From a8c9a453387640dbe45761970f41301a6985e7fa Mon Sep 17 00:00:00 2001 From: Nikita Zhandarovich Date: Thu, 16 Jan 2025 06:24:36 -0800 Subject: [PATCH 01/41] ASoC: fsl_micfil: Enable default case in micfil_set_quality() If 'micfil->quality' received from micfil_quality_set() somehow ends up with an unpredictable value, switch() operator will fail to initialize local variable qsel before regmap_update_bits() tries to utilize it. While it is unlikely, play it safe and enable a default case that returns -EINVAL error. Found by Linux Verification Center (linuxtesting.org) with static analysis tool SVACE. Fixes: bea1d61d5892 ("ASoC: fsl_micfil: rework quality setting") Cc: stable@vger.kernel.org Signed-off-by: Nikita Zhandarovich Link: https://patch.msgid.link/20250116142436.22389-1-n.zhandarovich@fintech.ru Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 1075598a6647..fa4136683392 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -183,6 +183,8 @@ static int micfil_set_quality(struct fsl_micfil *micfil) case QUALITY_VLOW2: qsel = MICFIL_QSEL_VLOW2_QUALITY; break; + default: + return -EINVAL; } return regmap_update_bits(micfil->regmap, REG_MICFIL_CTRL2, From fc016ef7da64fd473d73ee6c261ba1b0b47afe2b Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 4 Feb 2025 13:39:41 +0800 Subject: [PATCH 02/41] ASoC: Intel: sof_sdw: Add lookup of quirk using PCI subsystem ID Add lookup of PCI subsystem vendor:device ID to find a quirk. The subsystem ID (SSID) is part of the PCI specification to uniquely identify a particular system-specific implementation of a hardware device. Unlike DMI information, it identifies the sound hardware itself, rather than a specific model of PC. SSID can be more reliable and stable than DMI strings, and is preferred by some vendors as the way to identify the actual sound hardware. Signed-off-by: Richard Fitzgerald Reviewed-by: Liam Girdwood Signed-off-by: Bard Liao Link: https://patch.msgid.link/20250204053943.93596-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index b0d35fda7b17..381fae5943fe 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include "sof_sdw_common.h" #include "../../codecs/rt711.h" @@ -751,6 +752,22 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { {} }; +static const struct snd_pci_quirk sof_sdw_ssid_quirk_table[] = { + {} +}; + +static void sof_sdw_check_ssid_quirk(const struct snd_soc_acpi_mach *mach) +{ + const struct snd_pci_quirk *quirk_entry; + + quirk_entry = snd_pci_quirk_lookup_id(mach->mach_params.subsystem_vendor, + mach->mach_params.subsystem_device, + sof_sdw_ssid_quirk_table); + + if (quirk_entry) + sof_sdw_quirk = quirk_entry->value; +} + static struct snd_soc_dai_link_component platform_component[] = { { /* name might be overridden during probe */ @@ -1278,6 +1295,13 @@ static int mc_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, ctx); + if (mach->mach_params.subsystem_id_set) { + snd_soc_card_set_pci_ssid(card, + mach->mach_params.subsystem_vendor, + mach->mach_params.subsystem_device); + sof_sdw_check_ssid_quirk(mach); + } + dmi_check_system(sof_sdw_quirk_table); if (quirk_override != -1) { @@ -1293,12 +1317,6 @@ static int mc_probe(struct platform_device *pdev) for (i = 0; i < ctx->codec_info_list_count; i++) codec_info_list[i].amp_num = 0; - if (mach->mach_params.subsystem_id_set) { - snd_soc_card_set_pci_ssid(card, - mach->mach_params.subsystem_vendor, - mach->mach_params.subsystem_device); - } - ret = sof_card_dai_links_create(card); if (ret < 0) return ret; From 0843449708085c4fb45a3c325c2fbced556f6abf Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 4 Feb 2025 13:39:42 +0800 Subject: [PATCH 03/41] ASoC: Intel: sof_sdw: Add quirk for Asus Zenbook S14 Asus laptops with sound PCI subsystem ID 1043:1e13 have the DMICs connected to the host instead of the CS42L43 so need the SOC_SDW_CODEC_MIC quirk. Signed-off-by: Richard Fitzgerald Reviewed-by: Liam Girdwood Signed-off-by: Bard Liao Link: https://patch.msgid.link/20250204053943.93596-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 381fae5943fe..683e15b459a1 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -753,6 +753,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }; static const struct snd_pci_quirk sof_sdw_ssid_quirk_table[] = { + SND_PCI_QUIRK(0x1043, 0x1e13, "ASUS Zenbook S14", SOC_SDW_CODEC_MIC), {} }; From d8989106287d3735c7e7fc6acb3811d62ebb666c Mon Sep 17 00:00:00 2001 From: Uday M Bhat Date: Tue, 4 Feb 2025 13:39:43 +0800 Subject: [PATCH 04/41] ASoC: Intel: sof_sdw: Add support for Fatcat board with BT offload enabled in PTL platform This change adds an entry for fatcat boards in soundwire quirk table and also, enables BT offload for PTL RVP. Signed-off-by: Uday M Bhat Signed-off-by: Jairaj Arava Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Link: https://patch.msgid.link/20250204053943.93596-4-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 683e15b459a1..203b07d4d833 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -749,6 +749,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOC_SDW_PCH_DMIC), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Fatcat"), + }, + .driver_data = (void *)(SOC_SDW_PCH_DMIC | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, {} }; From 3588b76db7ba798f54dee39a55708b16e1c61de4 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 4 Feb 2025 11:31:33 +0800 Subject: [PATCH 05/41] ASoC: Intel: soc-acpi-intel-tgl-match: declare adr as ull MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The adr is u64. Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20250204033134.92332-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- .../soc/intel/common/soc-acpi-intel-tgl-match.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 6f8c06413665..b77aafb0bfb6 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -658,25 +658,25 @@ static const struct snd_soc_acpi_endpoint cs35l56_7_fb_endpoints[] = { static const struct snd_soc_acpi_adr_device cs35l56_sdw_eight_1_4_fb_adr[] = { { - .adr = 0x00003301fa355601, + .adr = 0x00003301fa355601ull, .num_endpoints = ARRAY_SIZE(cs35l56_l_fb_endpoints), .endpoints = cs35l56_l_fb_endpoints, .name_prefix = "AMP1" }, { - .adr = 0x00003201fa355601, + .adr = 0x00003201fa355601ull, .num_endpoints = ARRAY_SIZE(cs35l56_2_fb_endpoints), .endpoints = cs35l56_2_fb_endpoints, .name_prefix = "AMP2" }, { - .adr = 0x00003101fa355601, + .adr = 0x00003101fa355601ull, .num_endpoints = ARRAY_SIZE(cs35l56_4_fb_endpoints), .endpoints = cs35l56_4_fb_endpoints, .name_prefix = "AMP3" }, { - .adr = 0x00003001fa355601, + .adr = 0x00003001fa355601ull, .num_endpoints = ARRAY_SIZE(cs35l56_6_fb_endpoints), .endpoints = cs35l56_6_fb_endpoints, .name_prefix = "AMP4" @@ -685,25 +685,25 @@ static const struct snd_soc_acpi_adr_device cs35l56_sdw_eight_1_4_fb_adr[] = { static const struct snd_soc_acpi_adr_device cs35l56_sdw_eight_5_8_fb_adr[] = { { - .adr = 0x00013701fa355601, + .adr = 0x00013701fa355601ull, .num_endpoints = ARRAY_SIZE(cs35l56_r_fb_endpoints), .endpoints = cs35l56_r_fb_endpoints, .name_prefix = "AMP8" }, { - .adr = 0x00013601fa355601, + .adr = 0x00013601fa355601ull, .num_endpoints = ARRAY_SIZE(cs35l56_3_fb_endpoints), .endpoints = cs35l56_3_fb_endpoints, .name_prefix = "AMP7" }, { - .adr = 0x00013501fa355601, + .adr = 0x00013501fa355601ull, .num_endpoints = ARRAY_SIZE(cs35l56_5_fb_endpoints), .endpoints = cs35l56_5_fb_endpoints, .name_prefix = "AMP6" }, { - .adr = 0x00013401fa355601, + .adr = 0x00013401fa355601ull, .num_endpoints = ARRAY_SIZE(cs35l56_7_fb_endpoints), .endpoints = cs35l56_7_fb_endpoints, .name_prefix = "AMP5" From 20efccc53abf99fa52ea30a43dec758f6b6b9940 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 4 Feb 2025 11:31:34 +0800 Subject: [PATCH 06/41] ASoC: Intel: soc-acpi-intel-mtl-match: declare adr as ull MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The adr is u64. Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20250204033134.92332-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 770e2194a283..9e611e3667ad 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -330,7 +330,7 @@ static const struct snd_soc_acpi_adr_device rt1316_3_single_adr[] = { static const struct snd_soc_acpi_adr_device rt1318_1_single_adr[] = { { - .adr = 0x000130025D131801, + .adr = 0x000130025D131801ull, .num_endpoints = 1, .endpoints = &single_endpoint, .name_prefix = "rt1318-1" From 6b24e67b4056ba83b1e95e005b7e50fdb1cc6cf4 Mon Sep 17 00:00:00 2001 From: John Keeping Date: Tue, 4 Feb 2025 16:13:10 +0000 Subject: [PATCH 07/41] ASoC: rockchip: i2s-tdm: fix shift config for SND_SOC_DAIFMT_DSP_[AB] Commit 2f45a4e289779 ("ASoC: rockchip: i2s_tdm: Fixup config for SND_SOC_DAIFMT_DSP_A/B") applied a partial change to fix the configuration for DSP A and DSP B formats. The shift control also needs updating to set the correct offset for frame data compared to LRCK. Set the correct values. Fixes: 081068fd64140 ("ASoC: rockchip: add support for i2s-tdm controller") Signed-off-by: John Keeping Link: https://patch.msgid.link/20250204161311.2117240-1-jkeeping@inmusicbrands.