- 26 Feb, 2020 8 commits
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Mark Brown authored
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>: The first two patches prepare the support of multi-cpu dais for synchronized playback and capture. We remove an unused set of prototypes and add a get_sdw_stream() callback prototype currently missing (the implementation will come later as part of the synchronized playback) The last exposes macros used internally, so that they can be reused to extract information from the _ADR 64-bit values in SOF platform drivers and related machine drivers. I think it's simpler if all these simple patches are merged through the SoundWire tree. With the additional changes to remove the platform drivers and the merge of interrupt handling, that will result in a single immutable tag provided to Mark Brown. Pierre-Louis Bossart (3): soundwire: cadence: remove useless prototypes ASoC: soc-dai: add get_sdw_stream() callback soundwire: add helper macros for devID fields drivers/soundwire/bus.c | 21 +++++---------------- drivers/soundwire/cadence_master.h | 8 -------- include/linux/soundwire/sdw.h | 23 +++++++++++++++++++++++ include/sound/soc-dai.h | 21 +++++++++++++++++++++ 4 files changed, 49 insertions(+), 24 deletions(-) -- 2.20.1
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Dan Murphy authored
The TAS2563 is register compatible with the TAS2562. The main difference is the TAS2563 has a programmable DSP to manage different audio profiles. Signed-off-by:
Dan Murphy <dmurphy@ti.com> Link: https://lore.kernel.org/r/20200226130305.12043-2-dmurphy@ti.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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Dan Murphy authored
Add the Texas Instruments TAS2563 audio amplifier to the TAS262 binding. Signed-off-by:
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Akshu Agrawal authored
ACP_PME_EN allows wake interrupt to be generated when I2S wake feature is enabled. On turning ACP On, ACP_PME_EN gets cleared. Setting the bit back ensures that wake event can be received when ACP is On. Signed-off-by:
Akshu Agrawal <akshu.agrawal@amd.com> Link: https://lore.kernel.org/r/20200226104746.208656-1-akshu.agrawal@amd.comSigned-off-by:
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Oder Chiou authored
This patch revises the function name. Signed-off-by:
Oder Chiou <oder_chiou@realtek.com> Link: https://lore.kernel.org/r/20200219102858.20166-2-oder_chiou@realtek.comSigned-off-by:
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Oder Chiou authored
This patch adds the soundwire support for ALC5682. Signed-off-by:
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Akshu Agrawal authored
Add machine driver for Raven based platform using RT5682 + MAX9836 + CROS_EC codecs Signed-off-by:
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Pierre-Louis Bossart authored
We only have a set() operation, provide the dual get() operation to retrieve the stream information. Signed-off-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200225170041.23644-3-pierre-louis.bossart@linux.intel.comSigned-off-by:
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- 25 Feb, 2020 1 commit
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Colin Ian King authored
The check on p->sink looks bogus, I believe it should be p->source since the following code blocks are related to p->source. Fix this by replacing p->sink with p->source. Fixes: 24c8d141 ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by:
Colin Ian King <colin.king@canonical.com> Addresses-Coverity: ("Copy-paste error") Link: https://lore.kernel.org/r/20191119113640.166940-1-colin.king@canonical.comSigned-off-by:
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- 24 Feb, 2020 14 commits
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Mark Brown authored
Merge branch 'for-5.6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-5.7
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Mark Brown authored
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>: Hi Mark These are soc-pcm cleanup step4. [01/13] - [06/13] are posted but not yet accepted patches. To avoid conflict, I merged these into step4. These patches are already received Reviewed-by from Pierre-Louis. Subject: [PATCH 0/6] ASoC: use for_each_rtd_codec_dai() macro Date: Thu, 13 Feb 2020 13:08:07 +0900 Kuninori Morimoto (13): 1) ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro 2) ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro 3) ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro 4) ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro 5) ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro 6) ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro 7) ASoC: soc-pcm: add snd_soc_dai_get_pcm_stream() 8) ASoC: soc-pcm: cleanup soc_pcm_apply_msb() 9) ASoC: