December 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 00/16] ASoC: Intel: avs: Data probing and fw logging
by Cezary Rojewski 05 Dec '22

05 Dec '22

The patchset focuses on debug functionality for the avs-driver. Two major blocks are covered here: data probing and AudioDSP firmware logging. Both are configured and controlled through debugfs. Data probing is a AudioDSP debug functionality which allows for gathering the actual data that is being routed to or from a module. Helps in debugging its processing capabilities - navigate to a specific module which may have caused a glitch within a pipeline (set of modules bound together). First few allow for assigning compress stream to a HDAudio stream, what is currently limited to pcm substreams only. These patches were already present on this list and reviewed in the past [1]. The next few tidy existing debug-related code up so it's ready for addition of new functionalities and make it clear which part of the avs is debug related and which is not. These also simplify the existing locking around the trace fifo. Afterward, debug-related IPCs are defined along with stub soc-component and compress DAI operations. Not much is done there as it's not a standard PCM streaming scenario. Most code found in compress operations is inherited from the HOST side of HDAudio streaming found in pcm.c file of the driver. Finally, a debugfs file operations are defined. These facilitate connecting to DSP modules from which the data shall be gathered as well as control and configuration of firmware logging. Additionally, entries are added to allow for dumping snapshots of key memory windows. Changes in v2: - reordered sign offs in patch 03 as reported by Takashi - fixed issue reported by ikp - unintilialized 'runtime' variable in patch 03 - added missing module_get_upon_open in component driver declaration [1]: https://lore.kernel.org/alsa-devel/20200124190413.18154-1-cezary.rojewski@i… Cezary Rojewski (16): ALSA: hda: Allow for compress stream to hdac_ext_stream assignment ALSA: hda: Prepare for compress stream support ALSA: hda: Interrupt servicing and BDL setup for compress streams ASoC: Intel: avs: Introduce avs_log_buffer_status_locked() ASoC: Intel: avs: Drop fifo_lock ASoC: Intel: avs: Introduce debug-context aware helpers ASoC: Intel: avs: Make enable_logs() dependent on DEBUG_FS ASoC: Intel: avs: Drop usage of debug members in non-debug code ASoC: Intel: avs: Add data probing requests ASoC: Intel: avs: Probe compress operations ASoC: Intel: avs: Data probing soc-component ASoC: Intel: avs: Add probe machine board ASoC: Intel: avs: Probing and firmware tracing over debugfs ASoC: Intel: avs: Gather remaining logs on strace_release() ASoC: Intel: avs: Allow for dumping FW_REGS area ASoC: Intel: avs: Allow for dumping debug window snapshot include/sound/hdaudio_ext.h | 2 + sound/hda/ext/hdac_ext_stream.c | 41 +++ sound/hda/hdac_controller.c | 4 +- sound/hda/hdac_stream.c | 51 +-- sound/soc/intel/Kconfig | 1 + sound/soc/intel/avs/Makefile | 4 + sound/soc/intel/avs/apl.c | 24 +- sound/soc/intel/avs/avs.h | 80 ++++- sound/soc/intel/avs/board_selection.c | 33 ++ sound/soc/intel/avs/boards/Kconfig | 8 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/probe.c | 64 ++++ sound/soc/intel/avs/core.c | 2 + sound/soc/intel/avs/debugfs.c | 436 ++++++++++++++++++++++++++ sound/soc/intel/avs/ipc.c | 2 +- sound/soc/intel/avs/messages.c | 104 +++++- sound/soc/intel/avs/messages.h | 53 ++++ sound/soc/intel/avs/pcm.c | 6 +- sound/soc/intel/avs/probes.c | 313 ++++++++++++++++++ sound/soc/intel/avs/registers.h | 3 +- sound/soc/intel/avs/skl.c | 17 +- sound/soc/intel/avs/utils.c | 22 -- 22 files changed, 1172 insertions(+), 100 deletions(-) create mode 100644 sound/soc/intel/avs/boards/probe.c create mode 100644 sound/soc/intel/avs/debugfs.c create mode 100644 sound/soc/intel/avs/probes.c -- 2.25.1

