- 05 Jun, 2023 21 commits
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Balamurugan C authored
Moving amp only boards into end of the match table to have better order and maintenance. Reviewed-by:
Bard Liao <yung-chuan.liao@linux.intel.com> Signed-off-by:
Balamurugan C <balamurugan.c@intel.com> Signed-off-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20230602202225.249209-21-pierre-louis.bossart@linux.intel.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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Balamurugan C authored
Adding HDMI-In capture support for the products doesn't have onboard I2S codec.but need to support HDMI-In capture via I2S and audio playback through HDMI/DP monitor. Reviewed-by:
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Pierre-Louis Bossart authored
This device has no 3.5mm jack, only a single amplifier and mic codec. Closes: https://github.com/thesofproject/linux/issues/4399Reviewed-by:
Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by:
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Pierre-Louis Bossart authored
Yet another permutation of devices. Closes: https://github.com/thesofproject/linux/issues/4399Reviewed-by:
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Pierre-Louis Bossart authored
The SKU numbering isn't quite consistent with the existing RaptorLake SKUs but the PCI ID is definitively RaptorLake. Closes: https://github.com/thesofproject/linux/issues/4380Reviewed-by:
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Pierre-Louis Bossart authored
This is a standard configuration we've seen before for TGL. Closes: https://github.com/thesofproject/linux/issues/4380Reviewed-by:
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Bard Liao authored
To support multiple codecs per SoundWire link, we have to assign multiple CPU DAIs to different DAI links sharing the same physical link. This is not possible with the existing code since we assume that only 'Pin2' is used for playback and 'Pin3' used for capture - additional DAIs cannot be handled. This patch enables more CPU DAIs to be used, e.g. "SDW0 Pin2", "SDW0 Pin3", and "SDW0 Pin4" for SDW0-Playback-SimpleJack, SDW0-Capture-SimpleJack, and SDW0-Playback-SmartAmp DAI links on physical link #0. Reviewed-by:
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Bard Liao authored
rt713 is rt712 but without amp. Reviewed-by:
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Bard Liao authored
Add rt712 ID for MTL. Signed-off-by:
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Bard Liao authored
Add rt712 ID for TGL. Signed-off-by:
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Bard Liao authored
Rt712 is a multi function codec which shpports headset, amp, and dmic functions. Rt712 has two sdw interfaces and codec drivers, one for jack and amp, the other for dmic. part id 0x712 is for jack and amp, and 0x1712 is for dmic. Signed-off-by:
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Bard Liao authored
Let rename rt711_sdca to rt_sdca_jack and let it be used for all Realtek sdca jacks. The commit uses component->name_prefix to construct card->components, and determine which codec it is. So, we have to set name_prefix properly. Signed-off-by:
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Bard Liao authored
Jack Detection source can be applied to all jacks, not only rt711. No function changes. Signed-off-by:
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Bard Liao authored
if (!SOF_RT711_JDSRC(sof_sdw_quirk)) is tested in rt711_sdca_add_codec_ device_props(), and we don't add software node to the device if jack source is not set. We need to do the same test in sof_sdw_rt711_sdca_exit(), and avoid removing software node if jack source is not set. Signed-off-by:
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Bard Liao authored
A codec may support multiple dais for different purpose. For example, the rt712 codec supports jack and amp on different dais and machine driver needs to create different dailink for those dais. Reviewed-by:
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Bard Liao authored
We append codec type to dailink name to distinguish different dailink on the same sdw link and direction. But we could create multi dailinks for a codec and the dailink name will be duplicated if we append codec type to the dailink name. Appending dai type instead of codec type can solve the issue. For example, if a codec supports JACK on dai 0 and AMP on dai 1, the existing code will create dailinks SDW0-Playback-SimpleJack or SDW0-Playback-SmartAmp for both dailinks, and it will be SDW0-Playback-SimpleJack for dailink 0 and SDW0-Playback-SmartAmp for dailink 1 after this change. Then codec type is not used any more and can be removed. Reviewed-by:
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Bard Liao authored
codec_info_list[codec_index] is used multiple times in the create_sdw_dailink() function. Adding a codec_info pointer to shorten the code. This is a preparation for the following up patches. No function changed. Reviewed-by:
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Bard Liao authored
Currently, we assign dailink ids in order, and shift with codec type. The purpose is to have consistent dailink ids for topologies. This can be simplified if we have a predefined dailink id in sof_sdw_dai_info. We reuse the existing ids as the predefine ids. So the dailink ids will not be changed by this commit. With this change, we no longer need to check the adr order described in a snd_soc_acpi_link_adr array. Reviewed-by:
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Bard Liao authored
The existing code create a dailink for a codec. However, we may need multi dailinks for a codec. This commit adds a new struct in sof_sdw_codec_info{} to store the dai info of a codec. The initial assumption if that we will create at most 3 dailink types for a codec, since this is the max known with upcoming SDCA devices. We may need to increase this number as new SDCA 'functions' become available. One strong assumption is that all dailinks exposed are independent, as per SDCA directions. This commit just moves some items into the new sof_sdw_dai_info struct. There is no function changed. Multi dais supported will be added in the follow up commits. Reviewed-by: -
Bard Liao authored
Somehow .exit = sof_sdw_rt_amp_exit was missing in rt1318 codec info. Reviewed-by:
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Mark Brown authored
Merge series from David Lin <CTLIN0@nuvoton.com>: This series adds nau8825c support. The driver can be used on NAU8825B and NAU8825C.
