- 02 Jul, 2024 3 commits
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Mark Brown authored
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>: simple-audio-mux is designed to be used generally, thus "Input 1" or "Input 2" are used to selecting MUX input. This numbered inputs would work, but might be not user friendly in some case, for example in case of system hardware design has some clear labels. Adds new "state-labels" property and enable to select MUX by own state names. Original > amixer set "MUX" "Input 1" > amixer set "MUX" "Input 2" Use mux-names sound_mux: mux { compatible = "simple-audio-mux"; mux-gpios = <...>; => state-labels = "Label_A", "Label_B"; }; > amixer set "MUX" "Label_A" > amixer set "MUX" "Label_B" -
Mark Brown authored
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>: These commits remove various code that is either no longer needed, or is redundant.
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Mark Brown authored
Merge series from srinivas.kandagatla@linaro.org: Existing way of allocating soundwire master ports on Qualcommm platforms is dynamic, and in linear order starting from 1 to MAX_PORTS. This will work as long as soundwire device ports are 1:1 mapped linearly. However on most Qcom SoCs like SM8550, SM8650, x1e80100, these are NOT mapped in that order. The result of this is that only one speaker among the pair of speakers is always silent, With recent changes for WSA codec to support codec versions and along with these patches we are able to get all speakers working on these SoCs.
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- 01 Jul, 2024 18 commits
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Krzysztof Kozlowski authored
The 'wsa->dev' is assigned closer to the end of the probe() function, so the dev_err() must not use it - it is still NULL at this point. Instead there is already a local 'dev' variable. Fixes: 727de4fb ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version") Signed-off-by:
Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> Reviewed-by:
Dmitry Baryshkov <dmitry.baryshkov@linaro.org> Link: https://patch.msgid.link/20240628095831.207942-2-krzysztof.kozlowski@linaro.orgSigned-off-by:
Mark Brown <broonie@kernel.org>
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Krzysztof Kozlowski authored
The 'rx->dev' is assigned closer to the end of the probe() function, so the dev_err() must not use it - it is still NULL at this point. Instead there is already a local 'dev' variable. Fixes: dbacef05 ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Signed-off-by:
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Nathan Chancellor authored
bitfield.h is not explicitly included but it is required for FIELD_PREP to be expanded by the preprocessor. If it is not implicitly included, there will be a compiler error (as seen with ARCH=hexagon allmodconfig): sound/soc/fsl/lpc3xxx-i2s.c:169:10: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration] 169 | tmp |= LPC3XXX_I2S_WW8 | LPC3XXX_I2S_WS_HP(LPC3XXX_I2S_WW8_HP); | ^ sound/soc/fsl/lpc3xxx-i2s.h:42:30: note: expanded from macro 'LPC3XXX_I2S_WW8' 42 | #define LPC3XXX_I2S_WW8 FIELD_PREP(0x3, 0) /* Word width is 8bit */ | ^ sound/soc/fsl/lpc3xxx-i2s.c:205:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration] 205 | LPC3XXX_I2S_DMA1_TX_EN | LPC3XXX_I2S_DMA0_TX_DEPTH(4)); | ^ sound/soc/fsl/lpc3xxx-i2s.h:65:38: note: expanded from macro 'LPC3XXX_I2S_DMA0_TX_DEPTH' 65 | #define LPC3XXX_I2S_DMA0_TX_DEPTH(s) FIELD_PREP(0xF0000, s) /* Set the DMA1 TX Request level */ | ^ sound/soc/fsl/lpc3xxx-i2s.c:210:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration] 210 | LPC3XXX_I2S_DMA0_RX_EN | LPC3XXX_I2S_DMA1_RX_DEPTH(4)); | ^ sound/soc/fsl/lpc3xxx-i2s.h:70:38: note: expanded from macro 'LPC3XXX_I2S_DMA1_RX_DEPTH' 70 | #define LPC3XXX_I2S_DMA1_RX_DEPTH(s) FIELD_PREP(0x700, s) /* Set the DMA1 RX Request level */ | ^ Include bitfield.h explicitly, so that FIELD_PREP is always expanded, clearing up the compiler error. Fixes: 0959de65 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs") Signed-off-by:Nathan Chancellor <nathan@kernel.org> Link: https://patch.msgid.link/20240701-lpc32xx-asoc-fix-include-for-field_prep-v1-1-0c5d7f71921b@kernel.orgSigned-off-by:
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Nathan Chancellor authored
clang points out that ret may be used uninitialized in lpc32xx_i2s_probe() in an error pointer path (which becomes fatal with CONFIG_WERROR): sound/soc/fsl/lpc3xxx-i2s.c:326:47: error: variable 'ret' is uninitialized when used here [-Werror,-Wuninitialized] 326 | "failed to init register map: %d\n", ret); | ^~~ sound/soc/fsl/lpc3xxx-i2s.c:310:9: note: initialize the variable 'ret' to silence this warning 310 | int ret; | ^ | = 0 1 error generated. One solution would be a small refactoring of the second parameter in dev_err_probe(), PTR_ERR(i2s_info_p->regs), to be the value of ret in the if statement. However, a nicer solution for debugging purposes, which is the point of this statement, would be to use the '%pe' specifier to symbolically print the error pointer value. Do so, which eliminates the uninitialized use of ret, clearing up the warning. Fixes: 0959de65 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs") Signed-off-by: -