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[PATCH] ASoC: sof_es8336: fix possible use-after-free in sof_es8336_remove()
by Yang Yingliang 05 Dec '22

05 Dec '22

sof_es8336_remove() calls cancel_delayed_work(). However, that function does not wait until the work function finishes. This means that the callback function may still be running after the driver's remove function has finished, which would result in a use-after-free. Fix by calling cancel_delayed_work_sync(), which ensures that the work is properly cancelled, no longer running, and unable to re-schedule itself. Fixes: 89cdb224f2ab ("ASoC: sof_es8336: reduce pop noise on speaker") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/intel/boards/sof_es8336.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 70713e4b07dc..773e5d1d87d4 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -783,7 +783,7 @@ static int sof_es8336_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); - cancel_delayed_work(&priv->pcm_pop_work); + cancel_delayed_work_sync(&priv->pcm_pop_work); gpiod_put(priv->gpio_speakers); device_remove_software_node(priv->codec_dev); put_device(priv->codec_dev); -- 2.25.1

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[PATCH 0/2] Add SOF panic dump support for AMD platform.
by V sujith kumar Reddy 05 Dec '22

05 Dec '22

This patch series consists of >> Use poll function instead of simple read ACP_SHA_DSP_FW_QUALIFIER >> Add support for IPC and DSP dumps Ajye Huang (1): ASoC: SOF: amd: Use poll function instead to read ACP_SHA_DSP_FW_QUALIFIER V sujith kumar Reddy (1): ASoC : SOF: amd: Add support for IPC and DSP dumps sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/amd/acp-common.c | 108 +++++++++++++++++++++++++++++++++ sound/soc/sof/amd/acp-ipc.c | 13 ++++ sound/soc/sof/amd/acp.c | 8 ++- sound/soc/sof/amd/acp.h | 6 ++ 5 files changed, 133 insertions(+), 3 deletions(-) -- 2.25.1

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[PATCH] ASoC: qcom: lpass-sc7280: Add maybe_unused tag for system PM ops
by Srinivasa Rao Mandadapu 05 Dec '22

05 Dec '22

Add __maybe_unused tag for system PM ops suspend and resume. This is required to fix allmodconfig compilation issue. Fixes: a3a96e93cc88 ("ASoC: qcom: lpass-sc7280: Add system suspend/resume PM ops") Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- sound/soc/qcom/lpass-sc7280.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-sc7280.c b/sound/soc/qcom/lpass-sc7280.c index 65a04d3..d43f480 100644 --- a/sound/soc/qcom/lpass-sc7280.c +++ b/sound/soc/qcom/lpass-sc7280.c @@ -235,14 +235,14 @@ static int sc7280_lpass_exit(struct platform_device *pdev) return 0; } -static int sc7280_lpass_dev_resume(struct device *dev) +static int __maybe_unused sc7280_lpass_dev_resume(struct device *dev) { struct lpass_data *drvdata = dev_get_drvdata(dev); return clk_bulk_prepare_enable(drvdata->num_clks, drvdata->clks); } -static int sc7280_lpass_dev_suspend(struct device *dev) +static int __maybe_unused sc7280_lpass_dev_suspend(struct device *dev) { struct lpass_data *drvdata = dev_get_drvdata(dev); @@ -253,6 +253,7 @@ static int sc7280_lpass_dev_suspend(struct device *dev) static const struct dev_pm_ops sc7280_lpass_pm_ops = { SET_SYSTEM_SLEEP_PM_OPS(sc7280_lpass_dev_suspend, sc7280_lpass_dev_resume) }; + static struct lpass_variant sc7280_data = { .i2sctrl_reg_base = 0x1000, .i2sctrl_reg_stride = 0x1000, -- 2.7.4

