From 49f6202ce991742f451fc724f03d0c17460d06cd Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Sat, 3 Aug 2024 13:41:40 +0300 Subject: ASoC: codecs: lpass-macro: fix version strings returned for 1.x codecs Add missing cases to lpass_macro_get_codec_version_string() to let it print the correct codec version for 1.x codec platforms. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Signed-off-by: Dmitry Baryshkov Link: https://patch.msgid.link/20240803-codec-version-v1-1-bc29baa5e417@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-macro-common.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/lpass-macro-common.h b/sound/soc/codecs/lpass-macro-common.h index 21cb30ab706d..fb4b96cb2b23 100644 --- a/sound/soc/codecs/lpass-macro-common.h +++ b/sound/soc/codecs/lpass-macro-common.h @@ -49,6 +49,12 @@ static inline void lpass_macro_pds_exit_action(void *pds) static inline const char *lpass_macro_get_codec_version_string(int version) { switch (version) { + case LPASS_CODEC_VERSION_1_0: + return "v1.0"; + case LPASS_CODEC_VERSION_1_1: + return "v1.1"; + case LPASS_CODEC_VERSION_1_2: + return "v1.2"; case LPASS_CODEC_VERSION_2_0: return "v2.0"; case LPASS_CODEC_VERSION_2_1: -- cgit v1.2.3-58-ga151 From a9a7a2d80790d06cd32c535e2e7b10f72ce592e7 Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Sat, 3 Aug 2024 13:41:41 +0300 Subject: ASoC: codecs: lpass-va-macro: warn on unknown version Warn the users if the driver doesn't know the codec version. This helps in debugging the issues with other codec not detecting the correct version. va_macro 3370000.codec: Unknown VA Codec version, ID: 00 / 0f / 00 Signed-off-by: Dmitry Baryshkov Link: https://patch.msgid.link/20240803-codec-version-v1-2-bc29baa5e417@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-va-macro.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index a62ccd09bacd..8454193ed22a 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1485,6 +1485,10 @@ static void va_macro_set_lpass_codec_version(struct va_macro *va) if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && (core_id_2 == 0x80 || core_id_2 == 0x81)) version = LPASS_CODEC_VERSION_2_8; + if (version == LPASS_CODEC_VERSION_UNKNOWN) + dev_warn(va->dev, "Unknown Codec version, ID: %02x / %02x / %02x\n", + core_id_0, core_id_1, core_id_2); + lpass_macro_set_codec_version(version); dev_dbg(va->dev, "LPASS Codec Version %s\n", lpass_macro_get_codec_version_string(version)); -- cgit v1.2.3-58-ga151 From c56ba3e44784527fd6efe5eb7a4fa6c9f6969a58 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Tue, 13 Aug 2024 16:29:43 +0530 Subject: ASoC: SOF: amd: move iram-dram fence register programming sequence The existing code modifies IRAM and DRAM size after sha dma start for vangogh platform. The problem with this sequence is that it might cause sha dma failure when firmware code binary size is greater than the default IRAM size. To fix this issue, Move the iram-dram fence register sequence prior to sha dma start. Fixes: 094d11768f74 ("ASoC: SOF: amd: Skip IRAM/DRAM size modification for Steam Deck OLED") Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20240813105944.3126903-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 74fd5f2b148b..9123427fab4e 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -263,6 +263,17 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_STRT_ADDR, start_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_DESTINATION_ADDR, dest_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_MSG_LENGTH, image_length); + + /* psp_send_cmd only required for vangogh platform (rev - 5) */ + if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { + /* Modify IRAM and DRAM size */ + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); + if (ret) + return ret; + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); + if (ret) + return ret; + } snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_CMD, ACP_SHA_RUN); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_TRANSFER_BYTE_CNT, @@ -280,17 +291,6 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { - /* Modify IRAM and DRAM size */ - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); - if (ret) - return ret; - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); - if (ret) - return ret; - } - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_DSP_FW_QUALIFIER, fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); -- cgit v1.2.3-58-ga151 From 897e91e995b338002b00454fd0018af26a098148 