Animesh Agarwal authored
Convert the RT5677 audio CODEC bindings to DT schema. Signed-off-by:
Animesh Agarwal <animeshagarwal28@gmail.com> Cc: Daniel Baluta <daniel.baluta@nxp.com> Reviewed-by:
Rob Herring (Arm) <robh@kernel.org> Link: https://patch.msgid.link/20240627105030.14360-1-animeshagarwal28@gmail.comSigned-off-by:
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Shenghao Ding authored
Add name_prefix as the prefix name of DSP firmwares and calibrated data files which stored speaker calibrated impedance. Signed-off-by:
Shenghao Ding <shenghao-ding@ti.com> Link: https://patch.msgid.link/20240629101112.628-1-shenghao-ding@ti.comSigned-off-by:
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Srinivas Kandagatla authored
Add support to parse static master port map information from device tree. This is required for correct port mapping between soundwire device and master ports. Reviewed-by:
Neil Armstrong <neil.armstrong@linaro.org> Tested-by: Neil Armstrong <neil.armstrong@linaro.org> # on SM8650-HDK Signed-off-by:
Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by:
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Srinivas Kandagatla authored
Document port mapping property for wsa884x. Port mapping is required to be able map correct master ports. All the device ports are not mapped in same order as master ports, so there is a need for having static port mapping for WSA codecs. Reviewed-by:
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Srinivas Kandagatla authored
Add support to parse static master port map information from device tree. Reviewed-by:
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Manikantan R authored
Document port mapping property for wsa883x. Port mapping is required to be able map correct master ports for VI feedback. All the device ports are not mapped in same order as master ports, so there is a need for having static port mapping for WSA codecs. Signed-off-by:
Manikantan R <quic_manrav@quicinc.com> Reviewed-by:
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Kuninori Morimoto authored
simple-audio-mux is designed to be used generally, thus "Input 1" or "Input 2" are used to selecting MUX input. This numbered inputs would work, but might be not user friendly in some case, for example in case of system hardware design has some clear labels. Adds new "state-labels" property and enable to select MUX by own state names. Signed-off-by:
Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by:
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Kuninori Morimoto authored
simple-audio-mux is designed to be used generally, thus "Input 1" or "Input 2" are used to selecting MUX input. This numbered inputs would work, but might be not user friendly in some case, for example in case of system hardware design has some clear labels. Adds new "state-labels" property and enable to select MUX by own state names. Original > amixer set "MUX" "Input 1" Use mux-names sound_mux: mux { compatible = "simple-audio-mux"; mux-gpios = <...>; state-labels = "Label_A", "Label_B"; }; > amixer set "MUX" "Label_A" Signed-off-by: -
Animesh Agarwal authored
Convert the RT5659/RT5658 audio CODEC bindings to DT schema. Signed-off-by:
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Chancel Liu authored
Add compatible string and specific soc data to support rpmsg sound card on i.MX95 platform. Signed-off-by:
Chancel Liu <chancel.liu@nxp.com> Acked-by:
Shengjiu Wang <shengjiu.wang@gmail.com> Link: https://patch.msgid.link/20240626071202.7149-2-chancel.liu@nxp.comSigned-off-by:
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Chancel Liu authored
Add compatible string for i.MX95 platform which supports audio function through rpmsg channel between Cortex-A and Cortex-M core. Signed-off-by:
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Richard Fitzgerald authored
struct sdw_slave_prop is zero-initialized by the SoundWire core so there is no need to clear clk_stop_mode1 to false. Removing this also avoids having an unnecessary build dependency on a struct member. Signed-off-by:
Richard Fitzgerald <rf@opensource.cirrus.com> Link: https://patch.msgid.link/20240701104444.172556-4-rf@opensource.cirrus.comSigned-off-by:
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Richard Fitzgerald authored
No product was ever released with A1 silicon so there is no need for the driver to include support for it. Signed-off-by:
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Richard Fitzgerald authored