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[PATCH] ASoC: dt-bindings: maxim,max98504: Convert to DT schema
by Krzysztof Kozlowski 05 Dec '22

05 Dec '22

Convert the Maxim Integrated MAX98504 amplifier bindings to DT schema. Few properties are made optional: 1. interrupts: current Linux driver implementation does not use them, 2. supplies: on some boards these might be wired to battery, for which no regulator is provided. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../devicetree/bindings/sound/max98504.txt | 44 ---------- .../bindings/sound/maxim,max98504.yaml | 86 +++++++++++++++++++ 2 files changed, 86 insertions(+), 44 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/max98504.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98504.yaml diff --git a/Documentation/devicetree/bindings/sound/max98504.txt b/Documentation/devicetree/bindings/sound/max98504.txt deleted file mode 100644 index 583ed5fdfb28..000000000000 --- a/Documentation/devicetree/bindings/sound/max98504.txt +++ /dev/null @@ -1,44 +0,0 @@ -Maxim MAX98504 class D mono speaker amplifier - -This device supports I2C control interface and an IRQ output signal. It features -a PCM and PDM digital audio interface (DAI) and a differential analog input. - -Required properties: - - - compatible : "maxim,max98504" - - reg : should contain the I2C slave device address - - DVDD-supply, DIOVDD-supply, PVDD-supply: power supplies for the device, - as covered in ../regulator/regulator.txt - - interrupts : should specify the interrupt line the device is connected to, - as described in ../interrupt-controller/interrupts.txt - -Optional properties: - - - maxim,brownout-threshold - the PVDD brownout threshold, the value must be - from 0, 1...21 range, corresponding to 2.6V, 2.65V...3.65V voltage range - - maxim,brownout-attenuation - the brownout attenuation to the speaker gain - applied during the "attack hold" and "timed hold" phase, the value must be - from 0...6 (dB) range - - maxim,brownout-attack-hold-ms - the brownout attack hold phase time in ms, - 0...255 (VBATBROWN_ATTK_HOLD, register 0x0018) - - maxim,brownout-timed-hold-ms - the brownout timed hold phase time in ms, - 0...255 (VBATBROWN_TIME_HOLD, register 0x0019) - - maxim,brownout-release-rate-ms - the brownout release phase step time in ms, - 0...255 (VBATBROWN_RELEASE, register 0x001A) - -The default value when the above properties are not specified is 0, -the maxim,brownout-threshold property must be specified to actually enable -the PVDD brownout protection. - -Example: - - max98504@31 { - compatible = "maxim,max98504"; - reg = <0x31>; - interrupt-parent = <&gpio_bank_0>; - interrupts = <2 0>; - - DVDD-supply = <&regulator>; - DIOVDD-supply = <&regulator>; - PVDD-supply = <&regulator>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98504.yaml b/Documentation/devicetree/bindings/sound/maxim,max98504.yaml new file mode 100644 index 000000000000..23f19a9d2c06 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98504.yaml @@ -0,0 +1,86 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98504.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98504 class D mono speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + +description: + Maxim Integrated MAX98504 speaker amplifier supports I2C control interface + with an IRQ output signal, PCM and PDM digital audio interface (DAI) and a + differential analog input. + +properties: + compatible: + const: maxim,max98504 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + DIOVDD-supply: true + DVDD-supply: true + PVDD-supply: true + + maxim,brownout-threshold: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 21 + default: 0 + description: + PVDD brownout threshold, where values correspond to 2.6V, 2.65V...3.65V + voltage range. Property also enables the PVDD brownout protection. + + maxim,brownout-attenuation: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 6 + default: 0 + description: + Brownout attenuation to the speaker gain applied during the "attack hold" + and "timed hold" phase, the value must be from 0...6 (dB) range. + + maxim,brownout-attack-hold-ms: + maximum: 255 + default: 0 + description: + Brownout attack hold phase time in ms, VBATBROWN_ATTK_HOLD, register 0x0018. + + maxim,brownout-timed-hold-ms: + maximum: 255 + default: 0 + description: + Brownout timed hold phase time in ms, VBATBROWN_TIME_HOLD, register 0x0019. + + maxim,brownout-release-rate-ms: + maximum: 255 + default: 0 + description: + Brownout release phase step time in ms, VBATBROWN_RELEASE, register 0x001A. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@31 { + compatible = "maxim,max98504"; + reg = <0x31>; + + DIOVDD-supply = <&ldo3_reg>; + DVDD-supply = <&ldo3_reg>; + }; + }; -- 2.34.1

