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

[PATCH v6 2/7] ASoC: fsl-asoc-card: add second dai link component for codecs
by Elinor Montmasson 27 Jun '24

27 Jun '24

Add a second dai link component for codecs that will be used for use cases with 2 codecs. It is needed for future integration of the SPDIF support, which will use spdif_receiver and spdif_transmitter drivers. To prevent deferring in use cases using only one codec, also set by default the number of codecs to 1 for the relevant dai links. Co-developed-by: Philip-Dylan Gleonec <philip-dylan.gleonec(a)savoirfairelinux.com> Signed-off-by: Philip-Dylan Gleonec <philip-dylan.gleonec(a)savoirfairelinux.com> Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index e8003fbc8092..805e2030bde4 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -296,7 +296,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, SND_SOC_DAILINK_DEFS(hifi, DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY()), + DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(hifi_fe, @@ -306,7 +306,7 @@ SND_SOC_DAILINK_DEFS(hifi_fe, SND_SOC_DAILINK_DEFS(hifi_be, DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); + DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY())); static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { /* Default ASoC DAI Link*/ @@ -622,6 +622,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + priv->dai_link[0].num_codecs = 1; + priv->dai_link[2].num_codecs = 1; priv->card.dapm_routes = audio_map; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); -- 2.34.1

1 0

[PATCH v6 1/7] ASoC: fsl-asoc-card: add support for dai links with multiple codecs
by Elinor Montmasson 27 Jun '24

27 Jun '24

Add support for dai links using multiple codecs for multi-codec use cases. Co-developed-by: Philip-Dylan Gleonec <philip-dylan.gleonec(a)savoirfairelinux.com> Signed-off-by: Philip-Dylan Gleonec <philip-dylan.gleonec(a)savoirfairelinux.com> Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index eb67689dcd6e..e8003fbc8092 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -542,6 +542,7 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) static int fsl_asoc_card_probe(struct platform_device *pdev) { struct device_node *cpu_np, *codec_np, *asrc_np; + struct snd_soc_dai_link_component *codec_comp; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; struct device_node *bitclkprovider = NULL; @@ -552,6 +553,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) const char *codec_dai_name; const char *codec_dev_name; u32 asrc_fmt = 0; + int codec_idx; u32 width; int ret; @@ -816,10 +818,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpus->of_node = cpu_np; - priv->dai_link[0].codecs->dai_name = codec_dai_name; + priv->dai_link[0].codecs[0].dai_name = codec_dai_name; if (!fsl_asoc_card_is_ac97(priv)) - priv->dai_link[0].codecs->of_node = codec_np; + priv->dai_link[0].codecs[0].of_node = codec_np; else { u32 idx; @@ -830,11 +832,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto asrc_fail; } - priv->dai_link[0].codecs->name = + priv->dai_link[0].codecs[0].name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "ac97-codec.%u", (unsigned int)idx); - if (!priv->dai_link[0].codecs->name) { + if (!priv->dai_link[0].codecs[0].name) { ret = -ENOMEM; goto asrc_fail; } @@ -848,10 +850,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* DPCM DAI Links only if ASRC exists */ priv->dai_link[1].cpus->of_node = asrc_np; priv->dai_link[1].platforms->of_node = asrc_np; - priv->dai_link[2].codecs->dai_name = codec_dai_name; - priv->dai_link[2].codecs->of_node = codec_np; - priv->dai_link[2].codecs->name = - priv->dai_link[0].codecs->name; + for_each_link_codecs((&(priv->dai_link[2])), codec_idx, codec_comp) { + codec_comp->dai_name = priv->dai_link[0].codecs[codec_idx].dai_name; + codec_comp->of_node = priv->dai_link[0].codecs[codec_idx].of_node; + codec_comp->name = priv->dai_link[0].codecs[codec_idx].name; + } priv->dai_link[2].cpus->of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; -- 2.34.1

1 0

0db0:543d Micro Star International USB Audio
by GitHub issues - opened 27 Jun '24

