From 49da1463c9e3d2082276c3e0e2a8b65a88711cd2 Mon Sep 17 00:00:00 2001 From: Zichen Xie Date: Sun, 6 Oct 2024 15:57:37 -0500 Subject: [PATCH 01/52] ASoC: qcom: Fix NULL Dereference in asoc_qcom_lpass_cpu_platform_probe() A devm_kzalloc() in asoc_qcom_lpass_cpu_platform_probe() could possibly return NULL pointer. NULL Pointer Dereference may be triggerred without addtional check. Add a NULL check for the returned pointer. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Cc: stable@vger.kernel.org Signed-off-by: Zichen Xie Link: https://patch.msgid.link/20241006205737.8829-1-zichenxie0106@gmail.com Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 5a47f661e0c6..242bc16da36d 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1242,6 +1242,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) /* Allocation for i2sctl regmap fields */ drvdata->i2sctl = devm_kzalloc(&pdev->dev, sizeof(struct lpaif_i2sctl), GFP_KERNEL); + if (!drvdata->i2sctl) + return -ENOMEM; /* Initialize bitfields for dai I2SCTL register */ ret = lpass_cpu_init_i2sctl_bitfields(dev, drvdata->i2sctl, From 8380dbf1b9ef66e3ce6c1d660fd7259637c2a929 Mon Sep 17 00:00:00 2001 From: Miquel Raynal Date: Thu, 3 Oct 2024 10:36:11 +0200 Subject: [PATCH 02/52] ASoC: dt-bindings: davinci-mcasp: Fix interrupt properties MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Combinations of "tx" alone, "rx" alone and "tx", "rx" together are supposedly valid (see link below), which is not the case today as "rx" alone is not accepted by the current binding. Let's rework the two interrupt properties to expose all correct possibilities. Cc: Péter Ujfalusi Link: https://lore.kernel.org/linux-sound/20241003102552.2c11840e@xps-13/T/#m277fce1d49c50d94e071f7890aed472fa2c64052 Fixes: 8be90641a0bb ("ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema") Signed-off-by: Miquel Raynal Acked-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20241003083611.461894-1-miquel.raynal@bootlin.com Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.yaml | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml index ab3206ffa4af..beef193aaaeb 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -102,21 +102,21 @@ properties: default: 2 interrupts: - oneOf: - - minItems: 1 - items: - - description: TX interrupt - - description: RX interrupt - - items: - - description: common/combined interrupt + minItems: 1 + maxItems: 2 interrupt-names: oneOf: - - minItems: 1 + - description: TX interrupt + const: tx + - description: RX interrupt + const: rx + - description: TX and RX interrupts items: - const: tx - const: rx - - const: common + - description: Common/combined interrupt + const: common fck_parent: $ref: /schemas/types.yaml#/definitions/string From 0dbb186c3510cad4e9f443e801bf2e6ab5770c00 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Tue, 8 Oct 2024 10:37:58 +0200 Subject: [PATCH 03/52] ASoC: Intel: avs: Update stream status in a separate thread MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Function snd_pcm_period_elapsed() is part of sequence servicing HDAudio stream IRQs. It's called under Global Interrupt Enable (GIE) disabled - no HDAudio interrupts will be raised. At the same time, the function may end up calling __snd_pcm_xrun() or snd_pcm_drain_done(). On the avs-driver side, this translates to IPCs and as GIE is disabled, these will never complete successfully. Improve system stability by scheduling stream-IRQ handling in a separate thread. Signed-off-by: Amadeusz Sławiński Reviewed-by: Cezary Rojewski Link: https://patch.msgid.link/20241008083758.756578-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/core.c | 3 ++- sound/soc/intel/avs/pcm.c | 19 +++++++++++++++++++ sound/soc/intel/avs/pcm.h | 16 ++++++++++++++++ 3 files changed, 37 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/avs/pcm.h diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index da7bac09acb4..73d4bde9b2f7 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -28,6 +28,7 @@ #include "avs.h" #include "cldma.h" #include "messages.h" +#include "pcm.h" static u32 pgctl_mask = AZX_PGCTL_LSRMD_MASK; module_param(pgctl_mask, uint, 0444); @@ -247,7 +248,7 @@ static void hdac_stream_update_pos(struct hdac_stream *stream, u64 buffer_size) static void hdac_update_stream(struct hdac_bus *bus, struct hdac_stream *stream) { if (stream->substream) { - snd_pcm_period_elapsed(stream->substream); + avs_period_elapsed(stream->substream); } else if (stream->cstream) { u64 buffer_size = stream->cstream->runtime->buffer_size; diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index afc0fc74cf94..4af811580356 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -16,6 +16,7 @@ #include #include "avs.h" #include "path.h" +#include "pcm.h" #include "topology.h" #include "../../codecs/hda.h" @@ -30,6 +31,7 @@ struct avs_dma_data { struct hdac_ext_stream *host_stream; }; + struct work_struct period_elapsed_work; struct snd_pcm_substream *substream; }; @@ -56,6 +58,22 @@ avs_dai_find_path_template(struct snd_soc_dai *dai, bool is_fe, int direction) return dw->priv; } +static void avs_period_elapsed_work(struct work_struct *work) +{ + struct avs_dma_data *data = container_of(work, struct avs_dma_data, period_elapsed_work); + + snd_pcm_period_elapsed(data->substream); +} + +void avs_period_elapsed(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_dai *dai = snd_soc_rtd_to_cpu(rtd, 0); + struct avs_dma_data *data = snd_soc_dai_get_dma_data(dai, substream); + + schedule_work(&data->period_elapsed_work); +} + static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); @@ -77,6 +95,7 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d data->substream = substream; data->template = template; data->adev = adev; + INIT_WORK(&data->period_elapsed_work, avs_period_elapsed_work); snd_soc_dai_set_dma_data(dai, substream, data); if (rtd->dai_link->ignore_suspend) diff --git a/sound/soc/intel/avs/pcm.h b/sound/soc/intel/avs/pcm.h new file mode 100644 index 000000000000..0f3615c90398 --- /dev/null +++ b/sound/soc/intel/avs/pcm.h @@ -0,0 +1,16 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright(c) 2024 Intel Corporation + * + * Authors: Cezary Rojewski + * Amadeusz Slawinski + */ + +#ifndef __SOUND_SOC_INTEL_AVS_PCM_H +#define __SOUND_SOC_INTEL_AVS_PCM_H + +#include + +void avs_period_elapsed(struct snd_pcm_substream *substream); + +#endif From 0a5c40393b123f3f08e428143985ab0c5ddb4d28 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 8 Oct 2024 14:43:44 +0530 Subject: [PATCH 04/52] ASoC: SOF: amd: Add error log for DSP firmware validation failure Add dev_err to print ACP_SHA_DSP_FW_QUALIFIER and ACP_SHA_PSP_ACK register values for PSP firmware validation failure case. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20241008091347.594378-1-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index d579c3849392..de3001f5b9bb 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -329,7 +329,9 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); if (ret < 0) { - dev_err(sdev->dev, "PSP validation failed\n"); + val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_SHA_PSP_ACK); + dev_err(sdev->dev, "PSP validation failed: fw_qualifier = %#x, ACP_SHA_PSP_ACK = %#x\n", + fw_qualifier, val); return ret; } From 494ddacd4a2ae5fd1c46ea49364eaab4fc1e5461 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 8 Oct 2024 14:43:45 +0530 Subject: [PATCH 05/52] ASoC: SOF: amd: Fix for ACP SRAM addr for acp7.0 platform Incorrect SRAM base addr for acp7.0 platform results firmware boot failure. Add condition check to support SRAM addr for various platforms. Fixes: 145d7e5ae8f4 ("ASoC: SOF: amd: add option to use sram for data bin loading") Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20241008091347.594378-2-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-loader.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp-loader.c b/sound/soc/sof/amd/acp-loader.c index 19f10dd77e4b..077af9e2af8d 100644 --- a/sound/soc/sof/amd/acp-loader.c +++ b/sound/soc/sof/amd/acp-loader.c @@ -206,7 +206,10 @@ int acp_dsp_pre_fw_run(struct snd_sof_dev *sdev) configure_pte_for_fw_loading(FW_SRAM_DATA_BIN, ACP_SRAM_PAGE_COUNT, adata); src_addr = ACP_SYSTEM_MEMORY_WINDOW + ACP_DEFAULT_SRAM_LENGTH + (page_count * ACP_PAGE_SIZE); - dest_addr = ACP_SRAM_BASE_ADDRESS; + if (adata->pci_rev > ACP63_PCI_ID) + dest_addr = ACP7X_SRAM_BASE_ADDRESS; + else + dest_addr = ACP_SRAM_BASE_ADDRESS; ret = configure_and_run_dma(adata, src_addr, dest_addr, adata->fw_sram_data_bin_size); From 9814c1447f9cc67c9e88e0a4423de3a496078360 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Tue, 8 Oct 2024 09:07:10 +0300 Subject: [PATCH 06/52] ASoC: SOF: Intel: hda-loader: do not wait for HDaudio IOC MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Commit 9ee3f0d8c999 ("ASOC: SOF: Intel: hda-loader: only wait for HDaudio IOC for IPC4 devices") removed DMA wait for IPC3 case. Proceed and remove the wait for IPC4 devices as well. There is no dependency to IPC version in the load logic and checking the firmware status is a sufficient check in case of errors. The removed code also had a bug in that -ETIMEDOUT is returned without stopping the DMA transfer. Cc: stable@vger.kernel.org Link: https://github.com/thesofproject/linux/issues/5135 Fixes: 9ee3f0d8c999 ("ASOC: SOF: Intel: hda-loader: only wait for HDaudio IOC for IPC4 devices") Suggested-by: Peter Ujfalusi Signed-off-by: Kai Vehmanen Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Link: https://patch.msgid.link/20241008060710.15409-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 17 ----------------- 1 file changed, 17 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 75f6240cf3e1..9d8ebb7c6a10 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -294,14 +294,9 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; - struct sof_intel_hda_stream *hda_stream; - unsigned long time_left; unsigned int reg; int ret, status; - hda_stream = container_of(hext_stream, struct sof_intel_hda_stream, - hext_stream); - dev_dbg(sdev->dev, "Code loader DMA starting\n"); ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_START); @@ -310,18 +305,6 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream return ret; } - if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) { - /* Wait for completion of transfer */ - time_left = wait_for_completion_timeout(&hda_stream->ioc, - msecs_to_jiffies(HDA_CL_DMA_IOC_TIMEOUT_MS)); - - if (!time_left) { - dev_err(sdev->dev, "Code loader DMA did not complete\n"); - return -ETIMEDOUT; - } - dev_dbg(sdev->dev, "Code loader DMA done\n"); - } - dev_dbg(sdev->dev, "waiting for FW_ENTERED status\n"); status = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, From 3fe9f5882cf71573516749b0bb687ef88f470d1d Mon Sep 17 00:00:00 2001 From: Benjamin Bara Date: Tue, 8 Oct 2024 13:36:14 +0200 Subject: [PATCH 07/52] ASoC: dapm: avoid container_of() to get component The current implementation does not work for widgets of DAPMs without component, as snd_soc_dapm_to_component() requires it. If the widget is directly owned by the card, e.g. as it is the case for the tegra implementation, the call leads to UB. Therefore directly access the component of the widget's DAPM to be able to check if a component is available. Fixes: f82eb06a40c8 ("ASoC: tegra: machine: Handle component name prefix") Cc: stable@vger.kernel.org # v6.7+ Signed-off-by: Benjamin Bara Link: https://patch.msgid.link/20241008-tegra-dapm-v2-1-5e999cb5f0e7@skidata.