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[PATCH 1/2] dt-bindings: Drop Jee Heng Sia
by Krzysztof Kozlowski 05 Dec '22

05 Dec '22

Emails to Jee Heng Sia bounce ("550 #5.1.0 Address rejected."). Add Keembay platform maintainers as Keembay I2S maintainers. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Documentation/devicetree/bindings/dma/snps,dw-axi-dmac.yaml | 1 - Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml | 3 ++- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/dma/snps,dw-axi-dmac.yaml b/Documentation/devicetree/bindings/dma/snps,dw-axi-dmac.yaml index 67aa7bb6d36a..ad107a4d3b33 100644 --- a/Documentation/devicetree/bindings/dma/snps,dw-axi-dmac.yaml +++ b/Documentation/devicetree/bindings/dma/snps,dw-axi-dmac.yaml @@ -8,7 +8,6 @@ title: Synopsys DesignWare AXI DMA Controller maintainers: - Eugeniy Paltsev <Eugeniy.Paltsev(a)synopsys.com> - - Jee Heng Sia <jee.heng.sia(a)intel.com> description: Synopsys DesignWare AXI DMA Controller DT Binding diff --git a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml index 2ac0a4b3cd18..33ab0be036a1 100644 --- a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml @@ -8,7 +8,8 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Intel KeemBay I2S maintainers: - - Sia, Jee Heng <jee.heng.sia(a)intel.com> + - Daniele Alessandrelli <daniele.alessandrelli(a)intel.com> + - Paul J. Murphy <paul.j.murphy(a)intel.com description: | Intel KeemBay I2S -- 2.34.1

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[PATCH][next] ASoC: rt715: Make read-only arrays capture_reg_H and capture_reg_L static const
by Colin Ian King 05 Dec '22