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s_tdm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 7f5fcaecee4b..78ab88843f86 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -451,11 +451,11 @@ static int rockchip_i2s_tdm_set_fmt(struct snd_soc_dai *cpu_dai, break; case SND_SOC_DAIFMT_DSP_A: val = I2S_TXCR_TFS_TDM_PCM; - tdm_val = TDM_SHIFT_CTRL(0); + tdm_val = TDM_SHIFT_CTRL(2); break; case SND_SOC_DAIFMT_DSP_B: val = I2S_TXCR_TFS_TDM_PCM; - tdm_val = TDM_SHIFT_CTRL(2); + tdm_val = TDM_SHIFT_CTRL(4); break; default: ret = -EINVAL; From dabbd325b25edb5cdd99c94391817202dd54b651 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 4 Feb 2025 23:50:08 +0000 Subject: [PATCH 08/41] ASoC: simple-card-utils.c: add missing dlc->of_node commit 90de551c1bf ("ASoC: simple-card-utils.c: enable multi Component support") added muiti Component support, but was missing to add dlc->of_node. Because of it, Sound device list will indicates strange name if it was DPCM connection and driver supports dai->driver->dai_args, like below > aplay -l card X: sndulcbmix [xxxx], device 0: fe.(null).rsnd-dai.0 (*) [] ... ^^^^^^ It will be fixed by this patch > aplay -l card X: sndulcbmix [xxxx], device 0: fe.sound@ec500000.rsnd-dai.0 (*) [] ... ^^^^^^^^^^^^^^ Signed-off-by: Kuninori Morimoto Reviewed-by: Daniel Baluta Link: https://patch.msgid.link/87ikpp2rtb.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index dd414634b4ac..c2445c5ccd84 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1092,6 +1092,7 @@ int graph_util_parse_dai(struct device *dev, struct device_node *ep, args.np = ep; dai = snd_soc_get_dai_via_args(&args); if (dai) { + dlc->of_node = node; dlc->dai_name = snd_soc_dai_name_get(dai); dlc->dai_args = snd_soc_copy_dai_args(dev, &args); if (!dlc->dai_args) From 76b0a22d4cf7dc9091129560fdc04e73eb9db4cb Mon Sep 17 00:00:00 2001 From: Edson Juliano Drosdeck Date: Sat, 1 Feb 2025 11:39:30 -0300 Subject: [PATCH 09/41] ALSA: hda/realtek: Limit mic boost on Positivo ARN50 The internal mic boost on the Positivo ARN50 is too high. Fix this by applying the ALC269_FIXUP_LIMIT_INT_MIC_BOOST fixup to the machine to limit the gain. Signed-off-by: Edson Juliano Drosdeck Link: https://patch.msgid.link/20250201143930.25089-1-edson.drosdeck@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 8192be394d0d..ae0beb52e7b0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11045,6 +11045,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1f66, 0x0105, "Ayaneo Portable Game Player", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x2014, 0x800a, "Positivo ARN50", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), From 796106e29e5df6cd4b4e2b51262a8a19e9fa0625 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 5 Feb 2025 00:20:36 +0000 Subject: [PATCH 10/41] ASoC: rsnd: indicate unsupported clock rate It will indicate "unsupported clock rate" when setup clock failed. But it is unclear what kind of rate was failed. Indicate it. Signed-off-by: Kuninori Morimoto Reviewed-by: Yoshihiro Shimoda Link: https://patch.msgid.link/874j192qej.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/renesas/rcar/ssi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/renesas/rcar/ssi.c b/sound/soc/renesas/rcar/ssi.c index b3d4e8ae07ef..0c6424a1fcac 100644 --- a/sound/soc/renesas/rcar/ssi.c +++ b/sound/soc/renesas/rcar/ssi.c @@ -336,7 +336,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, return 0; rate_err: - dev_err(dev, "unsupported clock rate\n"); + dev_err(dev, "unsupported clock rate (%d)\n", rate); + return ret; } From c3fc002b206c6c83d1e3702b979733002ba6fb2c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 5 Feb 2025 00:20:42 +0000 Subject: [PATCH 11/41] ASoC: rsnd: don't indicate warning on rsnd_kctrl_accept_runtime() rsnd_kctrl_accept_runtime() (1) is used for runtime convert rate (= Synchronous SRC Mode). Now, rsnd driver has 2 kctrls for it (A): "SRC Out Rate Switch" (B): "SRC Out Rate" // it calls (1) (A): can be called anytime (B): can be called only runtime, and will indicate warning if it was used at non-runtime. To use runtime convert rate (= Synchronous SRC Mode), user might uses command in below order. (X): > amixer set "SRC Out Rate" on > aplay xxx.wav & (Y): > amixer set "SRC Out Rate" 48010 // convert rate to 48010Hz (Y): calls B (X): calls both A and B. In this case, when user calls (X), it calls both (A) and (B), but it is not yet start running. So, (B) will indicate warning. This warning was added by commit b5c088689847 ("ASoC: rsnd: add warning message to rsnd_kctrl_accept_runtime()"), but the message sounds like the operation was not correct. Let's update warning message. The message is very SRC specific, implement it in src.c Signed-off-by: Kuninori Morimoto Reviewed-by: Yoshihiro Shimoda Link: https://patch.msgid.link/8734gt2qed.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/renesas/rcar/core.c | 14 -------------- sound/soc/renesas/rcar/rsnd.h | 1 - sound/soc/renesas/rcar/src.c | 18 +++++++++++++++++- 3 files changed, 17 insertions(+), 16 deletions(-) diff --git a/sound/soc/renesas/rcar/core.c b/sound/soc/renesas/rcar/core.c index d3709fd0409e..f3f0c3f0bb9f 100644 --- a/sound/soc/renesas/rcar/core.c +++ b/sound/soc/renesas/rcar/core.c @@ -1770,20 +1770,6 @@ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io) return 1; } -int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io) -{ - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - struct rsnd_priv *priv = rsnd_io_to_priv(io); - struct device *dev = rsnd_priv_to_dev(priv); - - if (!runtime) { - dev_warn(dev, "Can't update kctrl when idle\n"); - return 0; - } - - return 1; -} - struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg) { cfg->cfg.val = cfg->val; diff --git a/sound/soc/renesas/rcar/rsnd.h b/sound/soc/renesas/rcar/rsnd.h index a5f54b65313c..04c70690f7a2 100644 --- a/sound/soc/renesas/rcar/rsnd.h +++ b/sound/soc/renesas/rcar/rsnd.h @@ -742,7 +742,6 @@ struct rsnd_kctrl_cfg_s { #define rsnd_kctrl_vals(x) ((x).val) /* = (x).cfg.val[0] */ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io); -int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io); struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg); struct rsnd_kctrl_cfg *rsnd_kctrl_init_s(struct rsnd_kctrl_cfg_s *cfg); int rsnd_kctrl_new(struct rsnd_mod *mod, diff --git a/sound/soc/renesas/rcar/src.c b/sound/soc/renesas/rcar/src.c index e7f86db0d94c..309918029772 100644 --- a/sound/soc/renesas/rcar/src.c +++ b/sound/soc/renesas/rcar/src.c @@ -531,6 +531,22 @@ static irqreturn_t rsnd_src_interrupt(int irq, void *data) return IRQ_HANDLED; } +static int rsnd_src_kctrl_accept_runtime(struct rsnd_dai_stream *io) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + + if (!runtime) { + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_warn(dev, "\"SRC Out Rate\" can use during running\n"); + + return 0; + } + + return 1; +} + static int rsnd_src_probe_(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) @@ -594,7 +610,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate" : "SRC In Rate", - rsnd_kctrl_accept_runtime, + rsnd_src_kctrl_accept_runtime, rsnd_src_set_convert_rate, &src->sync, 192000); From 89f9cf185885d4358aa92b48e51d0f09b71775aa Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 5 Feb 2025 00:20:48 +0000 Subject: [PATCH 12/41] ASoC: rsnd: adjust convert rate limitation Current rsnd driver supports Synchronous SRC Mode, but HW allow to update rate only within 1% from current rate. Adjust to it. Becially, this feature is used to fine-tune subtle difference that occur during sampling rate conversion in SRC. So, it should be called within 1% margin of rate difference. If there was difference over 1%, it will apply with 1% increments by using loop without indicating error message. Cc: Yoshihiro Shimoda Signed-off-by: Kuninori Morimoto Reviewed-by: Yoshihiro Shimoda Tested-by: Yoshihiro Shimoda Link: https://patch.msgid.link/871pwd2qe8.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/renesas/rcar/src.c | 98 ++++++++++++++++++++++++++++-------- 1 file changed, 76 insertions(+), 22 deletions(-) diff --git a/sound/soc/renesas/rcar/src.c b/sound/soc/renesas/rcar/src.c index 309918029772..7d73b183bda6 100644 --- a/sound/soc/renesas/rcar/src.c +++ b/sound/soc/renesas/rcar/src.c @@ -35,6 +35,7 @@ struct rsnd_src { struct rsnd_mod *dma; struct rsnd_kctrl_cfg_s sen; /* sync convert enable */ struct rsnd_kctrl_cfg_s sync; /* sync convert */ + u32 current_sync_rate; int irq; }; @@ -100,7 +101,7 @@ static u32 rsnd_src_convert_rate(struct rsnd_dai_stream *io, if (!rsnd_src_sync_is_enabled(mod)) return rsnd_io_converted_rate(io); - convert_rate = src->sync.val; + convert_rate = src->current_sync_rate; if (!convert_rate) convert_rate = rsnd_io_converted_rate(io); @@ -201,13 +202,73 @@ static const u32 chan222222[] = { static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_src *src = rsnd_mod_to_src(mod); + u32 fin, fout, new_rate; + int inc, cnt, rate; + u64 base, val; + + if (!runtime) + return; + + if (!rsnd_src_sync_is_enabled(mod)) + return; + + fin = rsnd_src_get_in_rate(priv, io); + fout = rsnd_src_get_out_rate(priv, io); + + new_rate = src->sync.val; + + if (!new_rate) + new_rate = fout; + + /* Do nothing if no diff */ + if (new_rate == src->current_sync_rate) + return; + + /* + * SRCm_IFSVR::INTIFS can change within 1% + * see + * SRCm_IFSVR::INTIFS Note + */ + inc = fout / 100; + cnt = abs(new_rate - fout) / inc; + if (fout > new_rate) + inc *= -1; + + /* + * After start running SRC, we can update only SRC_IFSVR + * for Synchronous Mode + */ + base = (u64)0x0400000 * fin; + rate = fout; + for (int i = 0; i < cnt; i++) { + val = base; + rate += inc; + do_div(val, rate); + + rsnd_mod_write(mod, SRC_IFSVR, val); + } + val = base; + do_div(val, new_rate); + + rsnd_mod_write(mod, SRC_IFSVR, val); + + /* update current_sync_rate */ + src->current_sync_rate = new_rate; +} + +static void rsnd_src_init_convert_rate(struct rsnd_dai_stream *io, + struct rsnd_mod *mod) +{ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); int is_play = rsnd_io_is_play(io); int use_src = 0; u32 fin, fout; - u32 ifscr, fsrate, adinr; + u32 ifscr, adinr; u32 cr, route; u32 i_busif, o_busif, tmp; const u32 *bsdsr_table; @@ -245,26 +306,15 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, adinr = rsnd_get_adinr_bit(mod, io) | chan; /* - * SRC_IFSCR / SRC_IFSVR - */ - ifscr = 0; - fsrate = 0; - if (use_src) { - u64 n; - - ifscr = 1; - n = (u64)0x0400000 * fin; - do_div(n, fout); - fsrate = n; - } - - /* + * SRC_IFSCR * SRC_SRCCR / SRC_ROUTE_MODE0 */ + ifscr = 0; cr = 0x00011110; route = 0x0; if (use_src) { route = 0x1; + ifscr = 0x1; if (rsnd_src_sync_is_enabled(mod)) { cr |= 0x1; @@ -335,7 +385,6 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, rsnd_mod_write(mod, SRC_SRCIR, 1); /* initialize */ rsnd_mod_write(mod, SRC_ADINR, adinr); rsnd_mod_write(mod, SRC_IFSCR, ifscr); - rsnd_mod_write(mod, SRC_IFSVR, fsrate); rsnd_mod_write(mod, SRC_SRCCR, cr); rsnd_mod_write(mod, SRC_BSDSR, bsdsr_table[idx]); rsnd_mod_write(mod, SRC_BSISR, bsisr_table[idx]); @@ -348,6 +397,9 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, rsnd_adg_set_src_timesel_gen2(mod, io, fin, fout); + /* update SRC_IFSVR */ + rsnd_src_set_convert_rate(io, mod); + return; convert_rate_err: @@ -467,7 +519,8 @@ static int rsnd_src_init(struct rsnd_mod *mod, int ret; /* reset sync convert_rate */ - src->sync.val = 0; + src->sync.val = + src->current_sync_rate = 0; ret = rsnd_mod_power_on(mod); if (ret < 0) @@ -475,7 +528,7 @@ static int rsnd_src_init(struct rsnd_mod *mod, rsnd_src_activation(mod); - rsnd_src_set_convert_rate(io, mod); + rsnd_src_init_convert_rate(io, mod); rsnd_src_status_clear(mod); @@ -493,7 +546,8 @@ static int rsnd_src_quit(struct rsnd_mod *mod, rsnd_mod_power_off(mod); /* reset sync convert_rate */ - src->sync.val = 0; + src->sync.val = + src->current_sync_rate = 0; return 0; } @@ -601,7 +655,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, "SRC Out Rate Switch" : "SRC In Rate Switch", rsnd_kctrl_accept_anytime, - rsnd_src_set_convert_rate, + rsnd_src_init_convert_rate, &src->sen, 1); if (ret < 0) return ret; From d8d99c3b5c485f339864aeaa29f76269cc0ea975 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 5 Feb 2025 15:52:31 +0200 Subject: [PATCH 13/41] ASoC: SOF: stream-ipc: Check for cstream nullity in sof_ipc_msg_data() The nullity of sps->cstream should be checked similarly as it is done in sof_set_stream_data_offset() function. Assuming that it is not NULL if sps->stream is NULL is incorrect and can lead to NULL pointer dereference. Fixes: 090349a9feba ("ASoC: SOF: Add support for compress API for stream data/offset") Cc: stable@vger.kernel.org Reported-by: Curtis Malainey Closes: https://github.com/thesofproject/linux/pull/5214 Signed-off-by: Peter Ujfalusi Reviewed-by: Daniel Baluta Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Curtis Malainey Link: https://patch.msgid.link/20250205135232.19762-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/stream-ipc.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/stream-ipc.c b/sound/soc/sof/stream-ipc.c index 794c7bbccbaf..8262443ac89a 100644 --- a/sound/soc/sof/stream-ipc.c +++ b/sound/soc/sof/stream-ipc.c @@ -43,7 +43,7 @@ int sof_ipc_msg_data(struct snd_sof_dev *sdev, return -ESTRPIPE; posn_offset = stream->posn_offset; - } else { + } else if (sps->cstream) { struct sof_compr_stream *sstream = sps->cstream->runtime->private_data; @@ -51,6 +51,10 @@ int sof_ipc_msg_data(struct snd_sof_dev *sdev, return -ESTRPIPE; posn_offset = sstream->posn_offset; + + } else { + dev_err(sdev->dev, "%s: No stream opened\n", __func__); + return -EINVAL; } snd_sof_dsp_mailbox_read(sdev, posn_offset, p, sz); From 46c7b901e2a03536df5a3cb40b3b26e2be505df6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 5 Feb 2025 15:52:32 +0200 Subject: [PATCH 14/41] ASoC: SOF: pcm: Clear the susbstream pointer to NULL on close The spcm->stream[substream->stream].substream is set during open and was left untouched. After the first PCM stream it will never be NULL and we have code which checks for substream NULLity as indication if the stream is active or not. For the compressed cstream pointer the same has been done, this change will correct the handling of PCM streams. Fixes: 090349a9feba ("ASoC: SOF: Add support for compress API for stream data/offset") Cc: stable@vger.kernel.org Reported-by: Curtis Malainey Closes: https://github.com/thesofproject/linux/pull/5214 Signed-off-by: Peter Ujfalusi Reviewed-by: Daniel Baluta Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Curtis Malainey Link: https://patch.msgid.link/20250205135232.19762-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 35a7462d8b69..c5c6353f18ce 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -511,6 +511,8 @@ static int sof_pcm_close(struct snd_soc_component *component, */ } + spcm->stream[substream->stream].substream = NULL; + return 0; } From 679074942c2502a95842a80471d8fb718165ac77 Mon Sep 17 00:00:00 2001 From: Vitaly Rodionov Date: Wed, 5 Feb 2025 16:08:46 +0000 Subject: [PATCH 15/41] ASoC: arizona/madera: use fsleep() in up/down DAPM event delays. Using `fsleep` instead of `msleep` resolves some customer complaints regarding the precision of up/down DAPM event timing. `fsleep()` automatically selects the appropriate sleep function, making the delay time more predictable. Signed-off-by: Vitaly Rodionov Link: https://patch.msgid.link/20250205160849.500306-1-vitalyr@opensource.cirrus.com Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 14 +++++++------- sound/soc/codecs/madera.c | 10 +++++----- sound/soc/codecs/wm5110.c | 8 ++++---- 3 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 402b9a2ff024..68cdb1027d0c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -967,7 +967,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, case ARIZONA_OUT3L_ENA_SHIFT: case ARIZONA_OUT3R_ENA_SHIFT: priv->out_up_pending++; - priv->out_up_delay += 17; + priv->out_up_delay += 17000; break; case ARIZONA_OUT4L_ENA_SHIFT: case ARIZONA_OUT4R_ENA_SHIFT: @@ -977,7 +977,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, case WM8997: break; default: - priv->out_up_delay += 10; + priv->out_up_delay += 10000; break; } break; @@ -999,7 +999,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, if (!priv->out_up_pending && priv->out_up_delay) { dev_dbg(component->dev, "Power up delay: %d\n", priv->out_up_delay); - msleep(priv->out_up_delay); + fsleep(priv->out_up_delay); priv->out_up_delay = 0; } break; @@ -1017,7 +1017,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, case ARIZONA_OUT3L_ENA_SHIFT: case ARIZONA_OUT3R_ENA_SHIFT: priv->out_down_pending++; - priv->out_down_delay++; + priv->out_down_delay += 1000; break; case ARIZONA_OUT4L_ENA_SHIFT: case ARIZONA_OUT4R_ENA_SHIFT: @@ -1028,10 +1028,10 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, break; case WM8998: case WM1814: - priv->out_down_delay += 5; + priv->out_down_delay += 5000; break; default: - priv->out_down_delay++; + priv->out_down_delay += 1000; break; } break; @@ -1053,7 +1053,7 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, if (!priv->out_down_pending && priv->out_down_delay) { dev_dbg(component->dev, "Power down delay: %d\n", priv->out_down_delay); - msleep(priv->out_down_delay); + fsleep(priv->out_down_delay); priv->out_down_delay = 0; } break; diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index a840a2eb92b9..af109761f359 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -2323,10 +2323,10 @@ int madera_out_ev(struct snd_soc_dapm_widget *w, case CS42L92: case CS47L92: case CS47L93: - out_up_delay = 6; + out_up_delay = 6000; break; default: - out_up_delay = 17; + out_up_delay = 17000; break; } @@ -2357,7 +2357,7 @@ int madera_out_ev(struct snd_soc_dapm_widget *w, case MADERA_OUT3R_ENA_SHIFT: priv->out_up_pending--; if (!priv->out_up_pending) { - msleep(priv->out_up_delay); + fsleep(priv->out_up_delay); priv->out_up_delay = 0; } break; @@ -2376,7 +2376,7 @@ int madera_out_ev(struct snd_soc_dapm_widget *w, case MADERA_OUT3L_ENA_SHIFT: case MADERA_OUT3R_ENA_SHIFT: priv->out_down_pending++; - priv->out_down_delay++; + priv->out_down_delay += 1000; break; default: break; @@ -2393,7 +2393,7 @@ int madera_out_ev(struct snd_soc_dapm_widget *w, case MADERA_OUT3R_ENA_SHIFT: priv->out_down_pending--; if (!priv->out_down_pending) { - msleep(priv->out_down_delay); + fsleep(priv->out_down_delay); priv->out_down_delay = 0; } break; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 502196253d42..64eee0d2347d 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -302,7 +302,7 @@ static int wm5110_hp_pre_enable(struct snd_soc_dapm_widget *w) } else { wseq = wm5110_no_dre_left_enable; nregs = ARRAY_SIZE(wm5110_no_dre_left_enable); - priv->out_up_delay += 10; + priv->out_up_delay += 10000; } break; case ARIZONA_OUT1R_ENA_SHIFT: @@ -312,7 +312,7 @@ static int wm5110_hp_pre_enable(struct snd_soc_dapm_widget *w) } else { wseq = wm5110_no_dre_right_enable; nregs = ARRAY_SIZE(wm5110_no_dre_right_enable); - priv->out_up_delay += 10; + priv->out_up_delay += 10000; } break; default: @@ -338,7 +338,7 @@ static int wm5110_hp_pre_disable(struct snd_soc_dapm_widget *w) snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, ARIZONA_WS_TRG1, 0); - priv->out_down_delay += 27; + priv->out_down_delay += 27000; } break; case ARIZONA_OUT1R_ENA_SHIFT: @@ -350,7 +350,7 @@ static int wm5110_hp_pre_disable(struct snd_soc_dapm_widget *w) snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, ARIZONA_WS_TRG2, 0); - priv->out_down_delay += 27; + priv->out_down_delay += 27000; } break; default: From 1d44a30ae3f9195cb4eb7d81bb9ced2776232094 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Wed, 5 Feb 2025 16:48:04 +0000 Subject: [PATCH 16/41] ASoC: cs35l41: Fallback to using HID for system_name if no SUB is available MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For systems which load firmware on the cs35l41 which use ACPI, the _SUB value is used to differentiate firmware and tuning files for the individual systems. In the case where a system does not have a _SUB defined in ACPI node for cs35l41, there needs to be a fallback to allow the files for that system to be differentiated. Since all ACPI nodes for cs35l41 should have a HID defined, the HID should be a safe option. Signed-off-by: Stefan Binding Reviewed-by: André Almeida Tested-by: André Almeida Link: https://patch.msgid.link/20250205164806.414020-1-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41.c | 23 ++++++++++++++++------- 1 file changed, 16 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 07a5cab35fe1..30b89018b113 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -1150,19 +1150,28 @@ static int cs35l41_dsp_init(struct cs35l41_private *cs35l41) static int cs35l41_acpi_get_name(struct cs35l41_private *cs35l41) { - acpi_handle handle = ACPI_HANDLE(cs35l41->dev); + struct acpi_device *adev = ACPI_COMPANION(cs35l41->dev); + acpi_handle handle = acpi_device_handle(adev); + const char *hid; const char *sub; - /* If there is no ACPI_HANDLE, there is no ACPI for this system, return 0 */ - if (!handle) + /* If there is no acpi_device, there is no ACPI for this system, return 0 */ + if (!adev) return 0; sub = acpi_get_subsystem_id(handle); if (IS_ERR(sub)) { - /* If bad ACPI, return 0 and fallback to legacy firmware path, otherwise fail */ - if (PTR_ERR(sub) == -ENODATA) - return 0; - else + /* If no _SUB, fallback to _HID, otherwise fail */ + if (PTR_ERR(sub) == -ENODATA) { + hid = acpi_device_hid(adev); + /* If dummy hid, return 0 and fallback to legacy firmware path */ + if (!strcmp(hid, "device")) + return 0; + sub = kstrdup(hid, GFP_KERNEL); + if (!sub) + sub = ERR_PTR(-ENOMEM); + + } else return PTR_ERR(sub); } From 6fd60136d256b3b948333ebdb3835f41a95ab7ef Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 6 Feb 2025 10:46:42 +0200 Subject: [PATCH 17/41] ASoC: SOF: ipc4-topology: Harden loops for looking up ALH copiers Other, non DAI copier widgets could have the same stream name (sname) as the ALH copier and in that case the copier->data is NULL, no alh_data is attached, which could lead to NULL pointer dereference. We could check for this NULL pointer in sof_ipc4_prepare_copier_module() and avoid the crash, but a similar loop in sof_ipc4_widget_setup_comp_dai() will miscalculate the ALH device count, causing broken audio. The correct fix is to harden the matching logic by making sure that the 1. widget is a DAI widget - so dai = w->private is valid 2. the dai (and thus the copier) is ALH copier Fixes: a150345aa758 ("ASoC: SOF: ipc4-topology: add SoundWire/ALH aggregation support") Reported-by: Seppo Ingalsuo Link: https://github.com/thesofproject/sof/pull/9652 Signed-off-by: Peter Ujfalusi Reviewed-by: Liam Girdwood Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Link: https://patch.msgid.link/20250206084642.14988-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index c04c62478827..6d5cda813e48 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -765,10 +765,16 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) } list_for_each_entry(w, &sdev->widget_list, list) { - if (w->widget->sname && + struct snd_sof_dai *alh_dai; + + if (!WIDGET_IS_DAI(w->id) || !w->widget->sname || strcmp(w->widget->sname, swidget->widget->sname)) continue; + alh_dai = w->private; + if (alh_dai->type != SOF_DAI_INTEL_ALH) + continue; + blob->alh_cfg.device_count++; } @@ -2061,11 +2067,13 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, list_for_each_entry(w, &sdev->widget_list, list) { u32 node_type; - if (w->widget->sname && + if (!WIDGET_IS_DAI(w->id) || !w->widget->sname || strcmp(w->widget->sname, swidget->widget->sname)) continue; dai = w->private; + if (dai->type != SOF_DAI_INTEL_ALH) + continue; alh_copier = (struct sof_ipc4_copier *)dai->private; alh_data = &alh_copier->data; node_type = SOF_IPC4_GET_NODE_TYPE(alh_data->gtw_cfg.node_id); From 33b7dc7843dbdc9b90c91d11ba30b107f9138ffd Mon Sep 17 00:00:00 2001 From: Terry Cheong Date: Thu, 6 Feb 2025 11:47:23 +0200 Subject: [PATCH 18/41] ASoC: SOF: Intel: hda: add softdep pre to snd-hda-codec-hdmi module MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In enviornment without KMOD requesting module may fail to load snd-hda-codec-hdmi, resulting in HDMI audio not usable. Add softdep to loading HDMI codec module first to ensure we can load it correctly. Signed-off-by: Terry Cheong Reviewed-by: Bard Liao Reviewed-by: Johny Lin Reviewed-by: Péter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://patch.msgid.link/20250206094723.18013-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 568f3dfe822f..2f9925830d1d 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -454,6 +454,7 @@ int hda_codec_i915_exit(struct snd_sof_dev *sdev) } EXPORT_SYMBOL_NS_GPL(hda_codec_i915_exit, "SND_SOC_SOF_HDA_AUDIO_CODEC_I915"); +MODULE_SOFTDEP("pre: snd-hda-codec-hdmi"); #endif MODULE_LICENSE("Dual BSD/GPL"); From 0b06000704f8ae72056ad777a67742b7799d6660 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Thu, 6 Feb 2025 20:38:08 +0800 Subject: [PATCH 19/41] ASoC: tas2781: drop a redundant code Report from internal ticket, priv->cali_data.data devm_kzalloc twice, drop the first one, it is the unnecessary one. Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20250206123808.1590-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-i2c.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index a730ab6ad4e3..90c5b2e74d12 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -1260,8 +1260,6 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) (cali_data->cali_dat_sz_per_dev + 1) + 1 + 15 + 1; priv->cali_data.total_sz = priv->ndev * (cali_data->cali_dat_sz_per_dev + 1); - priv->cali_data.data = devm_kzalloc(priv->dev, - ext_cali_data->max, GFP_KERNEL); cali_ctrls[i].name = cali_name; cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; cali_ctrls[i].info = snd_soc_bytes_info_ext; From a1f7b7ff0e10ae574d388131596390157222f986 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 10 Feb 2025 10:17:27 +0200 Subject: [PATCH 20/41] PCI: pci_ids: add INTEL_HDA_PTL_H Add Intel PTL-H audio Device ID. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Bard Liao Acked-by: Bjorn Helgaas Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20250210081730.22916-2-peter.ujfalusi@linux.intel.com --- include/linux/pci_ids.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index de5deb1a0118..1a2594a38199 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -3134,6 +3134,7 @@ #define PCI_DEVICE_ID_INTEL_HDA_LNL_P 0xa828 #define PCI_DEVICE_ID_INTEL_S21152BB 0xb152 #define PCI_DEVICE_ID_INTEL_HDA_BMG 0xe2f7 +#define PCI_DEVICE_ID_INTEL_HDA_PTL_H 0xe328 #define PCI_DEVICE_ID_INTEL_HDA_PTL 0xe428 #define PCI_DEVICE_ID_INTEL_HDA_CML_R 0xf0c8 #define PCI_DEVICE_ID_INTEL_HDA_RKL_S 0xf1c8 From 214e6be2d91d5d58f28d3a37630480077a1aafbd Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 10 Feb 2025 10:17:28 +0200 Subject: [PATCH 21/41] ALSA: hda: intel-dsp-config: Add PTL-H support Use same recipes as PTL for PTL-H. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Bard Liao Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20250210081730.22916-3-peter.ujfalusi@linux.intel.com --- sound/hda/intel-dsp-config.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index f564ec7af194..ce3ae2cba660 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -539,6 +539,11 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_PTL, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = PCI_DEVICE_ID_INTEL_HDA_PTL_H, + }, + #endif }; From 4e9c87cfcd0584f2a2e2f352a43ff003d688f3a4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Feb 2025 10:17:29 +0200 Subject: [PATCH 22/41] ASoC: SOF: Intel: pci-ptl: Add support for PTL-H PTL-H uses the same configuration as PTL. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Bard Liao Acked-by: Mark Brown Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20250210081730.22916-4-peter.ujfalusi@linux.intel.com --- sound/soc/sof/intel/pci-ptl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/pci-ptl.c b/sound/soc/sof/intel/pci-ptl.c index 0aacdfac9fb4..c4fb6a2441b7 100644 --- a/sound/soc/sof/intel/pci-ptl.c +++ b/sound/soc/sof/intel/pci-ptl.c @@ -50,6 +50,7 @@ static const struct sof_dev_desc ptl_desc = { /* PCI IDs */ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_PTL, &ptl_desc) }, /* PTL */ + { PCI_DEVICE_DATA(INTEL, HDA_PTL_H, &ptl_desc) }, /* PTL-H */ { 0, } }; MODULE_DEVICE_TABLE(pci, sof_pci_ids); From d7e2447a4d51de5c3c03e3b7892898e98ddd9769 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 10 Feb 2025 10:17:30 +0200 Subject: [PATCH 23/41] ALSA: hda: hda-intel: add Panther Lake-H support Add Intel PTL-H audio Device ID. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Bard Liao Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20250210081730.22916-5-peter.ujfalusi@linux.intel.com --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7d7f9aac50a9..67540e037309 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2496,6 +2496,8 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_ARL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, /* Panther Lake */ { PCI_DEVICE_DATA(INTEL, HDA_PTL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_LNL) }, + /* Panther Lake-H */ + { PCI_DEVICE_DATA(INTEL, HDA_PTL_H, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_LNL) }, /* Apollolake (Broxton-P) */ { PCI_DEVICE_DATA(INTEL, HDA_APL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON) }, /* Gemini-Lake */ From 70e90680c2592c38c62e5716f1296a2d74bae7af Mon Sep 17 00:00:00 2001 From: Nam Cao Date: Wed, 5 Feb 2025 11:46:33 +0100 Subject: [PATCH 24/41] ALSA: Switch to use hrtimer_setup() hrtimer_setup() takes the callback function pointer as argument and initializes the timer completely. Replace hrtimer_init() and the open coded initialization of hrtimer::function with the new setup mechanism. Patch was created by using Coccinelle. Acked-by: Zack Rusin Signed-off-by: Nam Cao Cc: Takashi Iwai Link: https://patch.msgid.link/598031332ce738c82286a158cb66eb7e735b2e79.1738746904.git.namcao@linutronix.de Signed-off-by: Takashi Iwai --- sound/core/hrtimer.c | 3 +-- sound/drivers/dummy.c | 3 +-- sound/drivers/pcsp/pcsp.c | 3 +-- sound/sh/sh_dac_audio.c | 3 +-- 4 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 147c1fea4708..e9c60dce59fb 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -66,9 +66,8 @@ static int snd_hrtimer_open(struct snd_timer *t) stime = kzalloc(sizeof(*stime), GFP_KERNEL); if (!stime) return -ENOMEM; - hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); stime->timer = t; - stime->hrt.function = snd_hrtimer_callback; + hrtimer_setup(&stime->hrt, snd_hrtimer_callback, CLOCK_MONOTONIC, HRTIMER_MODE_REL); t->private_data = stime; return 0; } diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 8f5df9b3aaaa..c1a3efb633c5 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -457,8 +457,7 @@ static int dummy_hrtimer_create(struct snd_pcm_substream *substream) if (!dpcm) return -ENOMEM; substream->runtime->private_data = dpcm; - hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL_SOFT); - dpcm->timer.function = dummy_hrtimer_callback; + hrtimer_setup(&dpcm->timer, dummy_hrtimer_callback, CLOCK_MONOTONIC, HRTIMER_MODE_REL_SOFT); dpcm->substream = substream; atomic_set(&dpcm->running, 0); return 0; diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 78c9b1c7590f..e8482c2290c3 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -103,8 +103,7 @@ static int snd_card_pcsp_probe(int devnum, struct device *dev) if (devnum != 0) return -EINVAL; - hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); - pcsp_chip.timer.function = pcsp_do_timer; + hrtimer_setup(&pcsp_chip.timer, pcsp_do_timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); err = snd_devm_card_new(dev, index, id, THIS_MODULE, 0, &card); if (err < 0) diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index 3f5422145c5e..84a4b17a0cc2 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -312,8 +312,7 @@ static int snd_sh_dac_create(struct snd_card *card, chip->card = card; - hrtimer_init(&chip->hrtimer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); - chip->hrtimer.function = sh_dac_audio_timer; + hrtimer_setup(&chip->hrtimer, sh_dac_audio_timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); dac_audio_reset(chip); chip->rate = 8000; From 78ccf6a6bae11e451e20a52dd2bc2ab98f66326b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Feb 2025 11:19:53 +0800 Subject: [PATCH 25/41] ASoC: Intel: soc-acpi-intel-ptl-match: revise typo of rt712_vb + rt1320 support s/lnl/ptl Fixes: bd40d912728f ("ASoC: Intel: soc-acpi-intel-ptl-match: add rt712_vb + rt1320 support") Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Link: https://patch.msgid.link/20250210031954.6287-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c index 9eb4a43e3e7a..e487c4e1c034 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c @@ -270,7 +270,7 @@ static const struct snd_soc_acpi_link_adr lnl_sdw_rt713_vb_l2_rt1320_l13[] = { {} }; -static const struct snd_soc_acpi_link_adr lnl_sdw_rt712_vb_l2_rt1320_l1[] = { +static const struct snd_soc_acpi_link_adr ptl_sdw_rt712_vb_l2_rt1320_l1[] = { { .mask = BIT(2), .num_adr = ARRAY_SIZE(rt712_vb_2_group1_adr), @@ -337,10 +337,10 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = { }, { .link_mask = BIT(1) | BIT(2), - .links = lnl_sdw_rt712_vb_l2_rt1320_l1, + .links = ptl_sdw_rt712_vb_l2_rt1320_l1, .drv_name = "sof_sdw", .machine_check = snd_soc_acpi_intel_sdca_is_device_rt712_vb, - .sof_tplg_filename = "sof-lnl-rt712-l2-rt1320-l1.tplg" + .sof_tplg_filename = "sof-ptl-rt712-l2-rt1320-l1.tplg" }, { .link_mask = BIT(1) | BIT(2) | BIT(3), From cb78b8dc7834066539253c039f276b3625fecd9f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Feb 2025 11:19:54 +0800 Subject: [PATCH 26/41] ASoC: Intel: soc-acpi-intel-ptl-match: revise typo of rt713_vb_l2_rt1320_l13 s/lnl/ptl Fixes: a7ebb0255188 ("ASoC: Intel: soc-acpi-intel-ptl-match: add rt713_vb_l2_rt1320_l13 support") Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Link: https://patch.msgid.link/20250210031954.6287-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c index e487c4e1c034..dd7993b76dee 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c @@ -251,7 +251,7 @@ static const struct snd_soc_acpi_link_adr ptl_rvp[] = { {} }; -static const struct snd_soc_acpi_link_adr lnl_sdw_rt713_vb_l2_rt1320_l13[] = { +static const struct snd_soc_acpi_link_adr ptl_sdw_rt713_vb_l2_rt1320_l13[] = { { .mask = BIT(2), .num_adr = ARRAY_SIZE(rt713_vb_2_adr), @@ -344,10 +344,10 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = { }, { .link_mask = BIT(1) | BIT(2) | BIT(3), - .links = lnl_sdw_rt713_vb_l2_rt1320_l13, + .links = ptl_sdw_rt713_vb_l2_rt1320_l13, .drv_name = "sof_sdw", .machine_check = snd_soc_acpi_intel_sdca_is_device_rt712_vb, - .sof_tplg_filename = "sof-lnl-rt713-l2-rt1320-l13.tplg" + .sof_tplg_filename = "sof-ptl-rt713-l2-rt1320-l13.tplg" }, {}, }; From 91b98d5a6e8067c5226207487681a48f0d651e46 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Fri, 7 Feb 2025 13:46:02 +0200 Subject: [PATCH 27/41] ASoC: SOF: amd: Add post_fw_run_delay ACP quirk Stress testing resume from suspend on Valve Steam Deck OLED (Galileo) revealed that the DSP firmware could enter an unrecoverable faulty state, where the kernel ring buffer is flooded with IPC related error messages: [ +0.017002] snd_sof_amd_vangogh 0000:04:00.5: acp_sof_ipc_send_msg: Failed to acquire HW lock [ +0.000054] snd_sof_amd_vangogh 0000:04:00.5: ipc3_tx_msg_unlocked: ipc message send for 0x30100000 failed: -22 [ +0.000005] snd_sof_amd_vangogh 0000:04:00.5: Failed to setup widget PIPELINE.6.ACPHS1.IN [ +0.000004] snd_sof_amd_vangogh 0000:04:00.5: Failed to restore pipeline after resume -22 [ +0.000003] snd_sof_amd_vangogh 0000:04:00.5: PM: dpm_run_callback(): pci_pm_resume returns -22 [ +0.000009] snd_sof_amd_vangogh 0000:04:00.5: PM: failed to resume async: error -22 [...] [ +0.002582] PM: suspend exit [ +0.065085] snd_sof_amd_vangogh 0000:04:00.5: ipc tx error for 0x30130000 (msg/reply size: 12/0): -22 [ +0.000499] snd_sof_amd_vangogh 0000:04:00.5: error: failed widget list set up for pcm 1 dir 0 [ +0.000011] snd_sof_amd_vangogh 0000:04:00.5: error: set pcm hw_params after resume [ +0.000006] snd_sof_amd_vangogh 0000:04:00.5: ASoC: error at snd_soc_pcm_component_prepare on 0000:04:00.5: -22 [...] A system reboot would be necessary to restore the speakers functionality. However, by delaying a bit any host to DSP transmission right after the firmware boot completed, the issue could not be reproduced anymore and sound continued to work flawlessly even after performing thousands of suspend/resume cycles. Introduce the post_fw_run_delay ACP quirk to allow providing the aforementioned delay via the snd_sof_dsp_ops->post_fw_run() callback for the affected devices. Signed-off-by: Cristian Ciocaltea Link: https://patch.msgid.link/20250207-sof-vangogh-fixes-v1-1-67824c1e4c9a@collabora.