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9330f1a3f758..c34934c31ffe 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2785,10 +2785,10 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_update_dai); int snd_soc_dapm_widget_name_cmp(struct snd_soc_dapm_widget *widget, const char *s) { - struct snd_soc_component *component = snd_soc_dapm_to_component(widget->dapm); + struct snd_soc_component *component = widget->dapm->component; const char *wname = widget->name; - if (component->name_prefix) + if (component && component->name_prefix) wname += strlen(component->name_prefix) + 1; /* plus space */ return strcmp(wname, s); From a6134e7b4d4a14e0942f113a6df1d518baa2a0a4 Mon Sep 17 00:00:00 2001 From: Binbin Zhou Date: Wed, 9 Oct 2024 15:52:27 +0800 Subject: [PATCH 08/52] ASoC: loongson: Fix component check failed on FDT systems Add missing snd_soc_dai_link.platforms assignment to avoid soc_dai_link_sanity_check() failure. Fixes: d24028606e76 ("ASoC: loongson: Add Loongson ASoC Sound Card Support") Signed-off-by: Binbin Zhou Link: https://patch.msgid.link/6645888f2f9e8a1d8d799109f867d0f97fd78c58.1728459624.git.zhoubinbin@loongson.cn Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c index 7379f24d385c..7910d5d9ac4f 100644 --- a/sound/soc/loongson/loongson_card.c +++ b/sound/soc/loongson/loongson_card.c @@ -144,6 +144,7 @@ static int loongson_card_parse_of(struct loongson_card_data *data) dev_err(dev, "getting cpu dlc error (%d)\n", ret); goto err; } + loongson_dai_links[i].platforms->of_node = loongson_dai_links[i].cpus->of_node; ret = snd_soc_of_get_dlc(codec, NULL, loongson_dai_links[i].codecs, 0); if (ret < 0) { From d0e806b0cc6260b59c65e606034a63145169c04c Mon Sep 17 00:00:00 2001 From: Alexey Klimov Date: Wed, 9 Oct 2024 22:39:22 +0100 Subject: [PATCH 09/52] ASoC: qcom: sdm845: add missing soundwire runtime stream alloc During the migration of Soundwire runtime stream allocation from the Qualcomm Soundwire controller to SoC's soundcard drivers the sdm845 soundcard was forgotten. At this point any playback attempt or audio daemon startup, for instance on sdm845-db845c (Qualcomm RB3 board), will result in stream pointer NULL dereference: Unable to handle kernel NULL pointer dereference at virtual address 0000000000000020 Mem abort info: ESR = 0x0000000096000004 EC = 0x25: DABT (current EL), IL = 32 bits SET = 0, FnV = 0 EA = 0, S1PTW = 0 FSC = 0x04: level 0 translation fault Data abort info: ISV = 0, ISS = 0x00000004, ISS2 = 0x00000000 CM = 0, WnR = 0, TnD = 0, TagAccess = 0 GCS = 0, Overlay = 0, DirtyBit = 0, Xs = 0 user pgtable: 4k pages, 48-bit VAs, pgdp=0000000101ecf000 [0000000000000020] pgd=0000000000000000, p4d=0000000000000000 Internal error: Oops: 0000000096000004 [#1] PREEMPT SMP Modules linked in: ... CPU: 5 UID: 0 PID: 1198 Comm: aplay Not tainted 6.12.0-rc2-qcomlt-arm64-00059-g9d78f315a362-dirty #18 Hardware name: Thundercomm Dragonboard 845c (DT) pstate: 60400005 (nZCv daif +PAN -UAO -TCO -DIT -SSBS BTYPE=--) pc : sdw_stream_add_slave+0x44/0x380 [soundwire_bus] lr : sdw_stream_add_slave+0x44/0x380 [soundwire_bus] sp : ffff80008a2035c0 x29: ffff80008a2035c0 x28: ffff80008a203978 x27: 0000000000000000 x26: 00000000000000c0 x25: 0000000000000000 x24: ffff1676025f4800 x23: ffff167600ff1cb8 x22: ffff167600ff1c98 x21: 0000000000000003 x20: ffff167607316000 x19: ffff167604e64e80 x18: 0000000000000000 x17: 0000000000000000 x16: ffffcec265074160 x15: 0000000000000000 x14: 0000000000000000 x13: 0000000000000000 x12: 0000000000000000 x11: 0000000000000000 x10: 0000000000000000 x9 : 0000000000000000 x8 : 0000000000000000 x7 : 0000000000000000 x6 : ffff167600ff1cec x5 : ffffcec22cfa2010 x4 : 0000000000000000 x3 : 0000000000000003 x2 : ffff167613f836c0 x1 : 0000000000000000 x0 : ffff16761feb60b8 Call trace: sdw_stream_add_slave+0x44/0x380 [soundwire_bus] wsa881x_hw_params+0x68/0x80 [snd_soc_wsa881x] snd_soc_dai_hw_params+0x3c/0xa4 __soc_pcm_hw_params+0x230/0x660 dpcm_be_dai_hw_params+0x1d0/0x3f8 dpcm_fe_dai_hw_params+0x98/0x268 snd_pcm_hw_params+0x124/0x460 snd_pcm_common_ioctl+0x998/0x16e8 snd_pcm_ioctl+0x34/0x58 __arm64_sys_ioctl+0xac/0xf8 invoke_syscall+0x48/0x104 el0_svc_common.constprop.0+0x40/0xe0 do_el0_svc+0x1c/0x28 el0_svc+0x34/0xe0 el0t_64_sync_handler+0x120/0x12c el0t_64_sync+0x190/0x194 Code: aa0403fb f9418400 9100e000 9400102f (f8420f22) ---[ end trace 0000000000000000 ]--- 0000000000006108 : 6108: d503233f paciasp 610c: a9b97bfd stp x29, x30, [sp, #-112]! 6110: 910003fd mov x29, sp 6114: a90153f3 stp x19, x20, [sp, #16] 6118: a9025bf5 stp x21, x22, [sp, #32] 611c: aa0103f6 mov x22, x1 6120: 2a0303f5 mov w21, w3 6124: a90363f7 stp x23, x24, [sp, #48] 6128: aa0003f8 mov x24, x0 612c: aa0203f7 mov x23, x2 6130: a9046bf9 stp x25, x26, [sp, #64] 6134: aa0403f9 mov x25, x4 <-- x4 copied to x25 6138: a90573fb stp x27, x28, [sp, #80] 613c: aa0403fb mov x27, x4 6140: f9418400 ldr x0, [x0, #776] 6144: 9100e000 add x0, x0, #0x38 6148: 94000000 bl 0 614c: f8420f22 ldr x2, [x25, #32]! <-- offset 0x44 ^^^ This is 0x6108 + offset 0x44 from the beginning of sdw_stream_add_slave() where data abort happens. wsa881x_hw_params() is called with stream = NULL and passes it further in register x4 (5th argument) to sdw_stream_add_slave() without any checks. Value from x4 is copied to x25 and finally it aborts on trying to load a value from address in x25 plus offset 32 (in dec) which corresponds to master_list member in struct sdw_stream_runtime: struct sdw_stream_runtime { const char * name; /* 0 8 */ struct sdw_stream_params params; /* 8 12 */ enum sdw_stream_state state; /* 20 4 */ enum sdw_stream_type type; /* 24 4 */ /* XXX 4 bytes hole, try to pack */ here-> struct list_head master_list; /* 32 16 */ int m_rt_count; /* 48 4 */ /* size: 56, cachelines: 1, members: 6 */ /* sum members: 48, holes: 1, sum holes: 4 */ /* padding: 4 */ /* last cacheline: 56 bytes */ Fix this by adding required calls to qcom_snd_sdw_startup() and sdw_release_stream() to startup and shutdown routines which restores the previous correct behaviour when ->set_stream() method is called to set a valid stream runtime pointer on playback startup. Reproduced and then fix was tested on db845c RB3 board. Reported-by: Dmitry Baryshkov Cc: stable@vger.kernel.org Fixes: 15c7fab0e047 ("ASoC: qcom: Move Soundwire runtime stream alloc to soundcards") Cc: Srinivas Kandagatla Cc: Dmitry Baryshkov Cc: Krzysztof Kozlowski Cc: Pierre-Louis Bossart Signed-off-by: Alexey Klimov Tested-by: Steev Klimaszewski # Lenovo Yoga C630 Reviewed-by: Krzysztof Kozlowski Reviewed-by: Srinivas Kandagatla Link: https://patch.msgid.link/20241009213922.999355-1-alexey.klimov@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 75701546b6ea..a479d7e5b7fb 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -15,6 +15,7 @@ #include #include "common.h" #include "qdsp6/q6afe.h" +#include "sdw.h" #include "../codecs/rt5663.h" #define DRIVER_NAME "sdm845" @@ -416,7 +417,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); break; } - return 0; + return qcom_snd_sdw_startup(substream); } static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) @@ -425,6 +426,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) struct snd_soc_card *card = rtd->card; struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; switch (cpu_dai->id) { case PRIMARY_MI2S_RX: @@ -463,6 +465,9 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); break; } + + data->sruntime[cpu_dai->id] = NULL; + sdw_release_stream(sruntime); } static int sdm845_snd_prepare(struct snd_pcm_substream *substream) From 251ce34a446ef0e1d6acd65cf5947abd5d10b8b6 Mon Sep 17 00:00:00 2001 From: Zhu Jun Date: Wed, 9 Oct 2024 00:39:38 -0700 Subject: [PATCH 10/52] ASoC: codecs: Fix error handling in aw_dev_get_dsp_status function Added proper error handling for register value check that return -EPERM when register value does not meet expected condition Signed-off-by: Zhu Jun Link: https://patch.msgid.link/20241009073938.7472-1-zhujun2@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/codecs/aw88399.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/aw88399.c b/sound/soc/codecs/aw88399.c index 8dc2b8aa6832..bba59885242d 100644 --- a/sound/soc/codecs/aw88399.c +++ b/sound/soc/codecs/aw88399.c @@ -656,7 +656,7 @@ static int aw_dev_get_dsp_status(struct aw_device *aw_dev) if (ret) return ret; if (!