27 Jun '24

alsa-project/alsa-ucm-conf issue #427 was opened from vividnightmare: I have MSI TRX40 Pro 10G on Arch Linux. alsa-ucm-conf 1.2.12-1 linux-zen 6.9.6.zen1-1 The sound device is reported with lsusb as `0db0:543d Micro Star International USB Audio` With default configuration, all seems well until hearing 5.1/7.1 channel audio. At this point it is realized that my 2.1 configuration is playing audio in surround mode and I'm missing virtually all of the most important sound. After researching for a while, I've found the following to hacky-workaround? the issue: ``` 90c90 < Regex "USB((0414:a0(0e|1[0124]))|(0b05:(19(84|9[69])|1a(16|2[07]|5[23c]|97|f1)))|(0db0:(005a|124b|151f|1feb|3130|36e7|4(19c|22d|240|88c)|62a4|6c[0c]9|70d3|7696|82c7|8af7|961e|a(073|228|47c|74b)|b202|d1d7|d6e7))|(26ce:0a0[68]))" --- > Regex "USB((0414:a0(0e|1[0124]))|(0b05:(19(84|9[69])|1a(16|2[07]|5[23c]|97|f1)))|(0db0:(005a|124b|151f|1feb|3130|36e7|4(19c|22d|240|88c)|543d|62a4|6c[0c]9|70d3|7696|82c7|8af7|961e|a(073|228|47c|74b)|b202|d1d7|d6e7))|(26ce:0a0[68]))" ``` This adds matching under ALC4080 for my 0db0:543d. After this I have 2.1 as well as the ability to select 5.1 or 7.1. I have patched the alsa-ucm-conf package locally with epoch to finalize the configuration. However, I do not know that this is the appropriate solution. Whether my sound device is mis-identified by ALSA/UCM, misrepresented by MSI? Something else? Please advise and perhaps this can be resolved in a way that benefits more than just myself. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/427 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] Add master clock handling for nau8824
by Maxim Kochetkov 27 Jun '24

27 Jun '24

nau8824 has external MCLK pin. So add enable/disable external clock management. Maxim Kochetkov (2): ASoC: codecs: nau8824: Add master clock handling ASoC: dt-bindings: nau8824: Add master clock handling .../bindings/sound/nuvoton,nau8824.yaml | 8 +++++++ sound/soc/codecs/nau8824.c | 21 +++++++++++++++++-- sound/soc/codecs/nau8824.h | 1 + 3 files changed, 28 insertions(+), 2 deletions(-) -- 2.45.2