This patch reverts a series of commits that allowed for the ASP registers to be owned by either the driver or the firmware. Nothing currently depends on the functionality that is being reverted, so it is safe to remove. The commits being reverted are (last 3 are bugfixes to the first 2): commit 72a77d76 ("ASoC: cs35l56: Fix to ensure ASP1 registers match cache") commit 07f7d6e7 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers") commit 4703b014 ("ASoC: cs35l56: fix reversed if statement in cs35l56_dspwait_asp1tx_put()") commit c14f09f0 ("ASoC: cs35l56: Fix deadlock in ASP1 mixer register initialization") commit dfd2ffb3 ("ASoC: cs35l56: Prevent overwriting firmware ASP config") These reverts have been squashed into a single commit because there would be no reason to revert only some of them (which would just reintroduce bugs). The changes introduced by the commits were well-intentioned but somewhat misguided. ACPI does not provide any information about how audio hardware is linked together, so that information has to be hardcoded into drivers. On Windows the firmware is customized to statically setup appropriate configuration of the audio links, and the intent of the commits was to re-use this information if the Linux host drivers aren't taking control of the ASP. This would avoid having to hardcode the ASP config into the machine driver on some systems. However, this added complexity and race conditions into the driver. It also complicates implementation of new code. The only case where the ASP is used but the host is not taking ownership is when CS35L56 is used in SoundWire mode with the ASP as a reference audio interconnect. But even in that case it's not necessarily required even if the firmware initialized it. Typically it is used to avoid the host SDCA drivers having to be capable of aggregating capture paths from multiple SoundWire peripherals. But the SOF SoundWire support is capable of doing that aggregation. Reverting all these commits significantly simplifies the driver. Let's just use the normal Linux mechanisms of the machine driver and ALSA controls to set things up instead of trying to use the firmware to do use-case setup. Signed-off-by:
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- 28 Jun, 2024 8 commits
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Mark Brown authored
Merge series from Piotr Wojtaszczyk <piotr.wojtaszczyk@timesys.com>: This pach set is to bring back audio to machines with a LPC32XX CPU. The legacy LPC32XX SoC used to have audio spport in linux 2.6.27. The support was dropped due to lack of interest from mainaeners.
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Jerome Brunet authored
Add support for 705.6kHz and 768kHz sample rates Signed-off-by:
Jerome Brunet <jbrunet@baylibre.com> Link: https://patch.msgid.link/20240628123256.2019224-1-jbrunet@baylibre.comSigned-off-by:
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Shengjiu Wang authored
On i.MX8MP, there are 3 interrupts, the third interrupt is SPDIF wakeup interrupt from PHY Add ref: dai-common.yaml for #sound-dai-cells is needed and XCVR is a DAI device. Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Reviewed-by:
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Chen Ni authored
clk_prepare_enable() may fail, so we should better check its return value and propagate it in the case of error. Fixes: 62a7fc32 ("ASoC: max98088: Add master clock handling") Signed-off-by:
Chen Ni <nichen@iscas.ac.cn> Link: https://patch.msgid.link/20240628080534.843815-1-nichen@iscas.ac.cnSigned-off-by:
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Yang Li authored
./sound/soc/codecs/ak4619.c:757:2-3: Unneeded semicolon Reported-by:
Abaci Robot <abaci@linux.alibaba.com> Closes: https://bugzilla.openanolis.cn/show_bug.cgi?id=9442Signed-off-by:
Yang Li <yang.lee@linux.alibaba.com> Link: https://patch.msgid.link/20240628052406.36644-1-yang.lee@linux.alibaba.comSigned-off-by:
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Jerome Brunet authored
The dummy DAI should allow any (reasonable) rates possible. Make the rate continuous for dummy and set range from 5512Hz to 768kHz The change is mostly cosmetic as dummy is skipped when setting the hwparams. Signed-off-by:
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Piotr Wojtaszczyk authored
This driver was ported from an old version in linux 2.6.27 and adjusted for the new ASoC framework and DMA API. Signed-off-by:
Piotr Wojtaszczyk <piotr.wojtaszczyk@timesys.com> Link: https://patch.msgid.link/20240627150046.258795-12-piotr.wojtaszczyk@timesys.comSigned-off-by:
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Piotr Wojtaszczyk authored
Add nxp,lpc3220-i2s DT binding documentation. Signed-off-by:
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- 27 Jun, 2024 3 commits
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Krzysztof Kozlowski authored
The Qualcomm LPASS WSA macro codec driver uses now parts of common module, so it has to select SND_SOC_LPASS_MACRO_COMMON. sound/soc/codecs/lpass-wsa-macro.o: in function `wsa_macro_probe': sound/soc/codecs/lpass-wsa-macro.c:2767:(.text+0x1c9c): undefined reference to `lpass_macro_get_codec_version' Reported-by:
kernel test robot <lkp@intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202406272231.th1LtuLk-lkp@intel.com/ Fixes: 5dcf442b ("ASoC: codecs: lpass-wsa-macro: Prepare to accommodate new codec versions") Signed-off-by:
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Mark Brown authored
Merge series from Shengjiu Wang <shengjiu.wang@nxp.com>: The transmitter and receiver part of the SAI interface need to be configured with different master/slave mode, especially to work with the audiomix module. The SAI1 TX is in master mode, but SAI1 RX is in slave mode. So add another two DAIs for TX and RX separately in fsl_sai driver. There will be three devices for audiomix sound card, hw:x,0 is the playback device for one SAI, hw:x,1 is the playback device for another SAI, hw:x,2 is the capture device for audmix output.