(reg_val & (~AW88399_WDT_CNT_MASK))) - ret = -EPERM; + return -EPERM; return 0; } From 9eb2142a2ae8c8fdfce2aaa4c110f5a6f6b0b56e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 9 Oct 2024 10:12:30 +0200 Subject: [PATCH 11/52] ASoC: topology: Bump minimal topology ABI version MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When v4 topology support was removed, minimal topology ABI version should have been bumped. Fixes: fe4a07454256 ("ASoC: Drop soc-topology ABI v4 support") Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://patch.msgid.link/20241009081230.304918-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 99333cbd3114..c117672d4439 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -88,7 +88,7 @@ /* ABI version */ #define SND_SOC_TPLG_ABI_VERSION 0x5 /* current version */ -#define SND_SOC_TPLG_ABI_VERSION_MIN 0x4 /* oldest version supported */ +#define SND_SOC_TPLG_ABI_VERSION_MIN 0x5 /* oldest version supported */ /* Max size of TLV data */ #define SND_SOC_TPLG_TLV_SIZE 32 From 182fff3a2aafe4e7f3717a0be9df2fe2ed1a77de Mon Sep 17 00:00:00 2001 From: Christian Heusel Date: Thu, 10 Oct 2024 15:32:11 +0200 Subject: [PATCH 12/52] ASoC: amd: yc: Add quirk for ASUS Vivobook S15 M3502RA As reported the builtin microphone doesn't work on the ASUS Vivobook model S15 OLED M3502RA. Therefore add a quirk for it to make it work. Link: https://bugzilla.kernel.org/show_bug.cgi?id=219345 Signed-off-by: Christian Heusel Link: https://patch.msgid.link/20241010-bugzilla-219345-asus-vivobook-v1-1-3bb24834e2c3@heusel.eu Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index ace6328e91e3..98f9237b7ad7 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -339,6 +339,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M7600RE"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "M3502RA"), + } + }, { .driver_data = &acp6x_card, .matches = { From ca2803fadfd239abf155ef4a563b22a9507ee4b2 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Thu, 10 Oct 2024 19:20:32 +0100 Subject: [PATCH 13/52] ASoC: max98388: Fix missing increment of variable slot_found The variable slot_found is being initialized to zero and inside a for-loop is being checked if it's reached MAX_NUM_CH, however, this is currently impossible since slot_found is never changed. In a previous loop a similar coding pattern is used and slot_found is being incremented. It appears the increment of slot_found is missing from the loop, so fix the code by adding in the increment. Fixes: 6a8e1d46f062 ("ASoC: max98388: add amplifier driver") Signed-off-by: Colin Ian King Link: https://patch.msgid.link/20241010182032.776280-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98388.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98388.c b/sound/soc/codecs/max98388.c index b847d7c59ec0..99986090b4a6 100644 --- a/sound/soc/codecs/max98388.c +++ b/sound/soc/codecs/max98388.c @@ -763,6 +763,7 @@ static int max98388_dai_tdm_slot(struct snd_soc_dai *dai, addr = MAX98388_R2044_PCM_TX_CTRL1 + (cnt / 8); bits = cnt % 8; regmap_update_bits(max98388->regmap, addr, bits, bits); + slot_found++; if (slot_found >= MAX_NUM_CH) break; } From 9b064d200aa8fee9d1d7ced05d8a617e45966715 Mon Sep 17 00:00:00 2001 From: Lad Prabhakar Date: Thu, 10 Oct 2024 15:14:32 +0100 Subject: [PATCH 14/52] ASoC: rsnd: Fix probe failure on HiHope boards due to endpoint parsing On the HiHope boards, we have a single port with a single endpoint defined as below: .... rsnd_port: port { rsnd_endpoint: endpoint { remote-endpoint = <&dw_hdmi0_snd_in>; dai-format = "i2s"; bitclock-master = <&rsnd_endpoint>; frame-master = <&rsnd_endpoint>; playback = <&ssi2>; }; }; .... With commit 547b02f74e4a ("ASoC: rsnd: enable multi Component support for Audio Graph Card/Card2"), support for multiple ports was added. This caused probe failures on HiHope boards, as the endpoint could not be retrieved due to incorrect device node pointers being used. This patch fixes the issue by updating the `rsnd_dai_of_node()` and `rsnd_dai_probe()` functions to use the correct device node pointers based on the port names ('port' or 'ports'). It ensures that the endpoint is properly parsed for both single and multi-port configurations, restoring compatibility with HiHope boards. Fixes: 547b02f74e4a ("ASoC: rsnd: enable multi Component support for Audio Graph Card/Card2") Signed-off-by: Lad Prabhakar Acked-by: Kuninori Morimoto Link: https://patch.msgid.link/20241010141432.716868-1-prabhakar.mahadev-lad.rj@bp.renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 9784718a2b6f..eca5ce096e54 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1281,7 +1281,9 @@ static int rsnd_dai_of_node(struct rsnd_priv *priv, int *is_graph) if (!of_node_name_eq(ports, "ports") && !of_node_name_eq(ports, "port")) continue; - priv->component_dais[i] = of_graph_get_endpoint_count(ports); + priv->component_dais[i] = + of_graph_get_endpoint_count(of_node_name_eq(ports, "ports") ? + ports : np); nr += priv->component_dais[i]; i++; if (i >= RSND_MAX_COMPONENT) { @@ -1493,7 +1495,8 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) if (!of_node_name_eq(ports, "ports") && !of_node_name_eq(ports, "port")) continue; - for_each_endpoint_of_node(ports, dai_np) { + for_each_endpoint_of_node(of_node_name_eq(ports, "ports") ? + ports : np, dai_np) { __rsnd_dai_probe(priv, dai_np, dai_np, 0, dai_i); if (!rsnd_is_gen1(priv) && !rsnd_is_gen2(priv)) { rdai = rsnd_rdai_get(priv, dai_i); From 54c805c1eb264c839fa3027d0073bb7f323b0722 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 11 Oct 2024 12:53:53 +0800 Subject: [PATCH 15/52] ASoC: fsl_esai: change dev_warn to dev_dbg in irq handler Irq handler need to be executed as fast as possible, so the log in irq handler is better to use dev_dbg which needs to be enabled when debugging. Signed-off-by: Shengjiu Wang Reviewed-by: Iuliana Prodan Link: https://patch.msgid.link/1728622433-2873-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index a65f5b9935a2..0b247f16a163 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -119,10 +119,10 @@ static irqreturn_t esai_isr(int irq, void *devid) dev_dbg(&pdev->dev, "isr: Transmission Initialized\n"); if (esr & ESAI_ESR_RFF_MASK) - dev_warn(&pdev->dev, "isr: Receiving overrun\n"); + dev_dbg(&pdev->dev, "isr: Receiving overrun\n"); if (esr & ESAI_ESR_TFE_MASK) - dev_warn(&pdev->dev, "isr: Transmission underrun\n"); + dev_dbg(&pdev->dev, "isr: Transmission underrun\n"); if (esr & ESAI_ESR_TLS_MASK) dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n"); From b930d8647869802a0d430aae6b1b05c3acb24a41 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sat, 12 Oct 2024 12:09:57 +0200 Subject: [PATCH 16/52] ASoC: qcom: Select missing common Soundwire module code on SDM845 SDM845 sound card driver uses qcom_snd_sdw_startup() from the common Soundwire module, so select it to fix build failures: ERROR: modpost: "qcom_snd_sdw_startup" [sound/soc/qcom/snd-soc-sdm845.ko] undefined! Fixes: d0e806b0cc62 ("ASoC: qcom: sdm845: add missing soundwire runtime stream alloc") Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20241012100957.129103-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 762491d6f2f2..3687b9db5ed4 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -157,6 +157,7 @@ config SND_SOC_SDM845 depends on COMMON_CLK select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_SDW select SND_SOC_RT5663 select SND_SOC_MAX98927 imply SND_SOC_CROS_EC_CODEC From 3692a4ccacf3c44249e584aea3ae8568f953e7e4 Mon Sep 17 00:00:00 2001 From: Andrei Simion Date: Mon, 14 Oct 2024 12:28:31 +0300 Subject: [PATCH 17/52] MAINTAINERS: Update maintainer list for MICROCHIP ASOC, SSC and MCP16502 drivers To help Claudiu and offload the work, add myself to the maintainer list for those drivers. Acked-by: Claudiu Beznea Signed-off-by: Andrei Simion Link: https://patch.msgid.link/20241014092830.46709-1-andrei.simion@microchip.com Signed-off-by: Mark Brown --- MAINTAINERS | 3 +++ 1 file changed, 3 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index a097afd76ded..c1a2c296446c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -15089,6 +15089,7 @@ F: drivers/spi/spi-at91-usart.c MICROCHIP AUDIO ASOC DRIVERS M: Claudiu Beznea +M: Andrei Simion L: linux-sound@vger.kernel.org S: Supported F: Documentation/devicetree/bindings/sound/atmel* @@ -15197,6 +15198,7 @@ F: include/video/atmel_lcdc.h MICROCHIP MCP16502 PMIC DRIVER M: Claudiu Beznea +M: Andrei Simion L: linux-arm-kernel@lists.infradead.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/regulator/microchip,mcp16502.yaml @@ -15328,6 +15330,7 @@ F: drivers/spi/spi-atmel.* MICROCHIP SSC DRIVER M: Claudiu Beznea +M: Andrei Simion L: linux-arm-kernel@lists.infradead.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/misc/atmel-ssc.txt From 9822b4c90d77e3c6555fb21c459c4a61c6a8619f Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:07 +0800 Subject: [PATCH 18/52] ASoC: SOF: ipc4-topology: Do not set ALH node_id for aggregated DAIs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For aggregated DAIs, the node ID is set to the group_id during the DAI widget's ipc_prepare op. With the current logic, setting the dai_index for node_id in the dai_config is redundant as it will be overwritten with the group_id anyway. Removing it will also prevent any accidental clearing/resetting of the group_id for aggregated DAIs due to the dai_config calls could that happen before the allocated group_id is freed. Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao All: stable@vger.kernel.org # 6.10.x 6.11.x Link: https://patch.msgid.link/20241016032910.14601-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 87be7f16e8c2..240fee2166d1 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -3129,9 +3129,20 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * * group_id during copier's ipc_prepare op. */ if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { + struct sof_ipc4_alh_configuration_blob *blob; + + blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; ipc4_copier->dai_index = data->dai_node_id; - copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; - copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_node_id); + + /* + * no need to set the node_id for aggregated DAI's. These will be assigned + * a group_id during widget ipc_prepare + */ + if (blob->alh_cfg.device_count == 1) { + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= + SOF_IPC4_NODE_INDEX(data->dai_node_id); + } } break; From 6e38a7e098d32d128b00b42a536151de9ea1340b Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:08 +0800 Subject: [PATCH 19/52] ASoC: SOF: Intel: hda: Handle prepare without close for non-HDA DAI's MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When a PCM is restarted after a snd_pcm_drain/snd_pcm_drop(), the prepare callback will be invoked and the hw_params will be set again. For the HDA DAI's, the hw_params function handles this case already but not for the non-HDA DAI's. So, add the check for link_prepared to verify if the hw_params should be done again or not. Additionally, for SDW DAI's reset the PCMSyCM registers as would be done in the case of a start after a hw_free. Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao All: stable@vger.kernel.org # 6.10.x 6.11.x Link: https://patch.msgid.link/20241016032910.14601-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 36 +++++++++++++++++++++++++++++++---- 1 file changed, 32 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 1c823f9eea57..8cccf38967e7 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -370,6 +370,13 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream, return -EINVAL; } + sdev = widget_to_sdev(w); + hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream); + + /* nothing more to do if the link is already prepared */ + if (hext_stream && hext_stream->link_prepared) + return 0; + /* use HDaudio stream handling */ ret = hda_dai_hw_params_data(substream, params, cpu_dai, data, flags); if (ret < 0) { @@ -377,7 +384,6 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream, return ret; } - sdev = widget_to_sdev(w); if (sdev->dspless_mode_selected) return 0; @@ -482,6 +488,31 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, int ret; int i; + ops = hda_dai_get_ops(substream, cpu_dai); + if (!ops) { + dev_err(cpu_dai->dev, "DAI widget ops not set\n"); + return -EINVAL; + } + + sdev = widget_to_sdev(w); + hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream); + + /* nothing more to do if the link is already prepared */ + if (hext_stream && hext_stream->link_prepared) + return 0; + + /* + * reset the PCMSyCM registers to handle a prepare callback when the PCM is restarted + * due to xruns or after a call to snd_pcm_drain/drop() + */ + ret = hdac_bus_eml_sdw_map_stream_ch(sof_to_bus(sdev), link_id, cpu_dai->id, + 0, 0, substream->stream); + if (ret < 0) { + dev_err(cpu_dai->dev, "%s: hdac_bus_eml_sdw_map_stream_ch failed %d\n", + __func__, ret); + return ret; + } + data.dai_index = (link_id << 8) | cpu_dai->id; data.dai_node_id = intel_alh_id; ret = non_hda_dai_hw_params_data(substream, params, cpu_dai, &data, flags); @@ -490,10 +521,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, return ret; } - ops = hda_dai_get_ops(substream, cpu_dai); - sdev = widget_to_sdev(w); hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream); - if (!hext_stream) return -ENODEV; From c78f1e15e46ac82607eed593b22992fd08644d96 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:09 +0800 Subject: [PATCH 20/52] soundwire: intel_ace2x: Send PDI stream number during prepare MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In the case of a prepare callback after an xrun or when the PCM is restarted after a call to snd_pcm_drain/snd_pcm_drop, avoid reprogramming the SHIM registers but send the PDI stream number so that the link DMA data can be set. This is needed for the case that the DMA data is cleared when the PCM is stopped and restarted without being closed. Link: https://github.com/thesofproject/sof/issues/9502 Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao Acked-by: Vinod Koul All: stable@vger.kernel.org # 6.10.x 6.11.x Link: https://patch.msgid.link/20241016032910.14601-4-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- drivers/soundwire/intel_ace2x.c | 19 ++++++------------- 1 file changed, 6 insertions(+), 13 deletions(-) diff --git a/drivers/soundwire/intel_ace2x.c b/drivers/soundwire/intel_ace2x.c index fff312c6968d..4f3dd70d6a1a 100644 --- a/drivers/soundwire/intel_ace2x.c +++ b/drivers/soundwire/intel_ace2x.c @@ -376,11 +376,12 @@ static int intel_hw_params(struct snd_pcm_substream *substream, static int intel_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_cdns_dai_runtime *dai_runtime; + struct snd_pcm_hw_params *hw_params; int ch, dir; - int ret = 0; dai_runtime = cdns->dai_runtime_array[dai->id]; if (!dai_runtime) { @@ -389,12 +390,8 @@ static int intel_prepare(struct snd_pcm_substream *substream, return -EIO; } + hw_params = &rtd->dpcm[substream->stream].hw_params; if (dai_runtime->suspended) { - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_pcm_hw_params *hw_params; - - hw_params = &rtd->dpcm[substream->stream].hw_params; - dai_runtime->suspended = false; /* @@ -415,15 +412,11 @@ static int intel_prepare(struct snd_pcm_substream *substream, /* the SHIM will be configured in the callback functions */ sdw_cdns_config_stream(cdns, ch, dir, dai_runtime->pdi); - - /* Inform DSP about PDI stream number */ - ret = intel_params_stream(sdw, substream, dai, - hw_params, - sdw->instance, - dai_runtime->pdi->intel_alh_id); } - return ret; + /* Inform DSP about PDI stream number */ + return intel_params_stream(sdw, substream, dai, hw_params, sdw->instance, + dai_runtime->pdi->intel_alh_id); } static int From ab5593793e9088abcddce30ba8e376e31b7285fd Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:10 +0800 Subject: [PATCH 21/52] ASoC: SOF: Intel: hda: Always clean up link DMA during stop MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This is required to reset the DMA read/write pointers when the stream is prepared and restarted after a call to snd_pcm_drain()/snd_pcm_drop(). Also, now that the stream is reset during stop, do not save LLP registers in the case of STOP/suspend to avoid erroneous delay reporting. Link: https://github.com/thesofproject/sof/issues/9502 Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao All: stable@vger.kernel.org # 6.10.x 6.11.x Link: https://patch.msgid.link/20241016032910.14601-5-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai-ops.c | 23 ++++++++++------------- sound/soc/sof/intel/hda-dai.c | 1 + 2 files changed, 11 insertions(+), 13 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 484c76147885..92681ca7f24d 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -346,20 +346,21 @@ static int hda_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: snd_hdac_ext_stream_start(hext_stream); break; - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - snd_hdac_ext_stream_clear(hext_stream); - /* - * Save the LLP registers in case the stream is - * restarting due PAUSE_RELEASE, or START without a pcm - * close/open since in this case the LLP register is not reset - * to 0 and the delay calculation will return with invalid - * results. + * Save the LLP registers since in case of PAUSE the LLP + * register are not reset to 0, the delay calculation will use + * the saved offsets for compensating the delay calculation. */ hext_stream->pplcllpl = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); hext_stream->pplcllpu = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); + snd_hdac_ext_stream_clear(hext_stream); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + hext_stream->pplcllpl = 0; + hext_stream->pplcllpu = 0; + snd_hdac_ext_stream_clear(hext_stream); break; default: dev_err(sdev->dev, "unknown trigger command %d\n", cmd); @@ -512,7 +513,6 @@ static const struct hda_dai_widget_dma_ops sdw_ipc4_chain_dma_ops = { static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, struct snd_pcm_substream *substream, int cmd) { - struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream); struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); switch (cmd) { @@ -527,9 +527,6 @@ static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *c if (ret < 0) return ret; - if (cmd == SNDRV_PCM_TRIGGER_STOP) - return hda_link_dma_cleanup(substream, hext_stream, cpu_dai); - break; } case SNDRV_PCM_TRIGGER_PAUSE_PUSH: diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 8cccf38967e7..ac505c7ad342 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -302,6 +302,7 @@ static int __maybe_unused hda_dai_trigger(struct snd_pcm_substream *substream, i } switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: ret = hda_link_dma_cleanup(substream, hext_stream, dai); if (ret < 0) { From b0867999e3282378a0b26a7ad200233044d31eca Mon Sep 17 00:00:00 2001 From: Ilya Dudikov Date: Wed, 16 Oct 2024 10:40:37 +0700 Subject: [PATCH 22/52] ASoC: amd: yc: Fix non-functional mic on ASUS E1404FA ASUS Vivobook E1404FA needs a quirks-table entry for the internal microphone to function properly. Signed-off-by: Ilya Dudikov Link: https://patch.msgid.link/20241016034038.13481-1-ilyadud25@gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 98f9237b7ad7..438865d5e376 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -325,6 +325,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M6500RC"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "E1404FA"), + } + }, { .driver_data = &acp6x_card, .matches = { From 6924565a04e5f424c95e6d894584e3059f257373 Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Wed, 16 Oct 2024 11:07:03 +0800 Subject: [PATCH 23/52] ASoC: Intel: soc-acpi: lnl: Add match entry for TM2 laptops MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add a new match table entry on Lunarlake for the TM2 laptops with rt713 and rt1318. Signed-off-by: Derek Fang Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Link: https://patch.msgid.link/20241016030703.13669-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-lnl-match.c | 38 +++++++++++++++++++ 1 file changed, 38 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index 3c4e0c7ca8ee..094ed4b27cb0 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -225,6 +225,15 @@ static const struct snd_soc_acpi_adr_device rt1316_3_group1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1318_1_adr[] = { + { + .adr = 0x000133025D131801ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt1318-1" + } +}; + static const struct snd_soc_acpi_adr_device rt1318_1_group1_adr[] = { { .adr = 0x000130025D131801ull, @@ -243,6 +252,15 @@ static const struct snd_soc_acpi_adr_device rt1318_2_group1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt713_0_adr[] = { + { + .adr = 0x000031025D071301ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt713" + } +}; + static