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Tue, 13 Aug 2024 16:29:44 +0530 Subject: ASoC: SOF: amd: Fix for incorrect acp error register offsets Addition of 'dsp_intr_base' to ACP error register offsets points to wrong register offsets in irq handler. Correct the acp error register offsets. ACP error status register offset and acp error reason register offset got changed from ACP6.0 onwards. Add 'acp_error_stat' and 'acp_sw0_i2s_err_reason' as descriptor fields in sof_amd_acp_desc structure and update the values based on the ACP variant. >From Rembrandt platform onwards, errors related to SW1 Soundwire manager instance/I2S controller connected on P1 power tile is reported with ACP_SW1_I2S_ERROR_REASON register. Add conditional check for the same. Fixes: 96eb81851012 ("ASoC: SOF: amd: add interrupt handling for SoundWire manager devices") Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20240813105944.3126903-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-dsp-offset.h | 6 ++++-- sound/soc/sof/amd/acp.c | 11 +++++++---- sound/soc/sof/amd/acp.h | 2 ++ sound/soc/sof/amd/pci-acp63.c | 2 ++ sound/soc/sof/amd/pci-rmb.c | 2 ++ sound/soc/sof/amd/pci-rn.c | 2 ++ 6 files changed, 19 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index 59afbe2e0f42..072b703f9b3f 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -76,13 +76,15 @@ #define DSP_SW_INTR_CNTL_OFFSET 0x0 #define DSP_SW_INTR_STAT_OFFSET 0x4 #define DSP_SW_INTR_TRIG_OFFSET 0x8 -#define ACP_ERROR_STATUS 0x18C4 +#define ACP3X_ERROR_STATUS 0x18C4 +#define ACP6X_ERROR_STATUS 0x1A4C #define ACP3X_AXI2DAGB_SEM_0 0x1880 #define ACP5X_AXI2DAGB_SEM_0 0x1884 #define ACP6X_AXI2DAGB_SEM_0 0x1874 /* ACP common registers to report errors related to I2S & SoundWire interfaces */ -#define ACP_SW0_I2S_ERROR_REASON 0x18B4 +#define ACP3X_SW_I2S_ERROR_REASON 0x18C8 +#define ACP6X_SW0_I2S_ERROR_REASON 0x18B4 #define ACP_SW1_I2S_ERROR_REASON 0x1A50 /* Registers from ACP_SHA block */ diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 9123427fab4e..d95f865669a6 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -92,6 +92,7 @@ static int config_dma_channel(struct acp_dev_data *adata, unsigned int ch, unsigned int idx, unsigned int dscr_count) { struct snd_sof_dev *sdev = adata->dev; + const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int val, status; int ret; @@ -102,7 +103,7 @@ static int config_dma_channel(struct acp_dev_data *adata, unsigned int ch, val & (1 << ch), ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); if (ret < 0) { - status = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_ERROR_STATUS); + status = snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->acp_error_stat); val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_DMA_ERR_STS_0 + ch * sizeof(u32)); dev_err(sdev->dev, "ACP_DMA_ERR_STS :0x%x ACP_ERROR_STATUS :0x%x\n", val, status); @@ -402,9 +403,11 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) if (val & ACP_ERROR_IRQ_MASK) { snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, ACP_ERROR_IRQ_MASK); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_SW0_I2S_ERROR_REASON, 0); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_SW1_I2S_ERROR_REASON, 0); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_ERROR_STATUS, 0); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_sw0_i2s_err_reason, 0); + /* ACP_SW1_I2S_ERROR_REASON is newly added register from rmb platform onwards */ + if (desc->rev >= 6) + snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SW1_I2S_ERROR_REASON, 0); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_error_stat, 0); irq_flag = 1; } diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 87e79d500865..1af86b5b28db 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -203,6 +203,8 @@ struct sof_amd_acp_desc { u32 probe_reg_offset; u32 reg_start_addr; u32 reg_end_addr; + u32 acp_error_stat; + u32 acp_sw0_i2s_err_reason; u32 sdw_max_link_count; u64 sdw_acpi_dev_addr; }; diff --git a/sound/soc/sof/amd/pci-acp63.c b/sound/soc/sof/amd/pci-acp63.c index fc8984447365..986f5928caed 100644 --- a/sound/soc/sof/amd/pci-acp63.c +++ b/sound/soc/sof/amd/pci-acp63.c @@ -35,6 +35,8 @@ static const struct sof_amd_acp_desc acp63_chip_info = { .ext_intr_cntl = ACP6X_EXTERNAL_INTR_CNTL, .ext_intr_stat = ACP6X_EXT_INTR_STAT, .ext_intr_stat1 = ACP6X_EXT_INTR_STAT1, + .acp_error_stat = ACP6X_ERROR_STATUS, + .acp_sw0_i2s_err_reason = ACP6X_SW0_I2S_ERROR_REASON, .dsp_intr_base = ACP6X_DSP_SW_INTR_BASE, .sram_pte_offset = ACP6X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0, diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index 4bc30951f8b0..a366f904e6f3 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -33,6 +33,8 @@ static const struct sof_amd_acp_desc rembrandt_chip_info = { .