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- 02 Jun, 2023 4 commits
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Arnd Bergmann authored
The vangogh driver just gained a link time dependency that now causes randconfig builds to fail: x86_64-linux-ld: sound/soc/amd/vangogh/pci-acp5x.o: in function `snd_acp5x_probe': pci-acp5x.c:(.text+0xbb): undefined reference to `snd_amd_acp_find_config' Fixes: e89f45ed ("ASoC: amd: vangogh: Add check for acp config flags in vangogh platform") Signed-off-by:
Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20230602124447.863476-1-arnd@kernel.orgSigned-off-by:
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David Lin authored
Update the output control for NAU8825C. Signed-off-by:
David Lin <CTLIN0@nuvoton.com> Link: https://lore.kernel.org/r/20230602040924.188913-4-CTLIN0@nuvoton.comSigned-off-by:
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David Lin authored
The FLL is updated to 24 bit with lower power consumption. Signed-off-by:
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David Lin authored
The patch is to update default regmap and register a set of registers for NAU8825C. Signed-off-by:
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- 01 Jun, 2023 5 commits
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Mark Brown authored
Merge series from Walker Chen <walker.chen@starfivetech.com>: This patchset adds TDM audio driver for the StarFive JH7110 SoC. The first patch adds device tree binding for TDM module. The second patch adds tdm driver support for JH7110 SoC. The last patch adds device tree node and pins configuration of tdm to JH7110 dts. The series has been tested on the VisionFive 2 board by plugging an audio expansion board. For more information of audio expansion board, you can take a look at the following webpage: https://wiki.seeedstudio.com/ReSpeaker_2_Mics_Pi_HAT/
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Trevor Wu authored
Add mt8188 and mt8186 .dbg_dump callback to print some information when DSP panic occurs. Signed-off-by:
Trevor Wu <trevor.wu@mediatek.com> Reviewed-by:
Yaochun Hung <yc.hung@mediatek.com> Reviewed-by:
Péter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by:
AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com> Link: https://lore.kernel.org/r/20230601034939.15802-2-trevor.wu@mediatek.comSigned-off-by:
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Richard Fitzgerald authored
Change the message at the start of bin file loading from cs_dsp_dbg() to cs_dsp_info() so that there is confirmation in the kernel log that a bin file was loaded, and the name of the file. Signed-off-by:
Richard Fitzgerald <rf@opensource.cirrus.com> Link: https://lore.kernel.org/r/20230531170158.2744700-1-rf@opensource.cirrus.comSigned-off-by:
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Walker Chen authored
Add tdm driver support for the StarFive JH7110 SoC. Signed-off-by:
Walker Chen <walker.chen@starfivetech.com> Reviewed-by:
Claudiu Beznea <claudiu.beznea@microchip.com> Link: https://lore.kernel.org/r/20230526145402.450-3-walker.chen@starfivetech.comSigned-off-by:
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Walker Chen authored
Add bindings for TDM driver which supports multi-channel audio playback and capture on JH7110 platform. Reviewed-by:
Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> Signed-off-by:
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- 31 May, 2023 7 commits
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Mark Brown authored
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>: This is minor cleanup patches for soc_get_playback_capture().