const struct snd_soc_acpi_adr_device rt714_0_adr[] = { { .adr = 0x000030025D071401ull, @@ -378,6 +396,20 @@ static const struct snd_soc_acpi_link_adr lnl_sdw_rt1318_l12_rt714_l0[] = { {} }; +static const struct snd_soc_acpi_link_adr lnl_sdw_rt713_l0_rt1318_l1[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt713_0_adr), + .adr_d = rt713_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1318_1_adr), + .adr_d = rt1318_1_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { /* mockup tests need to be first */ @@ -447,6 +479,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-lnl-rt1318-l12-rt714-l0.tplg" }, + { + .link_mask = BIT(0) | BIT(1), + .links = lnl_sdw_rt713_l0_rt1318_l1, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-lnl-rt713-l0-rt1318-l1.tplg" + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_lnl_sdw_machines); From 740883fa6c7262036769aa54b50609c8043977e0 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin Date: Wed, 16 Oct 2024 23:58:10 +0200 Subject: [PATCH 24/52] ASoC: Change my e-mail to gmail Change my contact e-mail in pcm3060 driver and MAINTAINERS Signed-off-by: Kirill Marinushkin Cc: Kirill Marinushkin Cc: Liam Girdwood Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: linux-kernel@vger.kernel.org Cc: linux-sound@vger.kernel.org Link: https://patch.msgid.link/20241016215810.1544222-1-k.marinushkin@gmail.com Signed-off-by: Mark Brown --- MAINTAINERS | 2 +- sound/soc/codecs/pcm3060-i2c.c | 4 ++-- sound/soc/codecs/pcm3060-spi.c | 4 ++-- sound/soc/codecs/pcm3060.c | 4 ++-- sound/soc/codecs/pcm3060.h | 2 +- 5 files changed, 8 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index c1a2c296446c..9d6272c00fbd 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23290,7 +23290,7 @@ F: Documentation/devicetree/bindings/iio/adc/ti,lmp92064.yaml F: drivers/iio/adc/ti-lmp92064.c TI PCM3060 ASoC CODEC DRIVER -M: Kirill Marinushkin +M: Kirill Marinushkin L: linux-sound@vger.kernel.org S: Maintained F: Documentation/devicetree/bindings/sound/pcm3060.txt diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c index 5330cf46b127..3816b25a8ead 100644 --- a/sound/soc/codecs/pcm3060-i2c.c +++ b/sound/soc/codecs/pcm3060-i2c.c @@ -2,7 +2,7 @@ // // PCM3060 I2C driver // -// Copyright (C) 2018 Kirill Marinushkin +// Copyright (C) 2018 Kirill Marinushkin #include #include @@ -55,5 +55,5 @@ static struct i2c_driver pcm3060_i2c_driver = { module_i2c_driver(pcm3060_i2c_driver); MODULE_DESCRIPTION("PCM3060 I2C driver"); -MODULE_AUTHOR("Kirill Marinushkin "); +MODULE_AUTHOR("Kirill Marinushkin "); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c index 3b79734b832b..6095841f2f56 100644 --- a/sound/soc/codecs/pcm3060-spi.c +++ b/sound/soc/codecs/pcm3060-spi.c @@ -2,7 +2,7 @@ // // PCM3060 SPI driver // -// Copyright (C) 2018 Kirill Marinushkin +// Copyright (C) 2018 Kirill Marinushkin #include #include @@ -55,5 +55,5 @@ static struct spi_driver pcm3060_spi_driver = { module_spi_driver(pcm3060_spi_driver); MODULE_DESCRIPTION("PCM3060 SPI driver"); -MODULE_AUTHOR("Kirill Marinushkin "); +MODULE_AUTHOR("Kirill Marinushkin "); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 586ec8c7246c..8974200652e7 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -2,7 +2,7 @@ // // PCM3060 codec driver // -// Copyright (C) 2018 Kirill Marinushkin +// Copyright (C) 2018 Kirill Marinushkin #include #include @@ -343,5 +343,5 @@ int pcm3060_probe(struct device *dev) EXPORT_SYMBOL(pcm3060_probe); MODULE_DESCRIPTION("PCM3060 codec driver"); -MODULE_AUTHOR("Kirill Marinushkin "); +MODULE_AUTHOR("Kirill Marinushkin "); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index 5e1185e7b03d..1b96835600b4 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -2,7 +2,7 @@ /* * PCM3060 codec driver * - * Copyright (C) 2018 Kirill Marinushkin + * Copyright (C) 2018 Kirill Marinushkin */ #ifndef _SND_SOC_PCM3060_H From 9fc9ef05727ccb45fd881770f2aa5c3774b2e8e2 Mon Sep 17 00:00:00 2001 From: Alexey Klimov Date: Wed, 16 Oct 2024 23:10:49 +0100 Subject: [PATCH 25/52] ASoC: codecs: lpass-rx-macro: fix RXn(rx,n) macro for DSM_CTL and SEC7 regs Turns out some registers of pre-2.5 version of rxmacro codecs are not located at the expected offsets but 0xc further away in memory. So far the detected registers are CDC_RX_RX2_RX_PATH_SEC7 and CDC_RX_RX2_RX_PATH_DSM_CTL. CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macro incorrectly generates the address 0x540 for RX2 but it should be 0x54C and it also overwrites CDC_RX_RX2_RX_PATH_SEC7 which is located at 0x540. The same goes for CDC_RX_RXn_RX_PATH_SEC7(rx, n). Fix this by introducing additional rxn_reg_stride2 offset. For 2.5 version and above this offset will be equal to 0. With such change the corresponding RXn() macros will generate the same values for 2.5 codec version for all RX paths and the same old values for pre-2.5 version for RX0 and RX1. However for the latter case with RX2 path it will also add rxn_reg_stride2 on top. While at this, also remove specific if-check for INTERP_AUX from rx_macro_digital_mute() and rx_macro_enable_interp_clk(). These if-check was used to handle such special offset for AUX interpolator but since CDC_RX_RXn_RX_PATH_SEC7(rx, n) and CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macros will generate the correst addresses of dsm register, they are no longer needed. Cc: Srinivas Kandagatla Cc: Krzysztof Kozlowski Signed-off-by: Alexey Klimov Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20241016221049.1145101-1-alexey.klimov@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-rx-macro.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index ef7a70fa6966..febbbe073962 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -202,12 +202,14 @@ #define CDC_RX_RXn_RX_PATH_SEC3(rx, n) (0x042c + rx->rxn_reg_stride * n) #define CDC_RX_RX0_RX_PATH_SEC4 (0x0430) #define CDC_RX_RX0_RX_PATH_SEC7 (0x0434) -#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) (0x0434 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) \ + (0x0434 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0)) #define CDC_RX_DSM_OUT_DELAY_SEL_MASK GENMASK(2, 0) #define CDC_RX_DSM_OUT_DELAY_TWO_SAMPLE 0x2 #define CDC_RX_RX0_RX_PATH_MIX_SEC0 (0x0438) #define CDC_RX_RX0_RX_PATH_MIX_SEC1 (0x043C) -#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) (0x0440 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) \ + (0x0440 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0)) #define CDC_RX_RXn_DSM_CLK_EN_MASK BIT(0) #define CDC_RX_RX0_RX_PATH_DSM_CTL (0x0440) #define CDC_RX_RX0_RX_PATH_DSM_DATA1 (0x0444) @@ -645,6 +647,7 @@ struct rx_macro { int rx_mclk_cnt; enum lpass_codec_version codec_version; int rxn_reg_stride; + int rxn_reg_stride2; bool is_ear_mode_on; bool hph_pwr_mode; bool hph_hd2_mode; @@ -1929,9 +1932,6 @@ static int rx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) CDC_RX_PATH_PGA_MUTE_MASK, 0x0); } - if (j == INTERP_AUX) - dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2); - int_mux_cfg0 = CDC_RX_INP_MUX_RX_INT0_CFG0 + j * 8; int_mux_cfg1 = int_mux_cfg0 + 4; int_mux_cfg0_val = snd_soc_component_read(component, int_mux_cfg0); @@ -2702,9 +2702,6 @@ static int rx_macro_enable_interp_clk(struct snd_soc_component *component, main_reg = CDC_RX_RXn_RX_PATH_CTL(rx, interp_idx); dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, interp_idx); - if (interp_idx == INTERP_AUX) - dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2); - rx_cfg2_reg = CDC_RX_RXn_RX_PATH_CFG2(rx, interp_idx); if (SND_SOC_DAPM_EVENT_ON(event)) { @@ -3821,6 +3818,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_0: case LPASS_CODEC_VERSION_2_1: rx->rxn_reg_stride = 0x80; + rx->rxn_reg_stride2 = 0xc; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_pre_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) @@ -3834,6 +3832,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_7: case LPASS_CODEC_VERSION_2_8: rx->rxn_reg_stride = 0xc0; + rx->rxn_reg_stride2 = 0x0; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) From da95e891dd5d5de6c5ebc010bd028a2e028de093 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Thu, 17 Oct 2024 16:15:07 +0900 Subject: [PATCH 26/52] ASoC: fsl_micfil: Add a flag to distinguish with different volume control types On i.MX8MM the register of volume control has positive and negative values. It is different from other platforms like i.MX8MP and i.MX93 which only have positive values. Add a volume_sx flag to use SX_TLV volume control for this kind of platform. Use common TLV volume control for other platforms. Fixes: cdfa92eb90f5 ("ASoC: fsl_micfil: Correct the number of steps on SX controls") Signed-off-by: Chancel Liu Reviewed-by: Daniel Baluta Link: https://patch.msgid.link/20241017071507.2577786-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 43 +++++++++++++++++++++++++++++++++++++- 1 file changed, 42 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 193be098fa5e..84638c1a2863 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -67,6 +67,7 @@ struct fsl_micfil_soc_data { bool imx; bool use_edma; bool use_verid; + bool volume_sx; u64 formats; }; @@ -76,6 +77,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mm = { .fifo_depth = 8, .dataline = 0xf, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .volume_sx = true, }; static struct fsl_micfil_soc_data fsl_micfil_imx8mp = { @@ -84,6 +86,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mp = { .fifo_depth = 32, .dataline = 0xf, .formats = SNDRV_PCM_FMTBIT_S32_LE, + .volume_sx = false, }; static struct fsl_micfil_soc_data fsl_micfil_imx93 = { @@ -94,6 +97,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx93 = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .use_edma = true, .use_verid = true, + .volume_sx = false, }; static const struct of_device_id fsl_micfil_dt_ids[] = { @@ -317,7 +321,26 @@ static int hwvad_detected(struct snd_kcontrol *kcontrol, return 0; } -static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { +static const struct snd_kcontrol_new fsl_micfil_volume_controls[] = { + SOC_SINGLE_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(0), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(1), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH2 