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_stat = ACP6X_EXT_INTR_STAT, .dsp_intr_base = ACP6X_DSP_SW_INTR_BASE, + .acp_error_stat = ACP6X_ERROR_STATUS, + .acp_sw0_i2s_err_reason = ACP6X_SW0_I2S_ERROR_REASON, .sram_pte_offset = ACP6X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0, .fusion_dsp_offset = ACP6X_DSP_FUSION_RUNSTALL, diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index e08875bdfa8b..2b7c53470ce8 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -33,6 +33,8 @@ static const struct sof_amd_acp_desc renoir_chip_info = { .pgfsm_base = ACP3X_PGFSM_BASE, .ext_intr_stat = ACP3X_EXT_INTR_STAT, .dsp_intr_base = ACP3X_DSP_SW_INTR_BASE, + .acp_error_stat = ACP3X_ERROR_STATUS, + .acp_sw0_i2s_err_reason = ACP3X_SW_I2S_ERROR_REASON, .sram_pte_offset = ACP3X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP3X_AXI2DAGB_SEM_0, .acp_clkmux_sel = ACP3X_CLKMUX_SEL, -- cgit v1.2.3-58-ga151 From b919a27fab37e108164d657ac6e77bf870bf95e6 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Jul 2024 12:35:11 +0200 Subject: ASoC: MAINTAINERS: Drop Banajit Goswami from Qualcomm sound drivers There was no active maintenance from Banajit Goswami - last email is from 2019 - so make obvious that Qualcomm sound drivers are maintained by only one person. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240730103511.21728-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index d304054d661e..61a21efc357b 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -18522,7 +18522,6 @@ F: drivers/crypto/intel/qat/ QCOM AUDIO (ASoC) DRIVERS M: Srinivas Kandagatla -M: Banajit Goswami L: alsa-devel@alsa-project.org (moderated for non-subscribers) L: linux-arm-msm@vger.kernel.org S: Supported -- cgit v1.2.3-58-ga151 From c0196faaa927321a63e680427e075734ee656e42 Mon Sep 17 00:00:00 2001 From: Albert Jakieła Date: Fri, 9 Aug 2024 13:56:27 +0000 Subject: ASoC: SOF: mediatek: Add missing board compatible MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add Google Dojo compatible. Signed-off-by: Albert Jakieła Reviewed-by: Chen-Yu Tsai Link: https://patch.msgid.link/20240809135627.544429-1-jakiela@google.com Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8195/mt8195.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 1c6e035fd313..82d221f53a46 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -574,6 +574,9 @@ static struct snd_sof_of_mach sof_mt8195_machs[] = { { .compatible = "google,tomato", .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682.tplg" + }, { + .compatible = "google,dojo", + .sof_tplg_filename = "sof-mt8195-mt6359-max98390-rt5682.tplg" }, { .compatible = "mediatek,mt8195", .sof_tplg_filename = "sof-mt8195.tplg" -- cgit v1.2.3-58-ga151 From 2251db28edcc70b7ee8a8c6bcbaecf752b3ea5ec Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 6 Aug 2024 13:49:13 +0200 Subject: ASoC: codecs: wcd937x: Fix missing de-assert of reset GPIO The device never comes online from a reset/shutdown state, because the driver de-asserts reset GPIO when requesting it but then, at the end of probe() through wcd937x_reset(), leaves it asserted. Cc: stable@vger.kernel.org Fixes: 9be3ec196da4 ("ASoC: codecs: wcd937x: add wcd937x codec driver") Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240806114913.40022-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd937x.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index 13926f4b0d9f..af296b77a723 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -242,10 +242,9 @@ static const struct regmap_irq_chip wcd937x_regmap_irq_chip = { static void wcd937x_reset(struct wcd937x_priv *wcd937x) { - usleep_range(20, 30); - gpiod_set_value(wcd937x->reset_gpio, 1); - + usleep_range(20, 30); + gpiod_set_value(wcd937x->reset_gpio, 0); usleep_range(20, 30); } -- cgit v1.2.3-58-ga151 From ff9f065318e17a1a97981d9e535fcfc6ce5d5614 Mon Sep 17 00:00:00 2001 From: YR Yang Date: Thu, 1 Aug 2024 16:43:26 +0800 Subject: ASoC: mediatek: mt8188: Mark AFE_DAC_CON0 register as volatile Add AFE Control Register 0 to the volatile_register. AFE_DAC_CON0 can be modified by both the SOF and ALSA drivers. If this register is read and written in cache mode, the