From d7e2447a4d51de5c3c03e3b7892898e98ddd9769 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 10 Feb 2025 10:17:30 +0200 Subject: [PATCH 01/16] 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 */ -- 2.51.0 From 70e90680c2592c38c62e5716f1296a2d74bae7af Mon Sep 17 00:00:00 2001 From: Nam Cao Date: Wed, 5 Feb 2025 11:46:33 +0100 Subject: [PATCH 02/16] 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; -- 2.51.0 From 78ccf6a6bae11e451e20a52dd2bc2ab98f66326b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Feb 2025 11:19:53 +0800 Subject: [PATCH 03/16] 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), -- 2.51.0 From cb78b8dc7834066539253c039f276b3625fecd9f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Feb 2025 11:19:54 +0800 Subject: [PATCH 04/16] 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" }, {}, }; -- 2.51.0 From 91b98d5a6e8067c5226207487681a48f0d651e46 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Fri, 7 Feb 2025 13:46:02 +0200 Subject: [PATCH 05/16] 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; -- 2.51.0 From 2ecbc2e9f3b19e2199e8bc3ba603d299f1985f09 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Fri, 7 Feb 2025 13:46:03 +0200 Subject: [PATCH 06/16] 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 -- 2.51.0 From ac84ca815adb4171a4276b1d44096b75f6a150b7 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Fri, 7 Feb 2025 13:46:04 +0200 Subject: [PATCH 07/16] 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; } -- 2.51.0 From ccc8480d90e8cb60f06bd90e227f34784927e19f Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Fri, 7 Feb 2025 13:46:05 +0200 Subject: [PATCH 08/16] 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) { -- 2.51.0 From b19181638182d1f5c43757b471c056b6196c8ca3 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 10 Feb 2025 16:32:50 +0000 Subject: [PATCH 09/16] ASoC: cs35l41: Fix acpi_device_hid() not found MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 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 @@ err_dsp: 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) { -- 2.51.0 From 2afd96a4a0b1d62c7a44227e535b073926d73368 Mon Sep 17 00:00:00 2001 From: Baojun Xu Date: Tue, 11 Feb 2025 16:39:41 +0800 Subject: [PATCH 10/16] 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; } -- 2.51.0 From 174448badb4409491bfba2e6b46f7aa078741c5e Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 12 Feb 2025 14:40:46 +0800 Subject: [PATCH 11/16] 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 */ } } -- 2.51.0 From 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 13 Feb 2025 15:05:18 +0800 Subject: [PATCH 12/16] 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; -- 2.51.0 From 325735e83d7d0016e7b61069df2570e910898466 Mon Sep 17 00:00:00 2001 From: Baojun Xu Date: Fri, 14 Feb 2025 09:30:21 +0800 Subject: [PATCH 13/16] 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; -- 2.51.0 From 822b7ec657e99b44b874e052d8540d8b54fe8569 Mon Sep 17 00:00:00 2001 From: Wentao Liang Date: Thu, 13 Feb 2025 15:45:43 +0800 Subject: [PATCH 14/16] 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; } -- 2.51.0 From 362ff1e7c6c20f8d6ebe20682870d471373c608b Mon Sep 17 00:00:00 2001 From: Stefano Garzarella Date: Thu, 13 Feb 2025 17:18:25 +0100 Subject: [PATCH 15/16] virtio_snd.h: clarify that `controls` depends on VIRTIO_SND_F_CTLS MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 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; }; -- 2.51.0 From 08b613b9e2ba431db3bd15cb68ca72472a50ef5c Mon Sep 17 00:00:00 2001 From: Vitaly Rodionov Date: Fri, 14 Feb 2025 21:07:28 +0000 Subject: [PATCH 16/16] 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 */ -- 2.51.0