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(2), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH3 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(3), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH4 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(4), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH5 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(5), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH6 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(6), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(7), 0xF, 0, gain_tlv), +}; + +static const struct snd_kcontrol_new fsl_micfil_volume_sx_controls[] = { SOC_SINGLE_SX_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, MICFIL_OUTGAIN_CHX_SHIFT(0), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, @@ -334,6 +357,9 @@ static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { MICFIL_OUTGAIN_CHX_SHIFT(6), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, MICFIL_OUTGAIN_CHX_SHIFT(7), 0x8, 0xF, gain_tlv), +}; + +static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { SOC_ENUM_EXT("MICFIL Quality Select", fsl_micfil_quality_enum, micfil_quality_get, micfil_quality_set), @@ -801,6 +827,20 @@ static int fsl_micfil_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } +static int fsl_micfil_component_probe(struct snd_soc_component *component) +{ + struct fsl_micfil *micfil = snd_soc_component_get_drvdata(component); + + if (micfil->soc->volume_sx) + snd_soc_add_component_controls(component, fsl_micfil_volume_sx_controls, + ARRAY_SIZE(fsl_micfil_volume_sx_controls)); + else + snd_soc_add_component_controls(component, fsl_micfil_volume_controls, + ARRAY_SIZE(fsl_micfil_volume_controls)); + + return 0; +} + static const struct snd_soc_dai_ops fsl_micfil_dai_ops = { .probe = fsl_micfil_dai_probe, .startup = fsl_micfil_startup, @@ -821,6 +861,7 @@ static struct snd_soc_dai_driver fsl_micfil_dai = { static const struct snd_soc_component_driver fsl_micfil_component = { .name = "fsl-micfil-dai", + .probe = fsl_micfil_component_probe, .controls = fsl_micfil_snd_controls, .num_controls = ARRAY_SIZE(fsl_micfil_snd_controls), .legacy_dai_naming = 1, From 72cafe63b35d06b5cfbaf807e90ae657907858da Mon Sep 17 00:00:00 2001 From: Andrey Shumilin Date: Fri, 18 Oct 2024 09:00:18 +0300 Subject: [PATCH 27/52] ALSA: firewire-lib: Avoid division by zero in apply_constraint_to_size() The step variable is initialized to zero. It is changed in the loop, but if it's not changed it will remain zero. Add a variable check before the division. The observed behavior was introduced by commit 826b5de90c0b ("ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size"), and it is difficult to show that any of the interval parameters will satisfy the snd_interval_test() condition with data from the amdtp_rate_table[] table. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 826b5de90c0b ("ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size") Signed-off-by: Andrey Shumilin Reviewed-by: Takashi Sakamoto Link: https://patch.msgid.link/20241018060018.1189537-1-shum.sdl@nppct.ru Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index c72b2a754775..7fc51f829ecc 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -172,6 +172,9 @@ static int apply_constraint_to_size(struct snd_pcm_hw_params *params, step = max(step, amdtp_syt_intervals[i]); } + if (step == 0) + return -EINVAL; + t.min = roundup(s->min, step); t.max = rounddown(s->max, step); t.integer = 1; From 35fdc6e1c16099078bcbd73a6c8f1733ae7f1909 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jos=C3=A9=20Relvas?= Date: Sun, 20 Oct 2024 11:27:56 +0100 Subject: [PATCH 28/52] ALSA: hda/realtek: Add subwoofer quirk for Acer Predator G9-593 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The Acer Predator G9-593 has a 2+1 speaker system which isn't probed correctly. This patch adds a quirk with the proper pin connections. Note that I do not own this laptop, so I cannot guarantee that this fixes the issue. Testing was done by other users here: https://discussion.fedoraproject.org/t/-/118482 This model appears to have two different dev IDs... - 0x1177 (as seen on the forum link above) - 0x1178 (as seen on https://linux-hardware.org/?probe=127df9999f) I don't think the audio system was changed between model revisions, so the patch applies for both IDs. Signed-off-by: José Relvas Link: https://patch.msgid.link/20241020102756.225258-1-josemonsantorelvas@gmail.com Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3bbf5fab2881..edf688f989c8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7649,6 +7649,7 @@ enum { ALC286_FIXUP_ACER_AIO_HEADSET_MIC, ALC256_FIXUP_ASUS_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, + ALC255_FIXUP_PREDATOR_SUBWOOFER, ALC299_FIXUP_PREDATOR_SPK, ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, ALC289_FIXUP_DELL_SPK1, @@ -9063,6 +9064,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE }, + [ALC255_FIXUP_PREDATOR_SUBWOOFER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x17, 0x90170151 }, /* use as internal speaker (LFE) */ + { 0x1b, 0x90170152 } /* use as internal speaker (back) */ + } + }, [ALC299_FIXUP_PREDATOR_SPK] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10150,6 +10158,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1166, "Acer Veriton N4640G", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1025, 0x1167, "Acer Veriton N6640G", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x1025, 0x1177, "Acer Predator G9-593", ALC255_FIXUP_PREDATOR_SUBWOOFER), + SND_PCI_QUIRK(0x1025, 0x1178, "Acer Predator G9-593", ALC255_FIXUP_PREDATOR_SUBWOOFER), SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK), SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE), From 86c96e7289c5758284b562ac7b5c94429f48d2d9 Mon Sep 17 00:00:00 2001 From: Eric Biggers Date: Sun, 20 Oct 2024 10:56:24 -0700 Subject: [PATCH 29/52] ALSA: hda/tas2781: select CRC32 instead of CRC32_SARWATE Fix the kconfig option for the tas2781 HDA driver to select CRC32 rather than CRC32_SARWATE. CRC32_SARWATE is an option from the kconfig 'choice' that selects the specific CRC32 implementation. Selecting a 'choice' option seems to have no effect, but even if it did work, it would be incorrect for a random driver to override the user's choice. CRC32 is the correct option to select for crc32() to be available. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Cc: stable@vger.kernel.org Signed-off-by: Eric Biggers Link: https://patch.msgid.link/20241020175624.7095-1-ebiggers@kernel.org Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bb15a0248250..68f1eee9e5c9 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -198,7 +198,7 @@ config SND_HDA_SCODEC_TAS2781_I2C depends on SND_SOC select SND_SOC_TAS2781_COMLIB select SND_SOC_TAS2781_FMWLIB - select CRC32_SARWATE + select CRC32 help Say Y or M here to include TAS2781 I2C HD-audio side codec support in snd-hda-intel driver, such as ALC287. From 038fa6ddf5d22694f61ff7a7a53c8887c6b08c45 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Mon, 21 Oct 2024 06:15:44 +0000 Subject: [PATCH 30/52] ASoC: rt722-sdca: increase clk_stop_timeout to fix clock stop issue clk_stop_timeout should be increased to 900ms to fix clock stop issue. Signed-off-by: Jack Yu Link: https://patch.msgid.link/cd26275d9fc54374a18dc016755cb72d@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index 87354bb1564e..d5c985ff5ac5 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -253,7 +253,7 @@ static int rt722_sdca_read_prop(struct sdw_slave *slave) } /* set the timeout values */ - prop->clk_stop_timeout = 200; + prop->clk_stop_timeout = 900; /* wake-up event */ prop->wake_capable = 1; From b9a8ecf81066e01e8a3de35517481bc5aa0439e5 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 14 Oct 2024 13:38:33 +0800 Subject: [PATCH 31/52] ASoC: fsl_micfil: Add sample rate constraint On some platforms, for example i.MX93, there is only one audio PLL source, so some sample rate can't be supported. If the PLL source is used for 8kHz series rates, then 11kHz series rates can't be supported. So add constraints according to the frequency of available clock sources, then alsa-lib will help to convert the unsupported rate for the driver. Signed-off-by: Shengjiu Wang Link: https://patch.msgid.link/1728884313-6778-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 38 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 84638c1a2863..0c71a73476df 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -28,6 +28,13 @@ #define MICFIL_OSR_DEFAULT 16 +#define MICFIL_NUM_RATES 7 +#define MICFIL_CLK_SRC_NUM 3 +/* clock source ids */ +#define MICFIL_AUDIO_PLL1 0 +#define MICFIL_AUDIO_PLL2 1 +#define MICFIL_CLK_EXT3 2 + enum quality { QUALITY_HIGH, QUALITY_MEDIUM, @@ -45,9 +52,12 @@ struct fsl_micfil { struct clk *mclk; struct clk *pll8k_clk; struct clk *pll11k_clk; + struct clk *clk_src[MICFIL_CLK_SRC_NUM]; struct snd_dmaengine_dai_dma_data dma_params_rx; struct sdma_peripheral_config sdmacfg; struct snd_soc_card *card; + struct snd_pcm_hw_constraint_list constraint_rates; + unsigned int constraint_rates_list[MICFIL_NUM_RATES]; unsigned int dataline; char name[32]; int irq[MICFIL_IRQ_LINES]; @@ -475,12 +485,34 @@ static int fsl_micfil_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_micfil *micfil = snd_soc_dai_get_drvdata(dai); + unsigned int rates[MICFIL_NUM_RATES] = {8000, 11025, 16000, 22050, 32000, 44100, 48000}; + int i, j, k = 0; + u64 clk_rate; if (!micfil) { dev_err(dai->dev, "micfil dai priv_data not set\n"); return -EINVAL; } + micfil->constraint_rates.list = micfil->constraint_rates_list; + micfil->constraint_rates.count = 0; + + for (j = 0; j < MICFIL_NUM_RATES; j++) { + for (i = 0; i < MICFIL_CLK_SRC_NUM; i++) { + clk_rate = clk_get_rate(micfil->clk_src[i]); + if (clk_rate != 0 && do_div(clk_rate, rates[j]) == 0) { + micfil->constraint_rates_list[k++] = rates[j]; + micfil->constraint_rates.count++; + break; + } + } + } + + if (micfil->constraint_rates.count > 0) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &micfil->constraint_rates); + return 0; } @@ -1175,6 +1207,12 @@ static int