconfig | 5 +-
.../mediatek/mt8183/mt8183-da7219-max98357.c | 244 ++++++++++++++----
3 files changed, 197 insertions(+), 57 deletions(-)
--
2.27.0.383.g050319c2ae-goog
Add the TX offset slot programming. There is no RX offset slot
register.
Since there is no RX offset the check for slot symmetry can be removed.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200709185129.10505-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The CPU and the codec both are represented now as components, so for
PDMIC we are registering two componenets with the same name. Since
there is no actual codec, we will merge the codec component into the
CPU one and use a dummy codec instead, for the DAI link.
As a bonus, debugfs will no longer report an error when will try to
create entries for both componenets with the same name.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20200708163359.2698696-1-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The CPU and the codec both are represented now as components, so for
CLASS-D we are registering two componenets with the same name. Since
there is no actual codec, we will merge the codec component into the
CPU one and use a dummy codec instead, for the DAI link.
As a bonus, debugfs will no longer report an error when will try to
create entries for both componenets with the same name.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20200708101249.2626560-1-codrin.ciubotariu@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning and removed unused table. In this case this a
duplicate of
static const struct of_device_id max98390_of_match[] = {
{ .compatible = "maxim,max98390", },
{}
};
MODULE_DEVICE_TABLE(of, max98390_of_match);
already used in the rest of the code.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200707190612.97799-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like 'w' has remained unchecked since the driver's inception.
Fixes the following W=1 kernel build warning(s):
sound/soc/ti/omap-mcbsp-st.c: In function ‘omap_mcbsp_st_chgain’:
sound/soc/ti/omap-mcbsp-st.c:145:6: warning: variable ‘w’ set but not used [-Wunused-but-set-variable]
Peter suggested that the whole read can be removed, so that's
been done too.
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Samuel Ortiz <samuel.ortiz@nokia.com>
Cc: linux-omap@vger.kernel.org
Link: https://lore.kernel.org/r/20200707190612.97799-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following W=1 kernel build warning(s):
In file included from include/sound/tlv.h:10,
from sound/soc/codecs/jz4770.c:19:
sound/soc/codecs/jz4770.c:306:35: warning: ‘mic_boost_tlv’ defined but not used [-Wunused-const-variable=]
306 | static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 400, 0);
| ^~~~~~~~~~~~~
include/uapi/sound/tlv.h:64:15: note: in definition of macro ‘SNDRV_CTL_TLVD_DECLARE_DB_SCALE’
64 | unsigned int name[] = { | ^~~~
sound/soc/codecs/jz4770.c:306:14: note: in expansion of macro ‘DECLARE_TLV_DB_SCALE’
306 | static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 400, 0);
| ^~~~~~~~~~~~~~~~~~~~
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Paul Cercueil <paul@crapouillou.net>
Cc: ter Huurne <maarten@treewalker.org>
Link: https://lore.kernel.org/r/20200707190612.97799-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning, the kernel-doc syntax was probably from Doxygen?
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/20200707190612.97799-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warnings - missing fields in structure
Credits to Sylwester Nawrocki for the pclk and cclk descriptions.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Link: https://lore.kernel.org/r/20200707190612.97799-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix W=1 warning
Kernel-doc is not used in one file and missing argument in the second.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Link: https://lore.kernel.org/r/20200707190612.97799-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Disable Left and Right Spk pin after boot so that sof can get
suspended.
This follows the same logic added to another machine driver with
commit 94d2d08974 ("ASoC: Intel: Boards: tgl_max98373: add dai_trigger function")
Signed-off-by: randerwang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Reflect Kconfig changes and add both SoundWire and I2C modes
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add SoundWire specific parts and extend common ones already split from
I2C.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To prepare support for SoundWire, let's first split the I2C and common
parts. No new functionality, just indents and formatting to make
checkpatch happy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200708203215.231776-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Supports machine driver with rt1015 ("mt8183-da7219-rt1015"). Embeds in
existing mt8183-da7219-max98357.c because they share most of the code.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200709122445.1584497-6-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The HV/VREF should not turn off if the headphone jack plug-in.
This patch could solve the unexpected interrupt issue in some devices.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200709101345.11449-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In DSP_A mode, BIT7 of IFACE should bit 0 according to datasheet (ie.
inverted frame clock is not support in this mode).
Signed-off-by: Puyou Lu <puyou.lu@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1593657056-4989-1-git-send-email-puyou.lu@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A few places (except for ASoC) are left unconverted for the new
fallthrough pseudo keyword. Now replace them all.
Reviewed-by: Gustavo A. R. Silva <gustavoars@kernel.org>
Link: https://lore.kernel.org/r/20200709111750.8337-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "fall through" comments found in switch-cases in ALSA xen driver
are all superfluous. The kernel coding style allows the multiple
cases in a row. Let's remove them.
Reviewed-by: Gustavo A. R. Silva <gustavoars@kernel.org>
Link: https://lore.kernel.org/r/20200709111750.8337-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The comments about fall through in sound/atmel/ac97.c are just
superfluous and rather confusing. Let's remove them.
Reviewed-by: Gustavo A. R. Silva <gustavoars@kernel.org>
Link: https://lore.kernel.org/r/20200709111750.8337-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Distorted audio appears occasionally, affecting either playback or
capture and requiring the affected substream to be closed by all
applications and re-opened.
The best way I have found to reproduce the bug is to use dmix in
combination with Chromium, which opens the audio device multiple times
in threads. Anecdotally, the problems appear to have increased with
faster CPUs. I ruled out 32-bit counter wrapping; it often happens
much earlier.
Since applying this patch I have not had problems, where previously
they would occur several times a day.
The patch targets the following issues:
* Check for progress using the counter from the hardware, not after it
has been truncated to the buffer.