05 Dec '22

Don't populate the read-only arrays capture_reg_H and capture_reg_L on the stack but instead make them static const. Also makes the object code a little smaller. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/rt715.c | 48 ++++++++++++++++++++-------------------- 1 file changed, 24 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 917a04092da2..1a2036ccfbac 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -204,12 +204,12 @@ static int rt715_set_main_switch_put(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); - unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, - RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, - RT715_SET_GAIN_MIX_ADC2_H}; - unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, - RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, - RT715_SET_GAIN_MIX_ADC2_L}; + static const unsigned int capture_reg_H[] = { + RT715_SET_GAIN_MIC_ADC_H, RT715_SET_GAIN_LINE_ADC_H, + RT715_SET_GAIN_MIX_ADC_H, RT715_SET_GAIN_MIX_ADC2_H }; + static const unsigned int capture_reg_L[] = { + RT715_SET_GAIN_MIC_ADC_L, RT715_SET_GAIN_LINE_ADC_L, + RT715_SET_GAIN_MIX_ADC_L, RT715_SET_GAIN_MIX_ADC2_L }; unsigned int addr_h, addr_l, val_h = 0x0, val_ll, val_lr; unsigned int k_shift = RT715_DIR_IN_SFT, k_changed = 0; unsigned int read_ll, read_rl, i, j, loop_cnt = 4; @@ -284,12 +284,12 @@ static int rt715_set_main_switch_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); - unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, - RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, - RT715_SET_GAIN_MIX_ADC2_H}; - unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, - RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, - RT715_SET_GAIN_MIX_ADC2_L}; + static const unsigned int capture_reg_H[] = { + RT715_SET_GAIN_MIC_ADC_H, RT715_SET_GAIN_LINE_ADC_H, + RT715_SET_GAIN_MIX_ADC_H, RT715_SET_GAIN_MIX_ADC2_H }; + static const unsigned int capture_reg_L[] = { + RT715_SET_GAIN_MIC_ADC_L, RT715_SET_GAIN_LINE_ADC_L, + RT715_SET_GAIN_MIX_ADC_L, RT715_SET_GAIN_MIX_ADC2_L }; unsigned int addr_h, addr_l, val_h = 0x0, i, loop_cnt = 4; unsigned int read_ll, read_rl; @@ -312,12 +312,12 @@ static int rt715_set_main_vol_put(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); - unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, - RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, - RT715_SET_GAIN_MIX_ADC2_H}; - unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, - RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, - RT715_SET_GAIN_MIX_ADC2_L}; + static const unsigned int capture_reg_H[] = { + RT715_SET_GAIN_MIC_ADC_H, RT715_SET_GAIN_LINE_ADC_H, + RT715_SET_GAIN_MIX_ADC_H, RT715_SET_GAIN_MIX_ADC2_H }; + static const unsigned int capture_reg_L[] = { + RT715_SET_GAIN_MIC_ADC_L, RT715_SET_GAIN_LINE_ADC_L, + RT715_SET_GAIN_MIX_ADC_L, RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h = 0x0, val_ll, val_lr; unsigned int read_ll, read_rl, i, j, loop_cnt = 4, k_changed = 0; unsigned int k_shift = RT715_DIR_IN_SFT, k_max = 0x3f; @@ -393,12 +393,12 @@ static int rt715_set_main_vol_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); - unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, - RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, - RT715_SET_GAIN_MIX_ADC2_H}; - unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, - RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, - RT715_SET_GAIN_MIX_ADC2_L}; + static const unsigned int capture_reg_H[] = { + RT715_SET_GAIN_MIC_ADC_H, RT715_SET_GAIN_LINE_ADC_H, + RT715_SET_GAIN_MIX_ADC_H, RT715_SET_GAIN_MIX_ADC2_H }; + static const unsigned int capture_reg_L[] = { + RT715_SET_GAIN_MIC_ADC_L, RT715_SET_GAIN_LINE_ADC_L, + RT715_SET_GAIN_MIX_ADC_L, RT715_SET_GAIN_MIX_ADC2_L }; unsigned int addr_h, addr_l, val_h = 0x0, i, loop_cnt = 4; unsigned int read_ll, read_rl; -- 2.38.1

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[PATCH][next] ASoC: wcd938x: Make read-only array minCode_param static const
by Colin Ian King 05 Dec '22

05 Dec '22

Don't populate the read-only array minCode_param on the stack but instead make it static const. Also makes the object code a little smaller. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/wcd938x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index aca06a4026f3..fcac763b04d1 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3128,8 +3128,8 @@ static inline void wcd938x_mbhc_get_result_params(struct wcd938x_priv *wcd938x, s16 c1; s32 x1, d1; int32_t denom; - int minCode_param[] = { - 3277, 1639, 820, 410, 205, 103, 52, 26 + static const int minCode_param[] = { + 3277, 1639, 820, 410, 205, 103, 52, 26 }; regmap_update_bits(wcd938x->regmap, WCD938X_ANA_MBHC_ZDET, 0x20, 0x20); -- 2.38.1

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[PATCH][next] ASoC: uniphier: aio-core: Make some read-only arrays static const
by Colin Ian King 05 Dec '22