cached value might not reflect the actual value when the register is modified by another driver. It can cause playback or capture failures. Therefore, it is necessary to add AFE_DAC_CON0 to the list of volatile registers. Signed-off-by: YR Yang Reviewed-by: Fei Shao Reviewed-by: Trevor Wu Link: https://patch.msgid.link/20240801084326.1472-1-yr.yang@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c index ccb6c1f3adc7..73e5c63aeec8 100644 --- a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c +++ b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c @@ -2748,6 +2748,7 @@ static bool mt8188_is_volatile_reg(struct device *dev, unsigned int reg) case AFE_ASRC12_NEW_CON9: case AFE_LRCK_CNT: case AFE_DAC_MON0: + case AFE_DAC_CON0: case AFE_DL2_CUR: case AFE_DL3_CUR: case AFE_DL6_CUR: -- cgit v1.2.3-58-ga151 From 164199615ae230ace4519141285f06766d6d8036 Mon Sep 17 00:00:00 2001 From: Yuntao Liu Date: Thu, 15 Aug 2024 08:49:23 +0000 Subject: ASoC: amd: acp: fix module autoloading Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from platform_device_id table. Fixes: 9d8a7be88b336 ("ASoC: amd: acp: Add legacy sound card support for Chrome audio") Signed-off-by: Yuntao Liu Link: https://patch.msgid.link/20240815084923.756476-1-liuyuntao12@huawei.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-legacy-mach.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index 47c3b5f167f5..0d529e32e552 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -227,6 +227,8 @@ static const struct platform_device_id board_ids[] = { }, { } }; +MODULE_DEVICE_TABLE(platform, board_ids); + static struct platform_driver acp_asoc_audio = { .driver = { .pm = &snd_soc_pm_ops, -- cgit v1.2.3-58-ga151 From a42db293e5983aa1508d12644f23d73f0553b32c Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 16 Aug 2024 12:33:28 +0530 Subject: ASoC: SOF: amd: Fix for acp init sequence When ACP is not powered on by default, acp power on sequence explicitly invoked by programming pgfsm control mask. The existing implementation checks the same PGFSM status mask and programs the same PGFSM control mask in all ACP variants which breaks acp power on sequence for ACP6.0 and ACP6.3 variants. So to fix this issue, update ACP pgfsm control mask and status mask based on acp descriptor rev field, which will vary based on acp variant. Fixes: 846aef1d7cc0 ("ASoC: SOF: amd: Add Renoir ACP HW support") Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20240816070328.610360-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 19 +++++++++++++++++-- sound/soc/sof/amd/acp.h | 7 +++++-- 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index d95f865669a6..85b58c8ccd0d 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -433,6 +433,7 @@ static int acp_power_on(struct snd_sof_dev *sdev) const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int base = desc->pgfsm_base; unsigned int val; + unsigned int acp_pgfsm_status_mask, acp_pgfsm_cntl_mask; int ret; val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, base + PGFSM_STATUS_OFFSET); @@ -440,9 +441,23 @@ static int acp_power_on(struct snd_sof_dev *sdev) if (val == ACP_POWERED_ON) return 0; - if (val & ACP_PGFSM_STATUS_MASK) + switch (desc->rev) { + case 3: + case 5: + acp_pgfsm_status_mask = ACP3X_PGFSM_STATUS_MASK; + acp_pgfsm_cntl_mask = ACP3X_PGFSM_CNTL_POWER_ON_MASK; + break; + case 6: + acp_pgfsm_status_mask = ACP6X_PGFSM_STATUS_MASK; + acp_pgfsm_cntl_mask = ACP6X_PGFSM_CNTL_POWER_ON_MASK; + break; + default: + return -EINVAL; + } + + if (val & acp_pgfsm_status_mask) snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + PGFSM_CONTROL_OFFSET, - ACP_PGFSM_CNTL_POWER_ON_MASK); + acp_pgfsm_cntl_mask); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, base + PGFSM_STATUS_OFFSET, val, !val, ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 1af86b5b28db..61b28df8c908 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -25,8 +25,11 @@ #define ACP_REG_POLL_TIMEOUT_US 2000 #define ACP_DMA_COMPLETE_TIMEOUT_US 5000 -#define ACP_PGFSM_CNTL_POWER_ON_MASK 0x01 -#define ACP_PGFSM_STATUS_MASK 0x03 +#define ACP3X_PGFSM_CNTL_POWER_ON_MASK 0x01 +#define ACP3X_PGFSM_STATUS_MASK 0x03 +#define ACP6X_PGFSM_CNTL_POWER_ON_MASK 0x07 +#define ACP6X_PGFSM_STATUS_MASK 0x0F + #define ACP_POWERED_ON 0x00 #define ACP_ASSERT_RESET 0x01 #define ACP_RELEASE_RESET 0x00 -- cgit v1.2.3-58-ga151 From 56314c0d78d6f5a60c8804c517167991a879e14a Mon Sep 17 00:00:00 2001 From: John Sweeney Date: Sun, 18 Aug 2024 11:30:15 -0400 Subject: ALSA: hda/realtek: Enable mute/micmute LEDs on HP Laptop 14-ey0xxx HP Pavilion Plus 14-ey0xxx needs existing quirk ALC245_FIXUP_HP_X360_MUTE_LEDS to enable its mute/micmute LEDs. Signed-off-by: John Sweeney Cc: Link: https://patch.msgid.link/E1sfhrD-0007TA-HC@rmmprod05.runbox Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d022a25635f9..4eafbcb40120 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10315,6 +10315,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8c15, "HP Spectre x360 2-in-1 Laptop 14-eu0xxx", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8c21, "HP Pavilion Plus Laptop 14-ey0XXX", ALC245_FIXUP_HP_X360_MUTE_LEDS), SND_PCI_QUIRK(0x103c, 0x8c46, "HP EliteBook 830 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c47, "HP EliteBook 840 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c48, "HP EliteBook 860 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), -- cgit v1.2.3-58-ga151 From 32108c22ac619c32dd6db594319e259b63bfb387 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 19 Aug 2024 10:41:53 +0200 Subject: ALSA: seq: Skip event type filtering for UMP events UMP events don't use the event type field, hence it's invalid to apply the filter, which may drop the events unexpectedly. Skip the event filtering for UMP events, instead. Fixes: 46397622a3fa ("ALSA: seq: Add UMP support") Cc: Link: https://patch.msgid.link/20240819084156.10286-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 8c4ee5066afe..6be548baa6df 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -537,6 +537,9 @@ static struct snd_seq_client *get_event_dest_client(struct snd_seq_event *event, return NULL; if (! dest->accept_input) goto __not_avail; + if (snd_seq_ev_is_ump(event)) + return dest; /* ok - no filter checks */ + if ((dest->filter & SNDRV_SEQ_FILTER_USE_EVENT) && ! test_bit(event->type, dest->event_filter)) goto __not_avail; -- cgit v1.2.3-58-ga151 From 91191a6e50a2ff752da244493171037663536768 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Wed, 21 Aug 2024 12:47:11 +0000 Subject: ALSA: hda: cs35l56: Don't use the device index as a calibration index The HDA driver cannot assume that the order that the devices are specified in the cirrus,dev-index matches the order of calibration entries. Only a calibration entry with a matching silicon id will be used. Fixes: cfa43aaa7948 ("ALSA: hda: cs35l56: Apply amp calibration from EFI data") Signed-off-by: Simon Trimmer Link: https://patch.msgid.link/20240821124711.44325-1-simont@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l56_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index a9dfd62637cf..e3ac0e23ae32 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -1003,7 +1003,7 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) goto err; } - cs35l56->base.cal_index = cs35l56->index; + cs35l56->base.cal_index = -1; cs35l56_init_cs_dsp(&cs35l56->base, &cs35l56->cs_dsp); cs35l56->cs_dsp.client_ops = &cs35l56_hda_client_ops; -- cgit v1.2.3-58-ga151 From 0e9fdab1e8df490354562187cdbb8dec643eae2c Mon Sep 17 00:00:00 2001 From: Hongbo Li Date: Wed, 21 Aug 2024 14:19:54 +0800 Subject: ASoC: allow module autoloading for table db1200_pids Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from platform_device_id table. Signed-off-by: Hongbo Li Link: https://patch.msgid.link/20240821061955.2273782-2-lihongbo22@huawei.com Signed-off-by: Mark Brown --- sound/soc/au1x/db1200.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 83a75a38705b..81abe2e18402 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -44,6 +44,7 @@ static const struct platform_device_id db1200_pids[] = { }, {}, }; +MODULE_DEVICE_TABLE(platform, db1200_pids); /*------------------------- AC97 PART ---------------------------*/ -- cgit v1.2.3-58-ga151 From 5f7c98b7519a3a847d9182bd99d57ea250032ca1 Mon Sep 17 00:00:00 2001 From: Hongbo Li Date: Wed, 21 Aug 2024 14:19:55 +0800 Subject: ASoC: allow module autoloading for table board_ids Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from platform_device_id table. Signed-off-by: Hongbo Li Link: https://patch.msgid.link/20240821061955.2273782-3-lihongbo22@huawei.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sof-mach.