soc-pcm: add snd_soc_dai_get_widget() 10) ASoC: soc-pcm: merge dpcm_run_new/old_update() into dpcm_fe_runtime_update() 11) ASoC: soc-pcm: move dpcm_path_put() to soc-pcm.c 12) ASoC: soc-pcm: move CONFIG_DEBUG_FS functions to top side 13) ASoC: soc-pcm: add dpcm_create/remove_debugfs_state() drivers/soundwire/qcom.c | 7 +- include/sound/soc-dai.h | 15 + include/sound/soc-dapm.h | 1 + include/sound/soc-dpcm.h | 7 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 3 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 8 +- sound/soc/qcom/apq8016_sbc.c | 7 +- sound/soc/qcom/sdm845.c | 20 +- sound/soc/soc-dai.c | 7 +- sound/soc/soc-dapm.c | 20 +- sound/soc/soc-pcm.c | 462 ++++++++---------- 13 files changed, 266 insertions(+), 309 deletions(-) -- 2.17.1 Thank you for your help !! Best regards --- Kuninori Morimoto
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Kai Vehmanen authored
ASoC component open/close and snd_soc_component_module_get/put are called independently for each component-substream pair, so the logic added in commit dd03907b ("ASoC: soc-pcm: call snd_soc_component_open/close() once") was not sufficient and led to PCM playback and module unload errors. Implement handling of failures directly in soc_pcm_components_open(), so that any successfully opened components are closed upon error with other components. This allows to clean up error handling in soc_pcm_open() without adding more state tracking. Fixes: dd03907b ("ASoC: soc-pcm: call snd_soc_component_open/close() once") Signed-off-by:
Kai Vehmanen <kai.vehmanen@linux.intel.com> Tested-by:
Dmitry Osipenko <digetx@gmail.com> Link: https://lore.kernel.org/r/20200220094955.16968-1-kai.vehmanen@linux.intel.comSigned-off-by:
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Kuninori Morimoto authored
soc-pcm.c has implementation which depends on CONFIG_DEBUG_FS. But, we don't want to have random #ifdef. This patch adds dpcm_create/remove_debugfs_state() and care it. Signed-off-by:
Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/877e0jjc9b.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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Kuninori Morimoto authored
This is prepare for CONFIG_DEBUG_FS cleanup Signed-off-by:
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Kuninori Morimoto authored
dpcm_path_put() (A) is calling kfree(*list). The freed list is created by dapm_widget_list_create() (B) which is called from snd_soc_dapm_dai_get_connected_widgets() (C) which is called from dpcm_path_get() (D). (B) dapm_widget_list_create(**list, ...) { ... => *list = kzalloc(); ... } (C) snd_soc_dapm_dai_get_connected_widgets(..., **list, ...) { ... dapm_widget_list_create(list, ...); ... } (D) dpcm_path_get(..., **list) { ... snd_soc_dapm_dai_get_connected_widgets(..., list, ...); ... } (A) dpcm_path_put(**list) { => kfree(*list); } This kind of unbalance code is very difficult to read/understand. To avoid this issue, this patch adds each missing paired function dapm_widget_list_free() for dapm_widget_list_create() (B), and snd_soc_dapm_dai_free_widgets() for snd_soc_dapm_dai_get_connected_widgets() (C). This patch uses these, and moves dpcm_path_put() next to dpcm_path_get(). Signed-off-by: -
Kuninori Morimoto authored
soc-pcm has dpcm_run_new/old_update(), but these are used from dpcm_fe_runtime_update() only, and are very verbose functions. This patch disassembles these. Signed-off-by:
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Kuninori Morimoto authored
soc-pcm.c has dai_get_widget(), but it can be more generic. This patch renames it to snd_soc_dai_get_widget(), and use it. Signed-off-by:
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Kuninori Morimoto authored
soc_pcm_apply_msb() apply msb for CPU/Codec, but, it has duplicate code. The difference is only SNDRV_PCM_STREAM_PLAYBACK and SNDRV_PCM_STEAM_CAPTURE. It is very verbose and duplicate code. This patch simplify code by using snd_soc_dai_get_pcm_stream(). Signed-off-by:
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Kuninori Morimoto authored
DAI driver has playback/capture stream. OTOH, we have SNDRV_PCM_STREAM_PLAYBACK/CAPTURE. Because of this kind of implementation, ALSA SoC needs to have many verbose code. To solve this issue, this patch adds snd_soc_dai_get_pcm_stream() macro to get playback/capture stream pointer from stream. Signed-off-by:
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Amadeusz Sławiński authored
Incrementation of avail_clk_cnt was incorrectly moved to error path. Put it back to success path. Fixes: 6ee927f2 ('ASoC: Intel: Skylake: Fix NULL ptr dereference when unloading clk dev') Signed-off-by:
Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Reviewed-by:
Cezary Rojewski <cezary.rojewski@intel.com> Reviewed-by:
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Kuninori Morimoto authored