fsl_micfil_probe(struct platform_device *pdev) fsl_asoc_get_pll_clocks(&pdev->dev, &micfil->pll8k_clk, &micfil->pll11k_clk); + micfil->clk_src[MICFIL_AUDIO_PLL1] = micfil->pll8k_clk; + micfil->clk_src[MICFIL_AUDIO_PLL2] = micfil->pll11k_clk; + micfil->clk_src[MICFIL_CLK_EXT3] = devm_clk_get(&pdev->dev, "clkext3"); + if (IS_ERR(micfil->clk_src[MICFIL_CLK_EXT3])) + micfil->clk_src[MICFIL_CLK_EXT3] = NULL; + /* init regmap */ regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(regs)) From db7e59e6a39a4d3d54ca8197c796557e6d480b0d Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sat, 12 Oct 2024 12:11:08 +0200 Subject: [PATCH 32/52] ASoC: qcom: sc7280: Fix missing Soundwire runtime stream alloc Commit 15c7fab0e047 ("ASoC: qcom: Move Soundwire runtime stream alloc to soundcards") moved the allocation of Soundwire stream runtime from the Qualcomm Soundwire driver to each individual machine sound card driver, except that it forgot to update SC7280 card. Just like for other Qualcomm sound cards using Soundwire, the card driver should allocate and release the runtime. Otherwise sound playback will result in a NULL pointer dereference or other effect of uninitialized memory accesses (which was confirmed on SDM845 having similar issue). Cc: stable@vger.kernel.org Cc: Alexey Klimov Cc: Steev Klimaszewski Fixes: 15c7fab0e047 ("ASoC: qcom: Move Soundwire runtime stream alloc to soundcards") Link: https://lore.kernel.org/r/20241010054109.16938-1-krzysztof.kozlowski@linaro.org Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20241012101108.129476-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/sc7280.c | 10 +++++++++- 2 files changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 3687b9db5ed4..ca7a30ebd26a 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -209,6 +209,7 @@ config SND_SOC_SC7280 tristate "SoC Machine driver for SC7280 boards" depends on I2C && SOUNDWIRE select SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_SDW select SND_SOC_LPASS_SC7280 select SND_SOC_MAX98357A select SND_SOC_WCD938X_SDW diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c index 207ac5da4dd4..230af8d7b205 100644 --- a/sound/soc/qcom/sc7280.c +++ b/sound/soc/qcom/sc7280.c @@ -23,6 +23,7 @@ #include "common.h" #include "lpass.h" #include "qdsp6/q6afe.h" +#include "sdw.h" #define DEFAULT_MCLK_RATE 19200000 #define RT5682_PLL_FREQ (48000 * 512) @@ -316,6 +317,7 @@ static void sc7280_snd_shutdown(struct snd_pcm_substream *substream) struct snd_soc_card *card = rtd->card; struct sc7280_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; switch (cpu_dai->id) { case MI2S_PRIMARY: @@ -333,6 +335,9 @@ static void sc7280_snd_shutdown(struct snd_pcm_substream *substream) default: break; } + + data->sruntime[cpu_dai->id] = NULL; + sdw_release_stream(sruntime); } static int sc7280_snd_startup(struct snd_pcm_substream *substream) @@ -347,6 +352,8 @@ static int sc7280_snd_startup(struct snd_pcm_substream *substream) switch (cpu_dai->id) { case MI2S_PRIMARY: ret = sc7280_rt5682_init(rtd); + if (ret) + return ret; break; case SECONDARY_MI2S_RX: codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; @@ -360,7 +367,8 @@ static int sc7280_snd_startup(struct snd_pcm_substream *substream) default: break; } - return ret; + + return qcom_snd_sdw_startup(substream); } static const struct snd_soc_ops sc7280_ops = { From e3ea2757c312e51bbf62ebc434a6f7df1e3a201f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 23 Oct 2024 16:13:10 +0800 Subject: [PATCH 33/52] ALSA: hda/realtek: Update default depop procedure Old procedure has a chance to meet Headphone no output. Fixes: c2d6af53a43f ("ALSA: hda/realtek - Add default procedure for suspend and resume state") Signed-off-by: Kailang Yang Link: https://lore.kernel.org/17b717a0a0b04a77aea4a8ec820cba13@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 38 ++++++++++++++++------------------- 1 file changed, 17 insertions(+), 21 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index edf688f989c8..3567b14b52b7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3868,20 +3868,18 @@ static void alc_default_init(struct hda_codec *codec) hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); - if (hp_pin_sense) + if (hp_pin_sense) { msleep(2); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - if (hp_pin_sense) - msleep(85); + msleep(75); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - - if (hp_pin_sense) - msleep(100); + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + msleep(75); + } } static void alc_default_shutup(struct hda_codec *codec) @@ -3897,22 +3895,20 @@ static void alc_default_shutup(struct hda_codec *codec) hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); - if (hp_pin_sense) + if (hp_pin_sense) { msleep(2); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - - if (hp_pin_sense) - msleep(85); - - if (!spec->no_shutup_pins) snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - if (hp_pin_sense) - msleep(100); + msleep(75); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + msleep(75); + } alc_auto_setup_eapd(codec, false); alc_shutup_pins(codec); } From 032532f91a1d06d0750f16c49a9698ef5374a68f Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 24 Oct 2024 23:56:12 +0200 Subject: [PATCH 34/52] ASoC: codecs: rt5640: Always disable IRQs from rt5640_cancel_work() Disable IRQs from rt5640_cancel_work(), this fixes a crash caused by the IRQ never getting freed when the driver is unbound from the i2c_client with jack-detection active: [ 193.138780] rt5640 i2c-rt5640: ASoC: unknown pin LDO2 [ 193.138830] rt5640 i2c-rt5640: ASoC: unknown pin MICBIAS1 [ 193.671218] BUG: kernel NULL pointer dereference, address: 0000000000000078 [ 193.671239] #PF: supervisor read access in kernel mode [ 193.671248] #PF: error_code(0x0000) - not-present page ... [ 193.671531] ? asm_exc_page_fault+0x22/0x30 [ 193.671551] ? rt5640_jack_inserted+0x10/0x80 [snd_soc_rt5640] [ 193.671574] rt5640_detect_headset+0x93/0x130 [snd_soc_rt5640] [ 193.671596] rt5640_jack_work+0x93/0x355 [snd_soc_rt5640] Signed-off-by: Hans de Goede Link: https://patch.msgid.link/20241024215612.92147-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 16f3425a3e35..855139348edb 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2419,10 +2419,20 @@ static irqreturn_t rt5640_jd_gpio_irq(int irq, void *data) return IRQ_HANDLED; } -static void rt5640_cancel_work(void *data) +static void rt5640_disable_irq_and_cancel_work(void *data) { struct rt5640_priv *rt5640 = data; + if (rt5640->jd_gpio_irq_requested) { + free_irq(rt5640->jd_gpio_irq, rt5640); + rt5640->jd_gpio_irq_requested = false; + } + + if (rt5640->irq_requested) { + free_irq(rt5640->irq, rt5640); + rt5640->irq_requested = false; + } + cancel_delayed_work_sync(&rt5640->jack_work); cancel_delayed_work_sync(&rt5640->bp_work); } @@ -2463,13 +2473,7 @@ static void rt5640_disable_jack_detect(struct snd_soc_component *component) if (!rt5640->jack) return; - if (rt5640->jd_gpio_irq_requested) - free_irq(rt5640->jd_gpio_irq, rt5640); - - if (rt5640->irq_requested) - free_irq(rt5640->irq, rt5640); - - rt5640_cancel_work(rt5640); + rt5640_disable_irq_and_cancel_work(rt5640); if (rt5640->jack->status & SND_JACK_MICROPHONE) { rt5640_disable_micbias1_ovcd_irq(component); @@ -2477,8 +2481,6 @@ static void rt5640_disable_jack_detect(struct snd_soc_component *component) snd_soc_jack_report(rt5640->jack, 0, SND_JACK_BTN_0); } - rt5640->jd_gpio_irq_requested = false; - rt5640->irq_requested = false; rt5640->jd_gpio = NULL; rt5640->jack = NULL; } @@ -2798,7 +2800,8 @@ static int rt5640_suspend(struct snd_soc_component *component) if (rt5640->jack) { /* disable jack interrupts during system suspend */ disable_irq(rt5640->irq); - rt5640_cancel_work(rt5640); + cancel_delayed_work_sync(&rt5640->jack_work); + cancel_delayed_work_sync(&rt5640->bp_work); } snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); @@ -3032,7 +3035,7 @@ static int rt5640_i2c_probe(struct i2c_client *i2c) INIT_DELAYED_WORK(&rt5640->jack_work, rt5640_jack_work); /* Make sure work is stopped on probe-error / remove */ - ret = devm_add_action_or_reset(&i2c->dev, rt5640_cancel_work, rt5640); + ret = devm_add_action_or_reset(&i2c->dev, rt5640_disable_irq_and_cancel_work, rt5640); if (ret) return ret; From d48696b915527b5bcdd207a299aec03fb037eb17 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 24 Oct 2024 23:16:14 +0200 Subject: [PATCH 35/52] ASoC: Intel: bytcr_rt5640: Add support for non ACPI instantiated codec MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit On some x86 Bay Trail tablets which shipped with Android as factory OS, the DSDT is so broken that the codec needs to be manually instantatiated by the special x86-android-tablets.ko "fixup" driver for cases like this. This means that the codec-dev cannot be retrieved through its ACPI fwnode, add support to the bytcr_rt5640 machine driver for such manually instantiated rt5640 i2c_clients. An example of a tablet which needs this is the Vexia EDU ATLA 10 tablet, which has been distributed to schools in the Spanish Andalucía region. Signed-off-by: Hans de Goede Link: https://patch.msgid.link/20241024211615.79518-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 33 ++++++++++++++++++++++++--- 1 file changed, 30 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 2ed49acb4e36..8dfd91cc3668 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -32,6 +33,8 @@ #include "../atom/sst-atom-controls.h" #include "../common/soc-intel-quirks.h" +#define BYT_RT5640_FALLBACK_CODEC_DEV_NAME "i2c-rt5640" + enum { BYT_RT5640_DMIC1_MAP, BYT_RT5640_DMIC2_MAP, @@ -1698,9 +1701,33 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) codec_dev = acpi_get_first_physical_node(adev); acpi_dev_put(adev); - if (!codec_dev) - return -EPROBE_DEFER; - priv->codec_dev = get_device(codec_dev); + + if (codec_dev) { + priv->codec_dev = get_device(codec_dev); + } else { + /* + * Special case for Android tablets where the codec i2c_client + * has been manually instantiated by x86_android_tablets.ko due + * to a broken DSDT. + */ + codec_dev = bus_find_device_by_name(&i2c_bus_type, NULL, + BYT_RT5640_FALLBACK_CODEC_DEV_NAME); + if (!codec_dev) + return -EPROBE_DEFER; + + if (!i2c_verify_client(codec_dev)) { + dev_err(dev, "Error '%s' is not an i2c_client\n", + BYT_RT5640_FALLBACK_CODEC_DEV_NAME); + put_device(codec_dev); + } + + /* fixup codec name */ + strscpy(byt_rt5640_codec_name, BYT_RT5640_FALLBACK_CODEC_DEV_NAME, + sizeof(byt_rt5640_codec_name)); + + /* bus_find_device() returns a reference no need to get() */ + priv->codec_dev = codec_dev; + } /* * swap SSP0 if bytcr is detected From 0107f28f135231da22a9ad5756bb16bd5cada4d5 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 24 Oct 2024 23:16:15 +0200 Subject: [PATCH 36/52] ASoC: Intel: bytcr_rt5640: Add DMI quirk for Vexia Edu Atla 10 tablet The Vexia Edu Atla 10 tablet mostly uses the BYTCR tablet defaults, but as happens on more models it is using IN1 instead of IN3 for its internal mic and JD_SRC_JD2_IN4N instead of JD_SRC_JD1_IN4P for jack-detection. Add a DMI quirk for this to fix the internal-mic and jack-detection. Signed-off-by: Hans de Goede Link: https://patch.msgid.link/20241024211615.79518-2-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 8dfd91cc3668..54f77f57ec8e 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1132,6 +1132,21 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { /* Vexia Edu Atla 10 tablet */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* Above strings are too generic, also match on BIOS date */ + DMI_MATCH(DMI_BIOS_DATE, "08/25/2014"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { /* Voyo Winpad A15 */ .