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David Lin authored
The patch is to manage HSD feature for power saving. The detail is to disable HSD feature after the headset detection is done. When the jack is inserted, the HSD feature will be enabled again. Signed-off-by:
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Kuninori Morimoto authored
soc_get_playback_capture() (A) returns number of substreams for playback/capture, and then, we can use playback/capture_only flag (X)(Y). (A) static int soc_get_playback_capture(...) { ... (X) if (dai_link->playback_only) { (*) *playback = 1; *capture = 0; } (Y) if (dai_link->capture_only) { *playback = 0; (*) *capture = 1; } ... } But this flag should not have effect to opposite side stream (*). This patch tidyup it. Signed-off-by:Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by:
Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Reviewed-by:
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Kuninori Morimoto authored
soc_get_playback_capture() (A) returns number of substreams for playback/capture (B). (A) static int soc_get_playback_capture(..., (B) int *playback, int *capture) { ... for_each_xxx(...) { if (xxx) return -EINVAL; => *playback = 1; ... => *capture = 1; ... } ... } But, it is directly updating playback/capture which is the result of this function even though it might be error. It should be updated in case of succeed only. This patch updates it. Signed-off-by: -
Kuninori Morimoto authored
soc_get_playback_capture() (A) checks dai_link status, and indicate error if it was not matching (B). (A) static int soc_get_playback_capture(...) { ... ^ if (dai_link->dynamic && dai_link->num_cpus > 1) { | dev_err(rtd->dev, (B) "DPCM doesn't support Multi CPU for Front-Ends yet\n"); | return -EINVAL; v } ... } We can use 100 char for 1 line today. This patch cleanup error code line. Signed-off-by: -
Kuninori Morimoto authored
soc_get_playback_capture() (A) is using rtd->dai_link->xxx everywhere. Because of that, 1 line is unnecessarily long and not readable. (A) static int soc_get_playback_capture(...) { if (rtd->dai_link->dynamic ...) { ^^^^^^^^^^^^^ ... } else { int cpu_capture = rtd->dai_link->c2c_params ? ^^^^^^^^^^^^^ ... } if (rtd->dai_link->playback_only) { ^^^^^^^^^^^^^ ... } ... } This patch uses variable "dai_link" to be clear code. Nothing changes the meanings. Signed-off-by: -
Kuninori Morimoto authored
soc_get_playback_capture() (A) returns number of substreams for playback/capture (B). ASoC will probe the Sound Card and mapps CPU<->Codec pair. (A) static int soc_get_playback_capture(..., (B) int *playback, int *capture) { ... if (rtd->dai_link->playback_only) { *playback = 1; *capture = 0; } if (rtd->dai_link->capture_only) { *playback = 0; *capture = 1; } (C) return 0; } But it might be no playback no capture if it returns playback=0, capture=0. It is very difficult to notice about it. This patch indicates error at (C) then. Signed-off-by:
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- 30 May, 2023 3 commits
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Mark Brown authored
Merge series from Cristian Ciocaltea <cristian.ciocaltea@collabora.com>: This patch series handles a few issues related to the ES8316 audio codec, discovered while doing some testing on the Rock 5B board.
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Cristian Ciocaltea authored
When using the codec through the generic audio graph card, there are at least two calls of es8316_set_dai_sysclk(), with the effect of limiting the allowed sample rates according to the MCLK/LRCK ratios supported by the codec: 1. During audio card setup, to set the initial MCLK - see asoc_simple_init_dai(). 2. Before opening a stream, to update MCLK, according to the stream sample rate and the multiplication factor - see asoc_simple_hw_params(). In some cases the initial MCLK might be set to a frequency that doesn't match any of the supported ratios, e.g. 12287999 instead of 12288000, which is only 1 Hz below the supported clock, as that is what the hardware reports. This creates an empty list of rate constraints, which is further passed to snd_pcm_hw_constraint_list() via es8316_pcm_startup(), and causes the following error on the very first access of the sound card: $ speaker-test -D hw:Analog,0 -F S16_LE -c 2 -t wav Broken configuration for playback: no configurations available: Invalid argument Setting of hwparams failed: Invalid argument Note that all subsequent retries succeed thanks to the updated MCLK set at point 2 above, which uses a computed frequency value instead of a reading from the hardware registers. Normally this would have mitigated the issue, but es8316_pcm_startup() executes before the 2nd call to es8316_set_dai_sysclk(), hence it cannot make use of the updated constraints. Since es8316_pcm_hw_params() performs anyway a final validation of MCLK against the stream sample rate and the supported MCLK/LRCK ratios, fix the issue by ensuring that sysclk_constraints list is only set when at least one supported sample rate is autodetected by the codec. Fixes: b8b88b70 ("ASoC: add es8316 codec driver") Signed-off-by:
Cristian Ciocaltea <cristian.ciocaltea@collabora.com> Link: https://lore.kernel.org/r/20230530181140.483936-3-cristian.ciocaltea@collabora.comSigned-off-by:
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Cristian Ciocaltea authored
The following error occurs when trying to restore a previously saved ALSA mixer state (tested on a Rock 5B board): $ alsactl --no-ucm -f /tmp/asound.state store hw:Analog $ alsactl --no-ucm -I -f /tmp/asound.state restore hw:Analog alsactl: set_control:1475: Cannot write control '2:0:0:ALC Capture Target Volume:0' : Invalid argument According to ES8316 datasheet, the register at address 0x2B, which is related to the above mixer control, contains by default the value 0xB0. Considering the corresponding ALC target bits (ALCLVL) are 7:4, the control is initialized with 11, which is one step above the maximum value allowed by the driver: ALCLVL | dB gain -------+-------- 0000 | -16.5 0001 | -15.0 0010 | -13.5 .... | ..... 0111 | -6.0 1000 | -4.5 1001 | -3.0 1010 | -1.5 .... | ..... 1111 | -1.5 The tests performed using the VU meter feature (--vumeter=TYPE) of arecord/aplay confirm the specs are correct and there is no measured gain if the 1011-1111 range would have been mapped to 0 dB: dB gain | VU meter % --------+----------- -6.0 | 30-31 -4.5 | 35-36 -3.0 | 42-43 -1.5 | 50-51 0.0 | 50-51 Increment the max value allowed for ALC Capture Target Volume control, so that it matches the hardware default. Additionally, update the related TLV to prevent an artificial extension of the dB gain range. Fixes: b8b88b70 ("ASoC: add es8316 codec driver") Signed-off-by:
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