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 1 + sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/vangogh.c | 18 ++++++++++++++++++ 3 files changed, 20 insertions(+) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 33648ff8b833..9e13c96528be 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -27,6 +27,7 @@ MODULE_PARM_DESC(enable_fw_debug, "Enable Firmware debug"); static struct acp_quirk_entry quirk_valve_galileo = { .signed_fw_image = true, .skip_iram_dram_size_mod = true, + .post_fw_run_delay = true, }; const struct dmi_system_id acp_sof_quirk_table[] = { diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 800594440f73..2a19d82d6200 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -220,6 +220,7 @@ struct sof_amd_acp_desc { struct acp_quirk_entry { bool signed_fw_image; bool skip_iram_dram_size_mod; + bool post_fw_run_delay; }; /* Common device data struct for ACP devices */ diff --git a/sound/soc/sof/amd/vangogh.c b/sound/soc/sof/amd/vangogh.c index 8e2672106ac6..d5f1dddd43e7 100644 --- a/sound/soc/sof/amd/vangogh.c +++ b/sound/soc/sof/amd/vangogh.c @@ -11,6 +11,7 @@ * Hardware interface for Audio DSP on Vangogh platform */ +#include #include #include @@ -136,6 +137,20 @@ static struct snd_soc_dai_driver vangogh_sof_dai[] = { }, }; +static int sof_vangogh_post_fw_run_delay(struct snd_sof_dev *sdev) +{ + /* + * Resuming from suspend in some cases my cause the DSP firmware + * to enter an unrecoverable faulty state. Delaying a bit any host + * to DSP transmission right after firmware boot completion seems + * to resolve the issue. + */ + if (!sdev->first_boot) + usleep_range(100, 150); + + return 0; +} + /* Vangogh ops */ struct snd_sof_dsp_ops sof_vangogh_ops; EXPORT_SYMBOL_NS(sof_vangogh_ops, "SND_SOC_SOF_AMD_COMMON"); @@ -157,6 +172,9 @@ int sof_vangogh_ops_init(struct snd_sof_dev *sdev) if (quirks->signed_fw_image) sof_vangogh_ops.load_firmware = acp_sof_load_signed_firmware; + + if (quirks->post_fw_run_delay) + sof_vangogh_ops.post_fw_run = sof_vangogh_post_fw_run_delay; } return 0; From 2ecbc2e9f3b19e2199e8bc3ba603d299f1985f09 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Fri, 7 Feb 2025 13:46:03 +0200 Subject: [PATCH 28/41] ASoC: SOF: amd: Drop unused includes from Vangogh driver Remove all the includes for headers which are not (directly) used from the Vangogh SOF driver sources. Signed-off-by: Cristian Ciocaltea Reviewed-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20250207-sof-vangogh-fixes-v1-2-67824c1e4c9a@collabora.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/pci-vangogh.c | 2 -- sound/soc/sof/amd/vangogh.c | 4 ---- 2 files changed, 6 deletions(-) diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c index 53f64d6bc91b..28f2d4050a67 100644 --- a/sound/soc/sof/amd/pci-vangogh.c +++ b/sound/soc/sof/amd/pci-vangogh.c @@ -13,11 +13,9 @@ #include #include -#include #include #include -#include "../ops.h" #include "../sof-pci-dev.h" #include "../../amd/mach-config.h" #include "acp.h" diff --git a/sound/soc/sof/amd/vangogh.c b/sound/soc/sof/amd/vangogh.c index d5f1dddd43e7..6ed5f9aaa414 100644 --- a/sound/soc/sof/amd/vangogh.c +++ b/sound/soc/sof/amd/vangogh.c @@ -12,13 +12,9 @@ */ #include -#include #include -#include "../ops.h" -#include "../sof-audio.h" #include "acp.h" -#include "acp-dsp-offset.h" #define I2S_HS_INSTANCE 0 #define I2S_BT_INSTANCE 1 From ac84ca815adb4171a4276b1d44096b75f6a150b7 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Fri, 7 Feb 2025 13:46:04 +0200 Subject: [PATCH 29/41] ASoC: SOF: amd: Handle IPC replies before FW_BOOT_COMPLETE In some cases, e.g. during resuming from suspend, there is a possibility that some IPC reply messages get received by the host while the DSP firmware has not yet reached the complete boot state. Detect when this happens and do not attempt to process the unexpected replies from DSP. Instead, provide proper debugging support. Signed-off-by: Cristian Ciocaltea Link: https://patch.msgid.link/20250207-sof-vangogh-fixes-v1-3-67824c1e4c9a@collabora.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-ipc.c | 23 ++++++++++++++++------- 1 file changed, 16 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c index 5f371d9263f3..12caefd08788 100644 --- a/sound/soc/sof/amd/acp-ipc.c +++ b/sound/soc/sof/amd/acp-ipc.c @@ -167,6 +167,7 @@ irqreturn_t acp_sof_ipc_irq_thread(int irq, void *context) if (sdev->first_boot && sdev->fw_state != SOF_FW_BOOT_COMPLETE) { acp_mailbox_read(sdev, sdev->dsp_box.offset, &status, sizeof(status)); + if ((status & SOF_IPC_PANIC_MAGIC_MASK) == SOF_IPC_PANIC_MAGIC) { snd_sof_dsp_panic(sdev, sdev->dsp_box.offset + sizeof(status), true); @@ -188,13 +189,21 @@ irqreturn_t acp_sof_ipc_irq_thread(int irq, void *context) dsp_ack = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_SCRATCH_REG_0 + dsp_ack_write); if (dsp_ack) { - spin_lock_irq(&sdev->ipc_lock); - /* handle immediate reply from DSP core */ - acp_dsp_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, 0); - /* set the done bit */ - acp_dsp_ipc_dsp_done(sdev); - spin_unlock_irq(&sdev->ipc_lock); + if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) { + spin_lock_irq(&sdev->ipc_lock); + + /* handle immediate reply from DSP core */ + acp_dsp_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, 0); + /* set the done bit */ + acp_dsp_ipc_dsp_done(sdev); + + spin_unlock_irq(&sdev->ipc_lock); + } else { + dev_dbg_ratelimited(sdev->dev, "IPC reply before FW_BOOT_COMPLETE: %#x\n", + dsp_ack); + } + ipc_irq = true; } From ccc8480d90e8cb60f06bd90e227f34784927e19f Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Fri, 7 Feb 2025 13:46:05 +0200 Subject: [PATCH 30/41] ASoC: SOF: amd: Add branch prediction hint in ACP IRQ handler The conditional involving sdev->first_boot in acp_sof_ipc_irq_thread() will succeed only once, i.e. during the very first run of the DSP firmware. Use the unlikely() annotation to help improve branch prediction accuracy. Signed-off-by: Cristian Ciocaltea Reviewed-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20250207-sof-vangogh-fixes-v1-4-67824c1e4c9a@collabora.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c index 12caefd08788..22d4b807e1bb 100644 --- a/sound/soc/sof/amd/acp-ipc.c +++ b/sound/soc/sof/amd/acp-ipc.c @@ -165,7 +165,7 @@ irqreturn_t acp_sof_ipc_irq_thread(int irq, void *context) int dsp_msg, dsp_ack; unsigned int status; - if (sdev->first_boot && sdev->fw_state != SOF_FW_BOOT_COMPLETE) { + if (unlikely(sdev->first_boot && sdev->fw_state != SOF_FW_BOOT_COMPLETE)) { acp_mailbox_read(sdev, sdev->dsp_box.offset, &status, sizeof(status)); if ((status & SOF_IPC_PANIC_MAGIC_MASK) == SOF_IPC_PANIC_MAGIC) { From b19181638182d1f5c43757b471c056b6196c8ca3 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 10 Feb 2025 16:32:50 +0000 Subject: [PATCH 31/41] ASoC: cs35l41: Fix acpi_device_hid() not found MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Function acpi_device_hid() is only defined if CONFIG_ACPI is set. Use #ifdef CONFIG_ACPI to ensure that cs35l41 driver only calls this function is CONFIG_ACPI is define. Fixes: 1d44a30ae3f9 ("ASoC: cs35l41: Fallback to using HID for system_name if no SUB is available") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202502090100.SbXmGFqs-lkp@intel.com/ Signed-off-by: Stefan Binding Reviewed-by: André Almeida Link: https://patch.msgid.link/20250210163256.1722350-1-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 30b89018b113..ff4134bee858 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -1148,6 +1148,7 @@ static int cs35l41_dsp_init(struct cs35l41_private *cs35l41) return ret; } +#ifdef CONFIG_ACPI static int cs35l41_acpi_get_name(struct cs35l41_private *cs35l41) { struct acpi_device *adev = ACPI_COMPANION(cs35l41->dev); @@ -1180,6 +1181,12 @@ static int cs35l41_acpi_get_name(struct cs35l41_private *cs35l41) return 0; } +#else +static int cs35l41_acpi_get_name(struct cs35l41_private *cs35l41) +{ + return 0; +} +#endif /* CONFIG_ACPI */ int cs35l41_probe(struct cs35l41_private *cs35l41, const struct cs35l41_hw_cfg *hw_cfg) { From 2afd96a4a0b1d62c7a44227e535b073926d73368 Mon Sep 17 00:00:00 2001 From: Baojun Xu Date: Tue, 11 Feb 2025 16:39:41 +0800 Subject: [PATCH 32/41] ALSA: hda/tas2781: Update tas2781 hda SPI driver Because firmware issue of platform, found spi device is not stable, so add status check before firmware download, and remove some operations which is not must in current stage. Signed-off-by: Baojun Xu Fixes: bb5f86ea50ff ("ALSA: hda/tas2781: Add tas2781 hda SPI driver") Link: https://patch.msgid.link/20250211083941.5574-1-baojun.xu@ti.com Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_spi.c | 27 ++++++++++++++------------- 1 file changed, 14 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c index a42fa990e7b9..04db80af53c0 100644 --- a/sound/pci/hda/tas2781_hda_spi.c +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -912,7 +912,7 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) struct tasdevice_priv *tas_priv = context; struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); struct hda_codec *codec = tas_priv->codec; - int i, j, ret; + int i, j, ret, val; pm_runtime_get_sync(tas_priv->dev); guard(mutex)(&tas_priv->codec_lock); @@ -981,13 +981,16 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) /* Perform AMP reset before firmware download. */ tas_priv->rcabin.profile_cfg_id = TAS2781_PRE_POST_RESET_CFG; - tasdevice_spi_tuning_switch(tas_priv, 0); tas2781_spi_reset(tas_priv); tas_priv->rcabin.profile_cfg_id = 0; - tasdevice_spi_tuning_switch(tas_priv, 1); tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - ret = tasdevice_spi_prmg_load(tas_priv, 0); + ret = tasdevice_spi_dev_read(tas_priv, TAS2781_REG_CLK_CONFIG, &val); + if (ret < 0) + goto out; + + if (val == TAS2781_REG_CLK_CONFIG_RESET) + ret = tasdevice_spi_prmg_load(tas_priv, 0); if (ret < 0) { dev_err(tas_priv->dev, "FW download failed = %d\n", ret); goto out; @@ -1001,7 +1004,6 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) * If calibrated data occurs error, dsp will still works with default * calibrated data inside algo. */ - tas_priv->save_calibration(tas_priv); out: if (fmw) @@ -1160,7 +1162,8 @@ static int tas2781_runtime_suspend(struct device *dev) guard(mutex)(&tas_hda->priv->codec_lock); - tasdevice_spi_tuning_switch(tas_hda->priv, 1); + if (tas_hda->priv->playback_started) + tasdevice_spi_tuning_switch(tas_hda->priv, 1); tas_hda->priv->cur_book = -1; tas_hda->priv->cur_conf = -1; @@ -1174,7 +1177,8 @@ static int tas2781_runtime_resume(struct device *dev) guard(mutex)(&tas_hda->priv->codec_lock); - tasdevice_spi_tuning_switch(tas_hda->priv, 0); + if (tas_hda->priv->playback_started) + tasdevice_spi_tuning_switch(tas_hda->priv, 0); return 0; } @@ -1189,12 +1193,9 @@ static int tas2781_system_suspend(struct device *dev) return ret; /* Shutdown chip before system suspend */ - tasdevice_spi_tuning_switch(tas_hda->priv, 1); - tas2781_spi_reset(tas_hda->priv); - /* - * Reset GPIO may be shared, so cannot reset here. - * However beyond this point, amps may be powered down. - */ + if (tas_hda->priv->playback_started) + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + return 0; } From 174448badb4409491bfba2e6b46f7aa078741c5e Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 12 Feb 2025 14:40:46 +0800 Subject: [PATCH 33/41] ALSA: hda/realtek: Fixup ALC225 depop procedure Headset MIC will no function when power_save=0. Fixes: 1fd50509fe14 ("ALSA: hda/realtek: Update ALC225 depop procedure") Link: https://bugzilla.kernel.org/show_bug.cgi?id=219743 Signed-off-by: Kailang Yang Link: https://lore.kernel.org/0474a095ab0044d0939ec4bf4362423d@realtek.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 ae0beb52e7b0..224616fbec4f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3788,6 +3788,7 @@ static void alc225_init(struct hda_codec *codec) AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); msleep(75); + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x4); /* Hight power */ } } From 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 13 Feb 2025 15:05:18 +0800 Subject: [PATCH 34/41] ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device When defer probe happens, there may be below error: platform 59820000.sai: Resources present before probing The cpu_mclk clock is from the cpu dai device, if it is not released, then the cpu dai device probe will fail for the second time. The cpu_mclk is used to get rate for rate constraint, rate constraint may be specific for each platform, which is not necessary for machine driver, so remove it. Fixes: b86ef5367761 ("ASoC: fsl: Add Audio Mixer machine driver") Signed-off-by: Shengjiu Wang Link: https://patch.msgid.link/20250213070518.547375-1-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmix.c | 31 ------------------------------- 1 file changed, 31 deletions(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 231400661c90..50ecc5f51100 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -23,7 +23,6 @@ struct imx_audmix { struct snd_soc_card card; struct platform_device *audmix_pdev; struct platform_device *out_pdev; - struct clk *cpu_mclk; int num_dai; struct snd_soc_dai_link *dai; int num_dai_conf; @@ -32,34 +31,11 @@ struct imx_audmix { struct snd_soc_dapm_route *dapm_routes; }; -static const u32 imx_audmix_rates[] = { - 8000, 12000, 16000, 24000, 32000, 48000, 64000, 96000, -}; - -static const struct snd_pcm_hw_constraint_list imx_audmix_rate_constraints = { - .count = ARRAY_SIZE(imx_audmix_rates), - .list = imx_audmix_rates, -}; - static int imx_audmix_fe_startup(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct imx_audmix *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_pcm_runtime *runtime = substream->runtime; - struct device *dev = rtd->card->dev; - unsigned long clk_rate = clk_get_rate(priv->cpu_mclk); int ret; - if (clk_rate % 24576000 == 0) { - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &imx_audmix_rate_constraints); - if (ret < 0) - return ret; - } else { - dev_warn(dev, "mclk may be not supported %lu\n", clk_rate); - } - ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 1, 8); if (ret < 0) @@ -323,13 +299,6 @@ static int imx_audmix_probe(struct platform_device *pdev) } put_device(&cpu_pdev->dev); - priv->cpu_mclk = devm_clk_get(&cpu_pdev->dev, "mclk1"); - if (IS_ERR(priv->cpu_mclk)) { - ret = PTR_ERR(priv->cpu_mclk); - dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret); - return ret; - } - priv->audmix_pdev = audmix_pdev; priv->out_pdev = cpu_pdev; From 325735e83d7d0016e7b61069df2570e910898466 Mon Sep 17 00:00:00 2001 From: Baojun Xu Date: Fri, 14 Feb 2025 09:30:21 +0800 Subject: [PATCH 35/41] ALSA: hda/tas2781: Fix index issue in tas2781 hda SPI driver Correct wrong mask for device index. Signed-off-by: Baojun Xu Fixes: bb5f86ea50ff ("ALSA: hda/tas2781: Add tas2781 hda SPI driver") Link: https://patch.msgid.link/20250214013021.6072-1-baojun.xu@ti.com Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_spi_fwlib.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c index 0e2acbc3c900..131d9a77d140 100644 --- a/sound/pci/hda/tas2781_spi_fwlib.c +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -2,7 +2,7 @@ // // TAS2781 HDA SPI driver // -// Copyright 2024 Texas Instruments, Inc. +// Copyright 2024-2025 Texas Instruments, Inc. // // Author: Baojun Xu @@ -771,19 +771,19 @@ static int tasdevice_process_block(void *context, unsigned char *data, switch (subblk_typ) { case TASDEVICE_CMD_SING_W: subblk_offset = tasdevice_single_byte_wr(tas_priv, - dev_idx & 0x4f, data, sublocksize); + dev_idx & 0x3f, data, sublocksize); break; case TASDEVICE_CMD_BURST: subblk_offset = tasdevice_burst_wr(tas_priv, - dev_idx & 0x4f, data, sublocksize); + dev_idx & 0x3f, data, sublocksize); break; case TASDEVICE_CMD_DELAY: subblk_offset = tasdevice_delay(tas_priv, - dev_idx & 0x4f, data, sublocksize); + dev_idx & 0x3f, data, sublocksize); break; case TASDEVICE_CMD_FIELD_W: subblk_offset = tasdevice_field_wr(tas_priv, - dev_idx & 0x4f, data, sublocksize); + dev_idx & 0x3f, data, sublocksize); break; default: subblk_offset = 2; From 822b7ec657e99b44b874e052d8540d8b54fe8569 Mon Sep 17 00:00:00 2001 From: Wentao Liang Date: Thu, 13 Feb 2025 15:45:43 +0800 Subject: [PATCH 36/41] ALSA: hda: Add error check for snd_ctl_rename_id() in snd_hda_create_dig_out_ctls() Check the return value of snd_ctl_rename_id() in snd_hda_create_dig_out_ctls(). Ensure that failures are properly handled. [ Note: the error cannot happen practically because the only error condition in snd_ctl_rename_id() is the missing ID, but this is a rename, hence it must be present. But for the code consistency, it's safer to have always the proper return check -- tiwai ] Fixes: 5c219a340850 ("ALSA: hda: Fix kctl->id initialization") Cc: stable@vger.kernel.org # 6.4+ Signed-off-by: Wentao Liang Link: https://patch.msgid.link/20250213074543.1620-1-vulab@iscas.ac.cn Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 14763c0f31ad..46a220404999 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2470,7 +2470,9 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec, break; id = kctl->id; id.index = spdif_index; - snd_ctl_rename_id(codec->card, &kctl->id, &id); + err = snd_ctl_rename_id(codec->card, &kctl->id, &id); + if (err < 0) + return err; } bus->primary_dig_out_type = HDA_PCM_TYPE_HDMI; } From 362ff1e7c6c20f8d6ebe20682870d471373c608b Mon Sep 17 00:00:00 2001 From: Stefano Garzarella Date: Thu, 13 Feb 2025 17:18:25 +0100 Subject: [PATCH 37/41] virtio_snd.h: clarify that `controls` depends on VIRTIO_SND_F_CTLS MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit As defined in the specification, the `controls` field in the configuration space is only valid/present if VIRTIO_SND_F_CTLS is negotiated. From https://docs.oasis-open.org/virtio/virtio/v1.3/virtio-v1.3.html: 5.14.4 Device Configuration Layout ... controls (driver-read-only) indicates a total number of all available control elements if VIRTIO_SND_F_CTLS has been negotiated. Let's use the same style used in virtio_blk.h to clarify this and to avoid confusion as happened in QEMU (see link). Link: https://gitlab.com/qemu-project/qemu/-/issues/2805 Signed-off-by: Stefano Garzarella Acked-by: Eugenio Pérez Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20250213161825.139952-1-sgarzare@redhat.com --- include/uapi/linux/virtio_snd.