commit b0edff42 ("ASoC: soc-pcm/soc-compress: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP") uses snd_soc_dapm_stream_stop() for soc_compr_free_fe() and dpcm_fe_dai_shutdown() because it didn't care about pmdown_time. But, it didn't need to care. This patch rollback to original code. Some system will wait unneeded timed-out without this patch. Special Thanks for reporting to Chris Gorman. ... intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 intel_sst_acpi 808622A8:00: fw returned err -16 sst-mfld-platform sst-mfld-platform: ASoC: PRE_PMD: pcm0_in event failed: -16 ... Fixes: commit b0edff42 ("ASoC: soc-pcm/soc-compress: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP") Reported-by:
Chris Gorman <chrisjohgorman@gmail.com> Signed-off-by:
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kbuild test robot authored
sound/soc/meson/aiu-encoder-i2s.c:129:2-3: Unneeded semicolon Remove unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci Fixes: 3e25c445 ("ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families") Signed-off-by:
kbuild test robot <lkp@intel.com> CC: Martin Blumenstingl <martin.blumenstingl@googlemail.com> Link: https://lore.kernel.org/r/20200222170154.GA119396@e50d7db646c3Signed-off-by:
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Geert Uytterhoeven authored
On i386 randconfig: sound/soc/codecs/wm9705.o: In function `wm9705_soc_resume': wm9705.c:(.text+0x128): undefined reference to `snd_ac97_reset' sound/soc/codecs/wm9712.o: In function `wm9712_soc_resume': wm9712.c:(.text+0x2d1): undefined reference to `snd_ac97_reset' sound/soc/codecs/wm9713.o: In function `wm9713_soc_resume': wm9713.c:(.text+0x820): undefined reference to `snd_ac97_reset' Fix this by adding the missing dependencies on SND_SOC_AC97_BUS. Reported-by:Randy Dunlap <rdunlap@infradead.org> Signed-off-by:
Geert Uytterhoeven <geert@linux-m68k.org> Link: https://lore.kernel.org/r/20200224112537.14483-1-geert@linux-m68k.orgSigned-off-by:
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- 21 Feb, 2020 11 commits
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Dan Murphy authored
Add decimation filter selection support. Per Section 8.3.6.7 the Digital Decimation Filter is selectable between a Linear Phase, Low Latency, and Ultra Low Latency filer. Signed-off-by:
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Dan Murphy authored
The TLV320ADCx140 parts support Dynamic Range Enhancer (DRE) as defined in Section 8.3.2 of the data sheets. The DRE achieves a complete-channel dynamic range as high as 120 dB. At a system level, the DRE scheme enables far-field, high-fidelity recording of audio signals in very quiet environments and low-distortion recording in loud environments. There are 2 enables for DRE. The first is a global setting that enables the DRE engine in the device and the other enable is per channel. If the DRE is enabled globally then either DRE or AGC can be used per each configured channel. If global DRE is disabled then even setting the DRE enable bit in the channel config register will have no effect. Signed-off-by:
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Jerome Brunet authored
Make sure the axg audio card driver recognise the dai_link as a codec-to-codec link if the cpu dai is the internal dac glue. Signed-off-by:
Jerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20200221153607.1585499-4-jbrunet@baylibre.comSigned-off-by:
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Jerome Brunet authored
Add support for the internal audio DAC glue found on the Amlogic g12a and sm1 SoC families. This allows to connect the TDM outputs of the SoC to the internal t9015 audio DAC. Signed-off-by:
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Jerome Brunet authored
Add the DT bindings and documentation of the internal audio DAC glue found on Amlogic g12a and sm1 SoC families Signed-off-by:
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Dan Murphy authored
Add support for digital volume control. There is no dedicated register for volume control but instead there are 4. The values of the registers are determined with exponential floating point math. So a table was created with register values for 2dB step increments from -110dB to 0dB. Signed-off-by:
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Samuel Holland authored
This field is not used anywhere in the driver, so remove it. Fixes: 36c68493 ("ASoC: Add sun8i digital audio codec") Signed-off-by:
Samuel Holland <samuel@sholland.org> Acked-by:
Chen-Yu Tsai <wens@csie.org> Link: https://lore.kernel.org/r/20200217064250.15516-5-samuel@sholland.orgSigned-off-by:
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Martin Blumenstingl authored