3 6

[GIT PULL] ASoC fixes for v6.10-rc5
by Mark Brown 26 Jun '24

26 Jun '24

The following changes since commit c3f38fa61af77b49866b006939479069cd451173: Linux 6.10-rc2 (2024-06-02 15:44:56 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.10-rc5 for you to fetch changes up to 68f97fe330e01450ace63da0ce5cab676fc97f9a: ASoC: rt5645: fix issue of random interrupt from push-button (2024-06-26 16:34:48 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.10 A relatively large batch of updates, largely due to the long interval since I last sent fixes due to various travel and holidays. There's a lot of driver specific fixes and quirks in here, none of them too major, and also some fixes for recently introduced memory safety issues in the topology code. ---------------------------------------------------------------- Alibek Omarov (1): ASoC: rockchip: i2s-tdm: Fix trcm mode by setting clock on right mclk Amadeusz Sławiński (5): ASoC: topology: Fix references to freed memory ASoC: Intel: avs: Fix route override ASoC: topology: Do not assign fields that are already set ASoC: topology: Clean up route loading ASoC: topology: Fix route memory corruption Andrei Simion (1): ASoC: atmel: atmel-classd: Re-add dai_link->platform to fix card init Bard Liao (2): ASoC: SOF: Intel: hda-dai: skip tlv for dspless mode ASoC: SOF: Intel: hda-dai: remove skip_tlv label Chen-Yu Tsai (1): ASoC: mediatek: mt8195: Add platform entry for ETDM1_OUT_BE dai link Dmitry Baryshkov (1): MAINTAINERS: copy linux-arm-msm for sound/qcom changes Elinor Montmasson (1): ASoC: fsl-asoc-card: set priv->pdev before using it Hsin-Te Yuan (1): ASoC: mediatek: mt8183-da7219-max98357: Fix kcontrol name collision Jack Yu (3): ASoC: rt722-sdca-sdw: add silence detection register as volatile ASoC: rt722-sdca-sdw: add debounce time for type detection ASoC: rt5645: fix issue of random interrupt from push-button Jai Luthra (2): ALSA: dmaengine: Synchronize dma channel after drop() ASoC: ti: davinci-mcasp: Set min period size using FIFO config Jeff Johnson (3): ASoC: mxs: add missing MODULE_DESCRIPTION() macro ASoC: fsl: add missing MODULE_DESCRIPTION() macro ASoC: qcom: add missing MODULE_DESCRIPTION() macro Maciej Strozek (1): ASoC: cs42l43: Increase default type detect time and button delay Mark Brown (3): ASoC: SOF: Intel: hda-dai: skip tlv configuration for ASoC: topology: Fix route memory corruption Fixes for McASP and dmaengine_pcm Peter Ujfalusi (1): ASoC: SOF: sof-audio: Skip unprepare for in-use widgets on error rollback Primoz Fiser (1): ASoC: ti: omap-hdmi: Fix too long driver name Richard Fitzgerald (1): ASoC: cs35l56: Disconnect ASP1 TX sources when ASP1 DAI is hooked up Shuming Fan (1): ASoC: Intel: soc-acpi: mtl: fix speaker no sound on Dell SKU 0C64 Srinivas Kandagatla (1): ASoC: q6apm-lpass-dai: close graph on prepare errors Thomas GENTY (1): bytcr_rt5640 : inverse jack detect for Archos 101 cesium Vijendar Mukunda (3): ASoC: amd: acp: add a null check for chip_pdev structure ASoC: amd: acp: remove i2s configuration check in acp_i2s_probe() ASoC: amd: acp: move chip->flag variable assignment Vyacheslav Frantsishko (1): ASoC: amd: yc: Fix non-functional mic on ASUS M5602RA Zhang Yi (1): ASoC: codecs: ES8326: Solve headphone detection issue MAINTAINERS | 1 + include/sound/dmaengine_pcm.h | 1 + sound/core/pcm_dmaengine.c | 10 +++++++ sound/soc/amd/acp/acp-i2s.c | 8 ----- sound/soc/amd/acp/acp-pci.c | 12 ++++---- sound/soc/amd/yc/acp6x-mach.c | 7 +++++ sound/soc/atmel/atmel-classd.c | 7 +++-- sound/soc/codecs/cs35l56-shared.c | 4 +++ sound/soc/codecs/cs42l43-jack.c | 4 +-- sound/soc/codecs/es8326.c | 8 +++-- sound/soc/codecs/rt5645.c | 24 +++++++++++---- sound/soc/codecs/rt5645.h | 6 ++++ sound/soc/codecs/rt722-sdca-sdw.c | 4 ++- sound/soc/fsl/fsl-asoc-card.c | 3 +- sound/soc/fsl/imx-pcm-dma.c | 1 + sound/soc/intel/avs/topology.c | 19 +++++++++--- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++ sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 10 +++---- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 1 + sound/soc/mxs/mxs-pcm.c | 1 + sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 32 ++++++++++++-------- sound/soc/qcom/sdw.c | 1 + sound/soc/rockchip/rockchip_i2s_tdm.c | 13 ++++++-- sound/soc/soc-generic-dmaengine-pcm.c | 8 +++++ sound/soc/soc-topology.c | 35 ++++++++++------------ sound/soc/sof/intel/hda-dai.c | 6 ++-- sound/soc/sof/sof-audio.c | 2 +- sound/soc/ti/davinci-mcasp.c | 9 ++++-- sound/soc/ti/omap-hdmi.c | 6 +--- 30 files changed, 176 insertions(+), 80 deletions(-)

2 1

[PATCH] ASoC: qcom: Adjust issues in case of DT error in asoc_qcom_lpass_cpu_platform_probe()
by Aleksandr Mishin 26 Jun '24