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index fc59ea34e687..b3a702dcd991 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -158,6 +158,8 @@ static const struct platform_device_id board_ids[] = { }, { } }; +MODULE_DEVICE_TABLE(platform, board_ids); + static struct platform_driver acp_asoc_audio = { .driver = { .name = "sof_mach", -- cgit v1.2.3-58-ga151 From 9b82ff1362f50914c8292902e07be98a9f59d33d Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 22 Aug 2024 10:54:19 +0800 Subject: ALSA: hda/realtek - Fixed ALC256 headphone no sound Dell platform, plug headphone or headset, it had a chance to get no sound from headphone. Replace depop procedure will solve this issue. Signed-off-by: Kailang Yang Link: https://lore.kernel.org/bb8e2de30d294dc287944efa0667685a@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 50 ++++++++++++++++++++++++++++++++----------- 1 file changed, 37 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4eafbcb40120..bf098c6fedb5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4930,6 +4930,30 @@ static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, } } +static void alc_hp_mute_disable(struct hda_codec *codec, unsigned int delay) +{ + if (delay <= 0) + delay = 75; + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + msleep(delay); + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + msleep(delay); +} + +static void alc_hp_enable_unmute(struct hda_codec *codec, unsigned int delay) +{ + if (delay <= 0) + delay = 75; + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + msleep(delay); + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + msleep(delay); +} + static const struct coef_fw alc225_pre_hsmode[] = { UPDATE_COEF(0x4a, 1<<8, 0), UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), @@ -5031,6 +5055,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0256: case 0x19e58326: + alc_hp_mute_disable(codec, 75); alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5302,6 +5327,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); alc_process_coef_fw(codec, coef0256); + alc_hp_enable_unmute(codec, 75); break; case 0x10ec0234: case 0x10ec0274: @@ -5399,6 +5425,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0256: case 0x19e58326: alc_process_coef_fw(codec, coef0256); + alc_hp_enable_unmute(codec, 75); break; case 0x10ec0234: case 0x10ec0274: @@ -5514,6 +5541,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0256: case 0x19e58326: alc_process_coef_fw(codec, coef0256); + alc_hp_enable_unmute(codec, 75); break; case 0x10ec0234: case 0x10ec0274: @@ -5619,25 +5647,21 @@ static void alc_determine_headset_type(struct hda_codec *codec) alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - msleep(80); - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); - alc_process_coef_fw(codec, coef0255); msleep(300); val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x0070) == 0x0070; - + if (!is_ctia) { + alc_write_coef_idx(codec, 0x45, 0xe089); + msleep(100); + val = alc_read_coef_idx(codec, 0x46); + if ((val & 0x0070) == 0x0070) + is_ctia = false; + else + is_ctia = true; + } alc_write_coefex_idx(codec, 0x57, 0x3, 0x0da3); alc_update_coefex_idx(codec, 0x57, 0x5, 1<<14, 0); - - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - msleep(80); - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); break; case 0x10ec0234: case 0x10ec0274: -- cgit v1.2.3-58-ga151 From 1fa7b099d60ad64f559bd3b8e3f0d94b2e015514 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 22 Aug 2024 16:46:56 +0800 Subject: ALSA: hda/realtek - FIxed ALC285 headphone no sound Dell platform with ALC215 ALC285 ALC289 ALC225 ALC295 ALC299, plug headphone or headset. It had a chance to get no sound from headphone. Replace depop procedure will solve this issue. Signed-off-by: Kailang Yang Link: https://lore.kernel.org/d0de1b03fd174520945dde216d765223@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 26 ++++++++++++++------------ 1 file changed, 14 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bf098c6fedb5..b5cc3417138c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5090,6 +5090,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: + alc_hp_mute_disable(codec, 75); alc_process_coef_fw(codec, alc225_pre_hsmode); alc_process_coef_fw(codec, coef0225); break; @@ -5315,6 +5316,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0299: alc_process_coef_fw(codec, alc225_pre_hsmode); alc_process_coef_fw(codec, coef0225); + alc_hp_enable_unmute(codec, 75); break; case 0x10ec0255: alc_process_coef_fw(codec, coef0255); @@ -5474,6 +5476,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) alc_process_coef_fw(codec, coef0225_2); else alc_process_coef_fw(codec, coef0225_1); + alc_hp_enable_unmute(codec, 75); break; case 0x10ec0867: alc_update_coefex_idx(codec, 0x57, 0x5, 1<<14, 0); @@ -5579,6 +5582,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0289: case 0x10ec0299: alc_process_coef_fw(codec, coef0225); + alc_hp_enable_unmute(codec, 75); break; } codec_dbg(codec, "Headset jack set to Nokia-style headset mode.