This is a clean way to address a possible bug where if a whole
ringbuffer advances between interrupts, it goes unnoticed.
* Move last_period state from chip to pipe
This more logically belongs as part of pipe, and code is reasier to
read if it is "counter position last time a period elapsed".
Now the code has no references to period count. A period is just
when the regular counter crosses a threshold. This increases
readability and reduces scope for bugs.
* Treat period notification and buffer advance independently:
This helps to clarify what is the responsibility of the interrupt
handler, and what is pcm_pointer().
Removing shared state between these operations means race conditions
are fixed without introducing locks. Synchronisation is only around
the read of pipe->dma_counter. There may be cache line contention
around "struct audiopipe" but I did not have cause to profile this.
Pay attention to be robust where dma_counter wrapping is not a
multiple of period_size or buffer_size.
This is a revised patch based on feedback from Takashi and Giuliano.
Signed-off-by: Mark Hills <mark@xwax.org>
Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These are valid conditions in normal circumstances, so do not "warn" but
make them for debugging.
Signed-off-by: Mark Hills <mark@xwax.org>
Link: https://lore.kernel.org/r/20200708101848.3457-4-mark@xwax.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use of atomics does not make these statements robust:
atomic_inc(&chip->opencount);
if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
chip->can_set_rate=0;
and
if (atomic_read(&chip->opencount)) {
if (chip->opencount) {
changed = -EAGAIN;
} else {
changed = set_digital_mode(chip, dmode);
It would be necessary to atomically increment or decrement the value
and use the returned result. And yet we still need to prevent other
threads making use of "can_set_rate" while we set it.
However in all but one case the atomic is misleading as they are already
running with "mode_mutex" held.
Decisions are made on mode setting are often intrinsically connected
to "opencount" because some operations are not permitted unless
there is sole ownership.
So instead simplify this, and use "mode_mutex" as a lock for all reference
counting and mode setting.
Signed-off-by: Mark Hills <mark@xwax.org>
Link: https://lore.kernel.org/r/20200708101848.3457-2-mark@xwax.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This check is always false, as it's not the responsibilty of the
device-specific code to make this check. It is already checked
in snd_echo_digital_mode_put.
I do not have a Mona interface to test this change.
This patch is in preparation for follow-up patch to modify the
behavior of "opencount".
Signed-off-by: Mark Hills <mark@xwax.org>
Link: https://lore.kernel.org/r/20200708101848.3457-1-mark@xwax.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small, mostly device-specific fixes.
The significant one is the regression fix for USB-audio implicit
feedback devices due to the incorrect frame size calculation, which
landed in 5.8 and stable trees. In addition, a few usual HD-audio
and USB-audio quirks, Intel HDMI fixes, ASoC fsl and rt5682 fixes,
as well as the fix in compress-offload partial drain operation.
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Merge tag 'sound-5.8-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small, mostly device-specific fixes.
The significant one is the regression fix for USB-audio implicit
feedback devices due to the incorrect frame size calculation, which
landed in 5.8 and stable trees.
In addition, a few usual HD-audio and USB-audio quirks, Intel HDMI
fixes, ASoC fsl and rt5682 fixes, as well as the fix in
compress-offload partial drain operation"
* tag 'sound-5.8-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: compress: fix partial_drain completion state
ALSA: usb-audio: Add implicit feedback quirk for RTX6001
ALSA: usb-audio: add quirk for MacroSilicon MS2109
ALSA: hda/realtek: Enable headset mic of Acer Veriton N4660G with ALC269VC
ALSA: hda/realtek: Enable headset mic of Acer C20-820 with ALC269VC
ALSA: hda/realtek - Enable audio jacks of Acer vCopperbox with ALC269VC
ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id
ALSA: hda/hdmi: improve debug traces for stream lookups
ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
ALSA: opl3: fix infoleak in opl3
ALSA: usb-audio: Replace s/frame/packet/ where appropriate
ALSA: usb-audio: Fix packet size calculation
AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module
ALSA: hda - let hs_mic be picked ahead of hp_mic
ASoC: rt5682: fix the pop noise while OMTP type headset plugin
ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable
ASoC: fsl_mqs: Don't check clock is NULL before calling clk API
This series tries to reuse mt8183-mt6358-ts3a227-max98357.c for supporting
machine driver with rt1015 speaker amplifier.
The 1st patch is straightforward: re-order the header inclusions.
The 2nd patch adds document for the new proposed compatible string.
The 3rd patch changes the machine driver to support either "MAX98357A" or
"RT1015" codecs.
Tzung-Bi Shih (3):
ASoC: mediatek: mt8183: sort header inclusions in alphabetical
dt-bindings: mt8183: add compatible string for using rt1015
ASoC: mediatek: mt8183: support machine driver with rt1015
.../sound/mt8183-mt6358-ts3a227-max98357.txt | 5 +-
sound/soc/mediatek/Kconfig | 5 +-
.../mt8183/mt8183-mt6358-ts3a227-max98357.c | 171 +++++++++++++++---
3 files changed, 153 insertions(+), 28 deletions(-)
--
2.27.0.383.g050319c2ae-goog
While experimenting and introducing errors in Baytrail topology files
until I got them right, I encountered multiple kernel oopses and
memory leaks. This is a first batch to harden the code, but we should
probably think of a tool to fuzz the topology...
Pierre-Louis Bossart (5):
ASoC: topology: fix kernel oops on route addition error
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: use break on errors, not continue
ASoC: topology: factor kfree(se) in error handling
ASoC: topology: add more logs when topology load fails.
sound/soc/soc-topology.c | 97 ++++++++++++++++++++++++----------------
1 file changed, 58 insertions(+), 39 deletions(-)
base-commit: a5911ac579
--
2.25.1