26 Jun '24

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: b138706225c9 ("ASoC: qcom: Add regmap config support for codec dma driver") Signed-off-by: Aleksandr Mishin <amishin(a)t-argos.ru> --- sound/soc/qcom/lpass-cpu.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index b0f3e02cb043..5a47f661e0c6 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1166,9 +1166,13 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) } res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "lpass-rxtx-cdc-dma-lpm"); + if (!res) + return -EINVAL; drvdata->rxtx_cdc_dma_lpm_buf = res->start; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "lpass-va-cdc-dma-lpm"); + if (!res) + return -EINVAL; drvdata->va_cdc_dma_lpm_buf = res->start; } -- 2.30.2

2 1

[PATCH] ASoC: dt-bindings: amlogic,gx-sound-card: drop minItems for audio-widgets
by Neil Armstrong 26 Jun '24

26 Jun '24

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: Neil Armstrong <neil.armstrong(a)linaro.org> --- Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml | 1 - 1 file changed, 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index d4277d342e69..0ecdaf7190e9 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -23,7 +23,6 @@ properties: audio-widgets: $ref: /schemas/types.yaml#/definitions/non-unique-string-array - minItems: 2 description: |- A list off component DAPM widget. Each entry is a pair of strings, the first being the widget type, the second being the widget name --- base-commit: c3f38fa61af77b49866b006939479069cd451173 change-id: 20240605-topic-amlogic-upstream-bindings-fixes-audio-widgets-00e1afcda10b Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

3 4

[PATCH 1/2] ASoC: codecs: lpass-macro: Gracefully handle unknown version
by Krzysztof Kozlowski 26 Jun '24

26 Jun '24

Qualcomm LPASS macro codec driver parses registers in order to detect version of the codec. It recognizes codecs v2.0 - v2.8, however we know that there are earlier versions and 'enum lpass_codec_version' has also v1.0, v1.1 and v1.2. If by any chance we run on unrecognized version, driver will use random value from the stack as the codec version. Fix it by mapping such cases to an enum of value 0: LPASS_CODEC_VERSION_UNKNOWN. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-macro-common.h | 3 ++- sound/soc/codecs/lpass-va-macro.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-macro-common.h b/sound/soc/codecs/lpass-macro-common.h index f6f1bfe8eb77..94697d0ba8c9 100644 --- a/sound/soc/codecs/lpass-macro-common.h +++ b/sound/soc/codecs/lpass-macro-common.h @@ -19,7 +19,8 @@ enum lpass_version { }; enum lpass_codec_version { - LPASS_CODEC_VERSION_1_0 = 1, + LPASS_CODEC_VERSION_UNKNOWN, + LPASS_CODEC_VERSION_1_0, LPASS_CODEC_VERSION_1_1, LPASS_CODEC_VERSION_1_2, LPASS_CODEC_VERSION_2_0, diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index ae80865cd459..f9262d7d2c26 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1463,7 +1463,8 @@ static int va_macro_validate_dmic_sample_rate(u32 dmic_sample_rate, static void va_macro_set_lpass_codec_version(struct va_macro *va) { - int core_id_0 = 0, core_id_1 = 0, core_id_2 = 0, version; + int core_id_0 = 0, core_id_1 = 0, core_id_2 = 0; + int version = LPASS_CODEC_VERSION_UNKNOWN; regmap_read(va->regmap, CDC_VA_TOP_CSR_CORE_ID_0, &core_id_0); regmap_read(va->regmap, CDC_VA_TOP_CSR_CORE_ID_1, &core_id_1); -- 2.43.0

3 4

[PATCH] ASoC: rt5645: fix issue of random interrupt from push-button
by Jack Yu 26 Jun '24