\n"); @@ -5738,12 +5742,6 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - msleep(80); - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); - alc_process_coef_fw(codec, alc225_pre_hsmode); alc_update_coef_idx(codec, 0x67, 0xf000, 0x1000); val = alc_read_coef_idx(codec, 0x45); @@ -5760,15 +5758,19 @@ static void alc_determine_headset_type(struct hda_codec *codec) val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x00f0) == 0x00f0; } + if (!is_ctia) { + alc_update_coef_idx(codec, 0x45, 0x3f<<10, 0x38<<10); + alc_update_coef_idx(codec, 0x49, 3<<8, 1<<8); + msleep(100); + val = alc_read_coef_idx(codec, 0x46); + if ((val & 0x00f0) == 0x00f0) + is_ctia = false; + else + is_ctia = true; + } alc_update_coef_idx(codec, 0x4a, 7<<6, 7<<6); alc_update_coef_idx(codec, 0x4a, 3<<4, 3<<4); alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); - - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - msleep(80); - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); break; case 0x10ec0867: is_ctia = true; -- cgit v1.2.3-58-ga151 From bff980d8d9ca537fd5f3c0e9a99876c1e3713e81 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 22 Aug 2024 12:57:25 +0100 Subject: ASoC: cs-amp-lib-test: Force test calibration blob entries to be valid For a normal calibration blob the calTarget values must be non-zero and unique, and the calTime values must be non-zero. Don't rely on get_random_bytes() to be random enough to guarantee this. Force the calTarget and calTime values to be valid while retaining randomness in the values. Signed-off-by: Richard Fitzgerald Fixes: 177862317a98 ("ASoC: cs-amp-lib: Add KUnit test for calibration helpers") Link: https://patch.msgid.link/20240822115725.259568-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs-amp-lib-test.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/cs-amp-lib-test.c b/sound/soc/codecs/cs-amp-lib-test.c index 15f991b2e16e..8169ec88a8ba 100644 --- a/sound/soc/codecs/cs-amp-lib-test.c +++ b/sound/soc/codecs/cs-amp-lib-test.c @@ -38,6 +38,7 @@ static void cs_amp_lib_test_init_dummy_cal_blob(struct kunit *test, int num_amps { struct cs_amp_lib_test_priv *priv = test->priv; unsigned int blob_size; + int i; blob_size = offsetof(struct cirrus_amp_efi_data, data) + sizeof(struct cirrus_amp_cal_data) * num_amps; @@ -49,6 +50,14 @@ static void cs_amp_lib_test_init_dummy_cal_blob(struct kunit *test, int num_amps priv->cal_blob->count = num_amps; get_random_bytes(priv->cal_blob->data, sizeof(struct cirrus_amp_cal_data) * num_amps); + + /* Ensure all timestamps are non-zero to mark the entry valid. */ + for (i = 0; i < num_amps; i++) + priv->cal_blob->data[i].calTime[0] |= 1; + + /* Ensure that all UIDs are non-zero and unique. */ + for (i = 0; i < num_amps; i++) + *(u8 *)&priv->cal_blob->data[i].calTarget[0] = i + 1; } static u64 cs_amp_lib_test_get_target_uid(struct kunit *test) -- cgit v1.2.3-58-ga151 From bb4485562f5907708f1c218b5d70dce04165d1e1 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 22 Aug 2024 14:35:44 +0100 Subject: ASoC: cs-amp-lib: Ignore empty UEFI calibration entries If the timestamp of a calibration entry is 0 it is an unused entry and must be ignored. Some end-products reserve EFI space for calibration entries by shipping with a zero-filled EFI file. When searching the file for calibration data the driver must skip the empty entries. The timestamp of a valid entry is always non-zero. Signed-off-by: Richard Fitzgerald Fixes: 1cad8725f2b9 ("ASoC: cs-amp-lib: Add helpers for factory calibration data") Link: https://patch.msgid.link/20240822133544.304421-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs-amp-lib.