The AIU audio controller on the Meson8 and Meson8b SoC families is compatible with the one found in the later GXBB family. Add compatible strings for these two older SoC families so the driver can be loaded for them. Instead of using the I2S divider from the AIU_CLK_CTRL_MORE register we need to use the I2S divider from the AIU_CLK_CTRL register. This older register is less flexible because it only supports four divider settings (1, 2, 4, 8) compared to the AIU_CLK_CTRL_MORE register (which supports dividers in the range 0..64). Signed-off-by:
Martin Blumenstingl <martin.blumenstingl@googlemail.com> Reviewed-by:
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Martin Blumenstingl authored
Introduce a struct aiu_platform_data to make the driver aware of platform specific information. Convert the existing check for the internal stereo audio codec (only available on GXL) to this new struct. Support for the 32-bit SoCs will need this as well because the AIU_CLK_CTRL_MORE register doesn't have the I2S divider bits (and we need to use the I2S divider from AIU_CLK_CTRL instead). Signed-off-by:
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Martin Blumenstingl authored
The AIU audio output controller on the Meson8 and Meson8b SoC families is compatible with the one found in the GXBB family. Document the compatible string for these two older SoCs. Signed-off-by:
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Jerome Brunet authored
Reset the g12a hdmi codec glue on probe. This ensure a sane startup state. Signed-off-by:
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- 20 Feb, 2020 6 commits
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Dan Murphy authored
Add the tlv320adcx140 codec driver family. The TLV320ADCx140 is a Burr-Brown
™ highperformance, audio analog-to-digital converter (ADC) that supports simultaneous sampling of up to four analog channels or eight digital channels for the pulse density modulation (PDM) microphone input. The device supports line and microphone inputs, and allows for both single-ended and differential input configurations. Signed-off-by: -
Dan Murphy authored
Add dt bindings for the TLV320ADCx140 Burr-Brown ADC. The initial support is for the following: TLV320ADC3140 - http://www.ti.com/lit/gpn/tlv320adc3140 TLV320ADC5140 - http://www.ti.com/lit/gpn/tlv320adc5140 TLV320ADC6140 - http://www.ti.com/lit/gpn/tlv320adc6140Signed-off-by:
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Matthias Reichl authored
When we get a clock error during probe we have to call regulator_bulk_disable before bailing out, otherwise we trigger a warning in regulator_put. Fix this by using "goto err" like in the error cases above. Fixes: 5a3af129 ("ASoC: pcm512x: Add PCM512x driver") Signed-off-by:
Matthias Reichl <hias@horus.com> Reviewed-by:
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Kai Vehmanen authored
Add support to configure the HDA controller with an external HDA codec even if iDisp codec in i915 is not available. This can happen for multiple reasons: - internal graphics is disabled on the system - i915 driver is not enabled in kernel or it fails to init - i915 codec reports error in HDA codec probe - HDA codec driver probe fails Address all these scenarios, but keep using the existing topology. In case failures occur, HDMI PCM nodes are created, but they will report error if application tries to use them. No ALSA mixer controls are created. If the external HDA codec init fails as well, SOF probe will return error as before. Signed-off-by:
Ranjani Sridharan <ranjani.sridharan@linux.intel.com> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206085 BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1163677 BugLink: https://github.com/thesofproject/linux/issues/1658 Link: https://lore.kernel.org/r/20200220171028.22023-3-kai.vehmanen@linux.intel.comSigned-off-by:
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Kai Vehmanen authored
Extend the generic HDA driver to support systems where iDisp/HDMI audio codecs are disabled for some reason. Switch codecs to SoC dummy in the affected DAI links. This allows to reuse existing topologies for this case. Signed-off-by:
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Charles Keepax authored
Currently it is possible to get the following bad config: WARNING: unmet direct dependencies detected for SND_SOC_WM5110 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MFD_WM5110 [=n] commit ea00d952 ("ASoC: Use imply for SND_SOC_ALL_CODECS") commit d8dd3f92 ("ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout") After these two patches the machine drivers still selects the SND_SOC_WM5110 symbol which doesn't take account of the dependency added on the MFD_WM5110 symbol, fix this by also adding a dependency on MFD_WM5110 itself. Reported-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200220125654.7064-1-ckeepax@opensource.cirrus.comSigned-off-by:
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