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Mark Brown authored
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>: Introduce the ability for sound cards to manually order the startup of the various components in the card.
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- 26 Jun, 2024 8 commits
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Shuming Fan authored
The version B will support the multi-lane function and integrate the DMIC function in one SoundWire interface. Due to some registers having different default values between version A and B, this patch also removes the redundant default registers to avoid confusion. Signed-off-by:
Shuming Fan <shumingf@realtek.com> Link: https://patch.msgid.link/20240620103237.2124196-1-shumingf@realtek.comSigned-off-by:
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Shuming Fan authored
This patch doesn't have any change of functionality. Signed-off-by:
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Neil Armstrong authored
Like "audio-routing" drop the minItems: 2 from the "audio-widgets", because any limit here - lower or upper- is rather meaningless. This will also fix `dtbs_check` warnings like: sound: audio-widgets: ['Speaker', '7J4-14 LEFT', 'Speaker', '7J4-11 RIGHT'] is too long Signed-off-by:
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Aleksandr Mishin authored
If IORESOURCE_MEM "lpass-rxtx-cdc-dma-lpm" or "lpass-va-cdc-dma-lpm" resources is not provided in Device Tree due to any error, platform_get_resource_byname() will return NULL which is later dereferenced. According to sound/qcom,lpass-cpu.yaml, these resources are provided, but DT can be broken due to any error. In such cases driver must be able to protect itself, since the DT is external data for the driver. Adjust this issues by adding NULL return check. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: b1387062 ("ASoC: qcom: Add regmap config support for codec dma driver") Signed-off-by:
Aleksandr Mishin <amishin@t-argos.ru> Link: https://patch.msgid.link/20240605104953.12072-1-amishin@t-argos.ruSigned-off-by:
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Shengjiu Wang authored
There will be three devices for this sound card, hw:x,0 is the playback device for one SAI, hw:x,1 is the playback device for another SAI, hw:x,2 is the capture device for audmix output. then capture device and playback device can be configured with different master/slave mode. Signed-off-by:
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Shengjiu Wang authored
As audmix requires playback and capture stream in different master/slave mode, so separate playback and capture stream to different DAI. There are three DAIs required, two DAIs for playback one DAI for capture. Signed-off-by:
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Shengjiu Wang authored
The transmitter and receiver part of the SAI interface need to be configured with different master/slave mode, especially to work with the audiomix module. +-------+ +-----------+ | SAI1 | --TX--> | | | | <--RX-- | | +-------+ | | | AUDIOMIX | +-------+ | | | SAI2 | --TX--> | | +-------+ +-----------+ The SAI1 TX is in master mode, but SAI1 RX is in slave mode. So add another two DAIs for TX and RX separately. but only defined fsl_sai_set_dai_fmt_tx() and fsl_sai_set_dai_fmt_rx() ops function for current case, in the future, the other ops function for TX and RX can be defined if required. Signed-off-by: -
Kuninori Morimoto authored
Some Sound Card might need special trigger ordering which is based on CPU/Codec connection. It is already supported on ASoC, but Audio Graph Card2 still not yet support it. Let's support it. Cc: Maxim Kochetkov <fido_max@inbox.ru> Signed-off-by:
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