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c index 605964af8afa..51b128c80671 100644 --- a/sound/soc/codecs/cs-amp-lib.c +++ b/sound/soc/codecs/cs-amp-lib.c @@ -182,6 +182,10 @@ static int _cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, for (i = 0; i < efi_data->count; ++i) { u64 cal_target = cs_amp_cal_target_u64(&efi_data->data[i]); + /* Skip empty entries */ + if (!efi_data->data[i].calTime[0] && !efi_data->data[i].calTime[1]) + continue; + /* Skip entries with unpopulated silicon ID */ if (cal_target == 0) continue; @@ -193,7 +197,8 @@ static int _cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, } } - if (!cal && (amp_index >= 0) && (amp_index < efi_data->count)) { + if (!cal && (amp_index >= 0) && (amp_index < efi_data->count) && + (efi_data->data[amp_index].calTime[0] || efi_data->data[amp_index].calTime[1])) { u64 cal_target = cs_amp_cal_target_u64(&efi_data->data[amp_index]); /* -- cgit v1.2.3-58-ga151 From 44ceabdec12f4e5938f5668c5a691aa3aac703d7 Mon Sep 17 00:00:00 2001 From: YOUNGJIN JOO Date: Sun, 25 Aug 2024 18:25:15 +0900 Subject: ALSA: hda/realtek: Fix the speaker output on Samsung Galaxy Book3 Ultra 144d:c1cc requires the same workaround to enable the speaker amp as other Samsung models with the ALC298 codec. Signed-off-by: YOUNGJIN JOO Cc: Link: https://patch.msgid.link/20240825092515.28728-1-neoelec@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b5cc3417138c..c04eac6a5064 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10540,6 +10540,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc868, "Samsung Galaxy Book2 Pro (NP930XED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc1ca, "Samsung Galaxy Book3 Pro 360 (NP960QFG-KB1US)", ALC298_FIXUP_SAMSUNG_AMP2), + SND_PCI_QUIRK(0x144d, 0xc1cc, "Samsung Galaxy Book3 Ultra (NT960XFH-XD92G))", ALC298_FIXUP_SAMSUNG_AMP2), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), -- cgit v1.2.3-58-ga151 From 2dc43c5e212036458ed7c5586fb82ee183fee504 Mon Sep 17 00:00:00 2001 From: Hendrik Borghorst Date: Sun, 25 Aug 2024 19:43:47 +0200 Subject: ALSA: hda/realtek: support HP Pavilion Aero 13-bg0xxx Mute LED This patch adds the HP Pavilion Aero 13 (13-bg0xxx) (year 2024) to list of quirks for keyboard LED mute indication. The laptop has two LEDs (one for speaker and one for mic mute). The pre-existing quirk ALC245_FIXUP_HP_X360_MUTE_LEDS chains both the quirk for mic and speaker mute. Tested on 6.11.0-rc4 with the aforementioned laptop. Signed-off-by: Hendrik Borghorst Cc: Link: https://patch.msgid.link/20240825174351.5687-1-hendrikborghorst@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c04eac6a5064..588738ce7380 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10380,6 +10380,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8ca2, "HP ZBook Power", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ca4, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ca7, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8cbd, "HP Pavilion Aero Laptop 13-bg0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS), SND_PCI_QUIRK(0x103c, 0x8cdd, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8cde, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8cdf, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), -- cgit v1.2.3-58-ga151 From 28b329f431cef840fddd9a9b493bc3eff1aa06c0 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 26 Aug 2024 10:49:40 +0100 Subject: ALSA: hda: hda_component: Fix mutex crash if nothing ever binds Move the initialization of parent->mutex into hda_component_manager_init() so that it is always valid. In hda_component_manager_bind() do not clear the parent information. Only zero-fill the per-component data ready for it to be filled in by the components as they bind. Previously parent->mutex was being initialized only in hda_component_manager_bind(). This meant that it was only initialized if all components appeared and there was a bind callback. If there wasn't a bind the mutex object was not valid when the Realtek driver called any of the other functions. Signed-off-by: Richard Fitzgerald Fixes: 047b9cbbaa8e ("ALSA: hda: hda_component: Protect shared data with a mutex") Link: https://patch.msgid.link/20240826094940.45563-1-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_component.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_component.c b/sound/pci/hda/hda_component.c index 7b19cb38b4e0..b7dfdb10d156 100644 --- a/sound/pci/hda/hda_component.c +++ b/sound/pci/hda/hda_component.c @@ -141,8 +141,7 @@ int hda_component_manager_bind(struct hda_codec *cdc, int ret; /* Init shared and component specific data */ - memset(parent, 0, sizeof(*parent)); - mutex_init(&parent->mutex); + memset(parent->comps, 0, sizeof(parent->comps)); parent->codec = cdc; mutex_lock(&parent->mutex); @@ -164,6 +163,8 @@ int hda_component_manager_init(struct hda_codec *cdc, struct hda_scodec_match *sm; int ret, i; + mutex_init(&parent->mutex); + for (i = 0; i < count; i++) { sm = devm_kmalloc(dev, sizeof(*sm), GFP_KERNEL); if (!sm) -- cgit v1.2.3-58-ga151