26 Jun '24

Modify register setting sequence of enabling inline command to fix issue of random interrupt from push-button. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt5645.c | 24 ++++++++++++++++++------ sound/soc/codecs/rt5645.h | 6 ++++++ 2 files changed, 24 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index cdb7ff7020e9..51187b1e0ed2 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -81,7 +81,7 @@ static const struct reg_sequence init_list[] = { static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, {RT5645_PWR_ANLG1, 0x02}, - {RT5645_IL_CMD3, 0x0018}, + {RT5645_IL_CMD3, 0x6728}, }; static const struct reg_default rt5645_reg[] = { @@ -3130,20 +3130,32 @@ static void rt5645_enable_push_button_irq(struct snd_soc_component *component, bool enable) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + int ret; if (enable) { snd_soc_dapm_force_enable_pin(dapm, "ADC L power"); snd_soc_dapm_force_enable_pin(dapm, "ADC R power"); snd_soc_dapm_sync(dapm); + snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD2, + RT5645_EN_4BTN_IL_MASK | RT5645_RST_4BTN_IL_MASK, + RT5645_EN_4BTN_IL_EN | RT5645_RST_4BTN_IL_RST); + usleep_range(10000, 15000); + snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD2, + RT5645_EN_4BTN_IL_MASK | RT5645_RST_4BTN_IL_MASK, + RT5645_EN_4BTN_IL_EN | RT5645_RST_4BTN_IL_NORM); + msleep(50); + ret = snd_soc_component_read(component, RT5645_INT_IRQ_ST); + pr_debug("%s read %x = %x\n", __func__, RT5645_INT_IRQ_ST, + snd_soc_component_read(component, RT5645_INT_IRQ_ST)); + snd_soc_component_write(component, RT5645_INT_IRQ_ST, ret); + ret = snd_soc_component_read(component, RT5650_4BTN_IL_CMD1); + pr_debug("%s read %x = %x\n", __func__, RT5650_4BTN_IL_CMD1, + snd_soc_component_read(component, RT5650_4BTN_IL_CMD1)); + snd_soc_component_write(component, RT5650_4BTN_IL_CMD1, ret); snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD1, 0x3, 0x3); snd_soc_component_update_bits(component, RT5645_INT_IRQ_ST, 0x8, 0x8); - snd_soc_component_update_bits(component, - RT5650_4BTN_IL_CMD2, 0x8000, 0x8000); - snd_soc_component_read(component, RT5650_4BTN_IL_CMD1); - pr_debug("%s read %x = %x\n", __func__, RT5650_4BTN_IL_CMD1, - snd_soc_component_read(component, RT5650_4BTN_IL_CMD1)); } else { snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD2, 0x8000, 0x0); snd_soc_component_update_bits(component, RT5645_INT_IRQ_ST, 0x8, 0x0); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 90816b2c5489..bef74b29fd54 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2011,6 +2011,12 @@ #define RT5645_ZCD_HP_DIS (0x0 << 15) #define RT5645_ZCD_HP_EN (0x1 << 15) +/* Buttons Inline Command Function 2 (0xe0) */ +#define RT5645_EN_4BTN_IL_MASK (0x1 << 15) +#define RT5645_EN_4BTN_IL_EN (0x1 << 15) +#define RT5645_RST_4BTN_IL_MASK (0x1 << 14) +#define RT5645_RST_4BTN_IL_RST (0x0 << 14) +#define RT5645_RST_4BTN_IL_NORM (0x1 << 14) /* Codec Private Register definition */ /* DAC ADC Digital Volume (0x00) */ -- 2.34.1

2 1

[PATCH 0/3] ASoC: codecs: lpass-wsa-macro: Add support for newer v2.5 version
by Krzysztof Kozlowski 26 Jun '24

26 Jun '24

Hi, LPASS codec v2.5 and newer have differences in registers which we did not implement so far. Lack of proper support is visible during playback: on SoCs with v2.6 (e.g. Qualcomm SM8550) only one speaker plays. Add missing bits for v2.5 and newer codecs. This is similar work to already merged: https://lore.kernel.org/all/20240612-lpass-codec-v25-v4-0-f63d3676dbc4@lina… No dependencies. Best regards, Krzysztof --- Krzysztof Kozlowski (3): ASoC: codecs: lpass-wsa-macro: Drop unused define ASoC: codecs: lpass-wsa-macro: Prepare to accommodate new codec versions ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version sound/soc/codecs/lpass-wsa-macro.c | 644 ++++++++++++++++++++++++++++++------- 1 file changed, 521 insertions(+), 123 deletions(-) --- base-commit: d585065ee5b64c60496a2403bafee1540cb319ec change-id: 20240625-qcom-audio-wsa-second-speaker-ee1a5d2d267b Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 5