05 Dec '22

Don't populate the read-only arrays slotsel_2ch, slotsel_multi, v_pll and v_div on the stack but instead make them static const. Also makes the object code a little smaller. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/uniphier/aio-core.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index b8195778953e..0eba60758134 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -277,17 +277,18 @@ void aio_port_reset(struct uniphier_aio_sub *sub) static int aio_port_set_ch(struct uniphier_aio_sub *sub) { struct regmap *r = sub->aio->chip->regmap; - u32 slotsel_2ch[] = { + static const u32 slotsel_2ch[] = { 0, 0, 0, 0, 0, }; - u32 slotsel_multi[] = { + static const u32 slotsel_multi[] = { OPORTMXTYSLOTCTR_SLOTSEL_SLOT0, OPORTMXTYSLOTCTR_SLOTSEL_SLOT1, OPORTMXTYSLOTCTR_SLOTSEL_SLOT2, OPORTMXTYSLOTCTR_SLOTSEL_SLOT3, OPORTMXTYSLOTCTR_SLOTSEL_SLOT4, }; - u32 mode, *slotsel; + u32 mode; + const u32 *slotsel; int i; switch (params_channels(&sub->params)) { @@ -515,13 +516,13 @@ static int aio_port_set_clk(struct uniphier_aio_sub *sub) struct uniphier_aio_chip *chip = sub->aio->chip; struct device *dev = &sub->aio->chip->pdev->dev; struct regmap *r = sub->aio->chip->regmap; - u32 v_pll[] = { + static const u32 v_pll[] = { OPORTMXCTR2_ACLKSEL_A1, OPORTMXCTR2_ACLKSEL_F1, OPORTMXCTR2_ACLKSEL_A2, OPORTMXCTR2_ACLKSEL_F2, OPORTMXCTR2_ACLKSEL_A2PLL, OPORTMXCTR2_ACLKSEL_RX1, }; - u32 v_div[] = { + static const u32 v_div[] = { OPORTMXCTR2_DACCKSEL_1_2, OPORTMXCTR2_DACCKSEL_1_3, OPORTMXCTR2_DACCKSEL_1_1, OPORTMXCTR2_DACCKSEL_2_3, }; -- 2.38.1

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[PATCH v2] ALSA: patch_realtek: Fix Dell Inspiron Plus 16
by Philipp Jungkamp 05 Dec '22

05 Dec '22

The Dell Inspiron Plus 16, in both laptop and 2in1 form factor, has top speakers connected on NID 0x17, which the codec reports as unconnected. These speakers should be connected to the DAC on NID 0x03. Signed-off-by: Philipp Jungkamp <p.jungkamp(a)gmx.net> --- This is a pretty late follow up to the v1 in september. Two people have tested this patch, one for each subsystem id. The popping sound on the headphone jack after audio playback stopped was gone after adding the suggested alc_fixup_no_shutup. Regards, Philipp Jungkamp v2: - add 'Signed-off-by' to commit - add subsystem of the 2-in-1 version of the laptop - add alc_fixup_no_shutup to stop headphone jack popping sound on audio suspend sound/pci/hda/patch_realtek.c | 37 +++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cf7c825078dc..5f51f8fc7fdc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6903,6 +6903,34 @@ static void alc287_fixup_yoga9_14iap7_bass_spk_pin(struct hda_codec *codec, } } +static void alc295_fixup_dell_inspiron_top_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170151 }, + { 0x17, 0x90170150 }, + { } + }; + static const hda_nid_t conn[] = { 0x02, 0x03 }; + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, + 0x17, 0x03, + 0x21, 0x02, + 0 + }; + struct alc_spec *spec = codec->spec; + + alc_fixup_no_shutup(codec, fix, action); + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + } +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -7146,6 +7174,7 @@ enum { ALC287_FIXUP_LEGION_16ITHG6, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, + ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9095,6 +9124,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, }, + [ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_dell_inspiron_top_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9195,6 +9230,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.38.1

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