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/linux/virtio_snd.h b/include/uapi/linux/virtio_snd.h index 5f4100c2cf04..a4cfb9f6561a 100644 --- a/include/uapi/linux/virtio_snd.h +++ b/include/uapi/linux/virtio_snd.h @@ -25,7 +25,7 @@ struct virtio_snd_config { __le32 streams; /* # of available channel maps */ __le32 chmaps; - /* # of available control elements */ + /* # of available control elements (if VIRTIO_SND_F_CTLS) */ __le32 controls; }; From 08b613b9e2ba431db3bd15cb68ca72472a50ef5c Mon Sep 17 00:00:00 2001 From: Vitaly Rodionov Date: Fri, 14 Feb 2025 21:07:28 +0000 Subject: [PATCH 38/41] ALSA: hda/cirrus: Correct the full scale volume set logic This patch corrects the full-scale volume setting logic. On certain platforms, the full-scale volume bit is required. The current logic mistakenly sets this bit and incorrectly clears reserved bit 0, causing the headphone output to be muted. Fixes: 342b6b610ae2 ("ALSA: hda/cs8409: Fix Full Scale Volume setting for all variants") Signed-off-by: Vitaly Rodionov Link: https://patch.msgid.link/20250214210736.30814-1-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409-tables.c | 6 +++--- sound/pci/hda/patch_cs8409.c | 20 +++++++++++--------- sound/pci/hda/patch_cs8409.h | 5 +++-- 3 files changed, 17 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index 759f48038273..621f947e3817 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -121,7 +121,7 @@ static const struct cs8409_i2c_param cs42l42_init_reg_seq[] = { { CS42L42_MIXER_CHA_VOL, 0x3F }, { CS42L42_MIXER_CHB_VOL, 0x3F }, { CS42L42_MIXER_ADC_VOL, 0x3f }, - { CS42L42_HP_CTL, 0x03 }, + { CS42L42_HP_CTL, 0x0D }, { CS42L42_MIC_DET_CTL1, 0xB6 }, { CS42L42_TIPSENSE_CTL, 0xC2 }, { CS42L42_HS_CLAMP_DISABLE, 0x01 }, @@ -315,7 +315,7 @@ static const struct cs8409_i2c_param dolphin_c0_init_reg_seq[] = { { CS42L42_ASP_TX_SZ_EN, 0x01 }, { CS42L42_PWR_CTL1, 0x0A }, { CS42L42_PWR_CTL2, 0x84 }, - { CS42L42_HP_CTL, 0x03 }, + { CS42L42_HP_CTL, 0x0D }, { CS42L42_MIXER_CHA_VOL, 0x3F }, { CS42L42_MIXER_CHB_VOL, 0x3F }, { CS42L42_MIXER_ADC_VOL, 0x3f }, @@ -371,7 +371,7 @@ static const struct cs8409_i2c_param dolphin_c1_init_reg_seq[] = { { CS42L42_ASP_TX_SZ_EN, 0x00 }, { CS42L42_PWR_CTL1, 0x0E }, { CS42L42_PWR_CTL2, 0x84 }, - { CS42L42_HP_CTL, 0x01 }, + { CS42L42_HP_CTL, 0x0D }, { CS42L42_MIXER_CHA_VOL, 0x3F }, { CS42L42_MIXER_CHB_VOL, 0x3F }, { CS42L42_MIXER_ADC_VOL, 0x3f }, diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 614327218634..b760332a4e35 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -876,7 +876,7 @@ static void cs42l42_resume(struct sub_codec *cs42l42) { CS42L42_DET_INT_STATUS2, 0x00 }, { CS42L42_TSRS_PLUG_STATUS, 0x00 }, }; - int fsv_old, fsv_new; + unsigned int fsv; /* Bring CS42L42 out of Reset */ spec->gpio_data = snd_hda_codec_read(codec, CS8409_PIN_AFG, 0, AC_VERB_GET_GPIO_DATA, 0); @@ -893,13 +893,15 @@ static void cs42l42_resume(struct sub_codec *cs42l42) /* Clear interrupts, by reading interrupt status registers */ cs8409_i2c_bulk_read(cs42l42, irq_regs, ARRAY_SIZE(irq_regs)); - fsv_old = cs8409_i2c_read(cs42l42, CS42L42_HP_CTL); - if (cs42l42->full_scale_vol == CS42L42_FULL_SCALE_VOL_0DB) - fsv_new = fsv_old & ~CS42L42_FULL_SCALE_VOL_MASK; - else - fsv_new = fsv_old & CS42L42_FULL_SCALE_VOL_MASK; - if (fsv_new != fsv_old) - cs8409_i2c_write(cs42l42, CS42L42_HP_CTL, fsv_new); + fsv = cs8409_i2c_read(cs42l42, CS42L42_HP_CTL); + if (cs42l42->full_scale_vol) { + // Set the full scale volume bit + fsv |= CS42L42_FULL_SCALE_VOL_MASK; + cs8409_i2c_write(cs42l42, CS42L42_HP_CTL, fsv); + } + // Unmute analog channels A and B + fsv = (fsv & ~CS42L42_ANA_MUTE_AB); + cs8409_i2c_write(cs42l42, CS42L42_HP_CTL, fsv); /* we have to explicitly allow unsol event handling even during the * resume phase so that the jack event is processed properly @@ -920,7 +922,7 @@ static void cs42l42_suspend(struct sub_codec *cs42l42) { CS42L42_MIXER_CHA_VOL, 0x3F }, { CS42L42_MIXER_ADC_VOL, 0x3F }, { CS42L42_MIXER_CHB_VOL, 0x3F }, - { CS42L42_HP_CTL, 0x0F }, + { CS42L42_HP_CTL, 0x0D }, { CS42L42_ASP_RX_DAI0_EN, 0x00 }, { CS42L42_ASP_CLK_CFG, 0x00 }, { CS42L42_PWR_CTL1, 0xFE }, diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h index 5e48115caf09..14645d25e70f 100644 --- a/sound/pci/hda/patch_cs8409.h +++ b/sound/pci/hda/patch_cs8409.h @@ -230,9 +230,10 @@ enum cs8409_coefficient_index_registers { #define CS42L42_PDN_TIMEOUT_US (250000) #define CS42L42_PDN_SLEEP_US (2000) #define CS42L42_INIT_TIMEOUT_MS (45) +#define CS42L42_ANA_MUTE_AB (0x0C) #define CS42L42_FULL_SCALE_VOL_MASK (2) -#define CS42L42_FULL_SCALE_VOL_0DB (1) -#define CS42L42_FULL_SCALE_VOL_MINUS6DB (0) +#define CS42L42_FULL_SCALE_VOL_0DB (0) +#define CS42L42_FULL_SCALE_VOL_MINUS6DB (1) /* Dell BULLSEYE / WARLOCK / CYBORG Specific Definitions */ From 6a7ed7ee16a963f0ca028861eca8f8b365861dd1 Mon Sep 17 00:00:00 2001 From: Vitaly Rodionov Date: Fri, 14 Feb 2025 16:23:26 +0000 Subject: [PATCH 39/41] ALSA: hda/cirrus: Reduce codec resume time This patch reduces the resume time by half and introduces an option to include a delay after a single write operation before continuing. Signed-off-by: Vitaly Rodionov Link: https://patch.msgid.link/20250214162354.2675652-2-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409-tables.c | 6 +++--- sound/pci/hda/patch_cs8409.c | 6 +++++- sound/pci/hda/patch_cs8409.h | 2 +- 3 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index 621f947e3817..09240138e087 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -131,7 +131,7 @@ static const struct cs8409_i2c_param cs42l42_init_reg_seq[] = { { CS42L42_RSENSE_CTL3, 0x00 }, { CS42L42_TSENSE_CTL, 0x80 }, { CS42L42_HS_BIAS_CTL, 0xC0 }, - { CS42L42_PWR_CTL1, 0x02 }, + { CS42L42_PWR_CTL1, 0x02, 10000 }, { CS42L42_ADC_OVFL_INT_MASK, 0xff }, { CS42L42_MIXER_INT_MASK, 0xff }, { CS42L42_SRC_INT_MASK, 0xff }, @@ -328,7 +328,7 @@ static const struct cs8409_i2c_param dolphin_c0_init_reg_seq[] = { { CS42L42_RSENSE_CTL3, 0x00 }, { CS42L42_TSENSE_CTL, 0x80 }, { CS42L42_HS_BIAS_CTL, 0xC0 }, - { CS42L42_PWR_CTL1, 0x02 }, + { CS42L42_PWR_CTL1, 0x02, 10000 }, { CS42L42_ADC_OVFL_INT_MASK, 0xff }, { CS42L42_MIXER_INT_MASK, 0xff }, { CS42L42_SRC_INT_MASK, 0xff }, @@ -384,7 +384,7 @@ static const struct cs8409_i2c_param dolphin_c1_init_reg_seq[] = { { CS42L42_RSENSE_CTL3, 0x00 }, { CS42L42_TSENSE_CTL, 0x80 }, { CS42L42_HS_BIAS_CTL, 0xC0 }, - { CS42L42_PWR_CTL1, 0x06 }, + { CS42L42_PWR_CTL1, 0x06, 10000 }, { CS42L42_ADC_OVFL_INT_MASK, 0xff }, { CS42L42_MIXER_INT_MASK, 0xff }, { CS42L42_SRC_INT_MASK, 0xff }, diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index b760332a4e35..e50006757a2c 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -346,6 +346,11 @@ static int cs8409_i2c_bulk_write(struct sub_codec *scodec, const struct cs8409_i if (cs8409_i2c_wait_complete(codec) < 0) goto error; + /* Certain use cases may require a delay + * after a write operation before proceeding. + */ + if (seq[i].delay) + fsleep(seq[i].delay); } mutex_unlock(&spec->i2c_mux); @@ -888,7 +893,6 @@ static void cs42l42_resume(struct sub_codec *cs42l42) /* Initialize CS42L42 companion codec */ cs8409_i2c_bulk_write(cs42l42, cs42l42->init_seq, cs42l42->init_seq_num); - msleep(CS42L42_INIT_TIMEOUT_MS); /* Clear interrupts, by reading interrupt status registers */ cs8409_i2c_bulk_read(cs42l42, irq_regs, ARRAY_SIZE(irq_regs)); diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h index 14645d25e70f..e4bd2e12110b 100644 --- a/sound/pci/hda/patch_cs8409.h +++ b/sound/pci/hda/patch_cs8409.h @@ -229,7 +229,6 @@ enum cs8409_coefficient_index_registers { #define CS42L42_I2C_SLEEP_US (2000) #define CS42L42_PDN_TIMEOUT_US (250000) #define CS42L42_PDN_SLEEP_US (2000) -#define CS42L42_INIT_TIMEOUT_MS (45) #define CS42L42_ANA_MUTE_AB (0x0C) #define CS42L42_FULL_SCALE_VOL_MASK (2) #define CS42L42_FULL_SCALE_VOL_0DB (0) @@ -291,6 +290,7 @@ enum { struct cs8409_i2c_param { unsigned int addr; unsigned int value; + unsigned int delay; }; struct cs8409_cir_param { From 6d1f86610f23b0bc334d6506a186f21a98f51392 Mon Sep 17 00:00:00 2001 From: John Veness Date: Mon, 17 Feb 2025 12:15:50 +0000 Subject: [PATCH 40/41] ALSA: hda/conexant: Add quirk for HP ProBook 450 G4 mute LED Allows the LED on the dedicated mute button on the HP ProBook 450 G4 laptop to change colour correctly. Signed-off-by: John Veness Cc: Link: https://patch.msgid.link/2fb55d48-6991-4a42-b591-4c78f2fad8d7@pelago.org.uk Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4985e72b9094..34874039ad45 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1090,6 +1090,7 @@ static const struct hda_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x8231, "HP ProBook 450 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), From e77aa4b2eaa7fb31b2a7a50214ecb946b2a8b0f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 17 Feb 2025 18:00:30 +0100 Subject: [PATCH 41/41] ALSA: seq: Drop UMP events when no UMP-conversion is set When a destination client is a user client in the legacy MIDI mode and it sets the no-UMP-conversion flag, currently the all UMP events are still passed as-is. But this may confuse the user-space, because the event packet size is different from the legacy mode. Since we cannot handle UMP events in user clients unless it's running in the UMP client mode, we should filter out those events instead of accepting blindly. This patch addresses it by slightly adjusting the conditions for UMP event handling at the event delivery time. Fixes: 329ffe11a014 ("ALSA: seq: Allow suppressing UMP conversions") Link: https://lore.kernel.org/b77a2cd6-7b59-4eb0-a8db-22d507d3af5f@gmail.com Link: https://patch.msgid.link/20250217170034.21930-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 073b56dc2225..cb66ec42a3f8 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -678,12 +678,18 @@ static int snd_seq_deliver_single_event(struct snd_seq_client *client, dest_port->time_real); #if IS_ENABLED(CONFIG_SND_SEQ_UMP) - if (!(dest->filter & SNDRV_SEQ_FILTER_NO_CONVERT)) { - if (snd_seq_ev_is_ump(event)) { + if (snd_seq_ev_is_ump(event)) { + if (!(dest->filter & SNDRV_SEQ_FILTER_NO_CONVERT)) { result = snd_seq_deliver_from_ump(client, dest, dest_port, event, atomic, hop); goto __skip; - } else if (snd_seq_client_is_ump(dest)) { + } else if (dest->type == USER_CLIENT && + !snd_seq_client_is_ump(dest)) { + result = 0; // drop the event + goto __skip; + } + } else if (snd_seq_client_is_ump(dest)) { + if (!(dest->filter & SNDRV_SEQ_FILTER_NO_CONVERT)) { result = snd_seq_deliver_to_ump(client, dest, dest_port, event, atomic, hop); goto __skip;