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), From 78e7be018784934081afec77f96d49a2483f9188 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 18 Oct 2024 13:53:24 +0800 Subject: [PATCH 37/52] ALSA: hda/realtek: Limit internal Mic boost on Dell platform Dell want to limit internal Mic boost on all Dell platform. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/561fc5f5eff04b6cbd79ed173cd1c1db@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3567b14b52b7..784ac058418f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7521,6 +7521,7 @@ enum { ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL1_LIMIT_INT_MIC_BOOST, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, @@ -7555,6 +7556,7 @@ enum { ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC255_FIXUP_DELL1_LIMIT_INT_MIC_BOOST, ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, ALC255_FIXUP_HEADSET_MODE, ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, @@ -8114,6 +8116,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC269_FIXUP_DELL1_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC269_FIXUP_DELL2_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -8394,6 +8402,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_DELL1_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC255_FIXUP_DELL2_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -11076,6 +11090,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, {.id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, .name = "dell-headset3"}, {.id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, .name = "dell-headset4"}, + {.id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_QUIET, .name = "dell-headset4-quiet"}, {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, @@ -11630,16 +11645,16 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1b, 0x40000000}), - SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_QUIET, {0x19, 0x40000000}, {0x1b, 0x40000000}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1a, 0x40000000}), - SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_LIMIT_INT_MIC_BOOST, {0x19, 0x40000000}, {0x1a, 0x40000000}), - SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, + SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC269_FIXUP_DELL1_LIMIT_INT_MIC_BOOST, {0x19, 0x40000000}, {0x1a, 0x40000000}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC2XX_FIXUP_HEADSET_MIC, From 6668610b4d8ce9a3ee3ed61a9471f62fb5f05bf9 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Fri, 25 Oct 2024 11:02:21 +0200 Subject: [PATCH 38/52] ASoC: Intel: sst: Support LPE0F28 ACPI HID MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Some old Bay Trail tablets which shipped with Android as factory OS have the SST/LPE audio engine described by an ACPI device with a HID (Hardware-ID) of LPE0F28 instead of 80860F28. Add support for this. Note this uses a new sst_res_info for just the LPE0F28 case because it has a different layout for the IO-mem ACPI resources then the 80860F28. An example of a tablet which needs this is the Vexia EDU ATLA 10 tablet, which has been distributed to schools in the Spanish Andalucía region. Signed-off-by: Hans de Goede Link: https://patch.msgid.link/20241025090221.52198-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/hda/intel-dsp-config.c | 4 ++ sound/soc/intel/atom/sst/sst_acpi.c | 64 +++++++++++++++++++++++++---- 2 files changed, 59 insertions(+), 9 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index f018bd779862..9f849e05ce79 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -721,6 +721,10 @@ static const struct config_entry acpi_config_table[] = { #if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) || \ IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) /* BayTrail */ + { + .flags = FLAG_SST_OR_SOF_BYT, + .acpi_hid = "LPE0F28", + }, { .flags = FLAG_SST_OR_SOF_BYT, .acpi_hid = "80860F28", diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 9956dc63db74..f4c4774249ee 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -125,6 +125,28 @@ static const struct sst_res_info bytcr_res_info = { .acpi_ipc_irq_index = 0 }; +/* For "LPE0F28" ACPI device found on some Android factory OS models */ +static const struct sst_res_info lpe8086_res_info = { + .shim_offset = 0x140000, + .shim_size = 0x000100, + .shim_phy_addr = SST_BYT_SHIM_PHY_ADDR, + .ssp0_offset = 0xa0000, + .ssp0_size = 0x1000, + .dma0_offset = 0x98000, + .dma0_size = 0x4000, + .dma1_offset = 0x9c000, + .dma1_size = 0x4000, + .iram_offset = 0x0c0000, + .iram_size = 0x14000, + .dram_offset = 0x100000, + .dram_size = 0x28000, + .mbox_offset = 0x144000, + .mbox_size = 0x1000, + .acpi_lpe_res_index = 1, + .acpi_ddr_index = 0, + .acpi_ipc_irq_index = 0 +}; + static struct sst_platform_info byt_rvp_platform_data = { .probe_data = &byt_fwparse_info, .ipc_info = &byt_ipc_info, @@ -268,10 +290,38 @@ static int sst_acpi_probe(struct platform_device *pdev) mach->pdata = &chv_platform_data; pdata = mach->pdata; - ret = kstrtouint(id->id, 16, &dev_id); - if (ret < 0) { - dev_err(dev, "Unique device id conversion error: %d\n", ret); - return ret; + if (!strcmp(id->id, "LPE0F28")) { + struct resource *rsrc; + + /* Use regular BYT SST PCI VID:PID */ + dev_id = 0x80860F28; + byt_rvp_platform_data.res_info = &lpe8086_res_info; + + /* + * The "LPE0F28" ACPI device has separate IO-mem resources for: + * DDR, SHIM, MBOX, IRAM, DRAM, CFG + * None of which covers the entire LPE base address range. + * lpe8086_res_info.acpi_lpe_res_index points to the SHIM. + * Patch this to cover the entire base address range as expected + * by sst_platform_get_resources(). + */ + rsrc = platform_get_resource(pdev, IORESOURCE_MEM, + pdata->res_info->acpi_lpe_res_index); + if (!rsrc) { + dev_err(ctx->dev, "Invalid SHIM base\n"); + return -EIO; + } + rsrc->start -= pdata->res_info->shim_offset; + rsrc->end = rsrc->start + 0x200000 - 1; + } else { + ret = kstrtouint(id->id, 16, &dev_id); + if (ret < 0) { + dev_err(dev, "Unique device id conversion error: %d\n", ret); + return ret; + } + + if (soc_intel_is_byt_cr(pdev)) + byt_rvp_platform_data.res_info = &bytcr_res_info; } dev_dbg(dev, "ACPI device id: %x\n", dev_id); @@ -280,11 +330,6 @@ static int sst_acpi_probe(struct platform_device *pdev) if (ret < 0) return ret; - if (soc_intel_is_byt_cr(pdev)) { - /* override resource info */ - byt_rvp_platform_data.res_info = &bytcr_res_info; - } - /* update machine parameters */ mach->mach_params.acpi_ipc_irq_index = pdata->res_info->acpi_ipc_irq_index; @@ -344,6 +389,7 @@ static void sst_acpi_remove(struct platform_device *pdev) } static const struct acpi_device_id sst_acpi_ids[] = { + { "LPE0F28", (unsigned long)&snd_soc_acpi_intel_baytrail_machines}, { "80860F28", (unsigned long)&snd_soc_acpi_intel_baytrail_machines}, { "808622A8", (unsigned long)&snd_soc_acpi_intel_cherrytrail_machines}, { }, From d221b844ee79823ffc29b7badc4010bdb0960224 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Sat, 26 Oct 2024 22:46:34 +0200 Subject: [PATCH 39/52] ASoC: cs42l51: Fix some error handling paths in cs42l51_probe() If devm_gpiod_get_optional() fails, we need to disable previously enabled regulators, as done in the other error handling path of the function. Also, gpiod_set_value_cansleep(, 1) needs to be called to undo a potential gpiod_set_value_cansleep(, 0). If the "reset" gpio is not defined, this additional call is just a no-op. This behavior is the same as the one already in the .remove() function. Fixes: 11b9cd748e31 ("ASoC: cs42l51: add reset management") Signed-off-by: Christophe JAILLET Reviewed-by: Charles Keepax Link: https://patch.msgid.link/a5e5f4b9fb03f46abd2c93ed94b5c395972ce0d1.1729975570.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index e4827b8c2bde..6e51954bdb1e 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -747,8 +747,10 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) cs42l51->reset_gpio = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); - if (IS_ERR(cs42l51->reset_gpio)) - return PTR_ERR(cs42l51->reset_gpio); + if (IS_ERR(cs42l51->reset_gpio)) { + ret = PTR_ERR(cs42l51->reset_gpio); + goto error; + } if (cs42l51->reset_gpio) { dev_dbg(dev, "Release reset gpio\n"); @@ -780,6 +782,7 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) return 0; error: + gpiod_set_value_cansleep(cs42l51->reset_gpio, 1); regulator_bulk_disable(ARRAY_SIZE(cs42l51->supplies), cs42l51->supplies); return ret; From c1895ba181e560144601fafe46aeedbafdf4dbc4 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sat, 26 Oct 2024 16:36:15 +0200 Subject: [PATCH 40/52] ASoC: Intel: sst: Fix used of uninitialized ctx to log an error Fix the new "LPE0F28" code path using the uninitialized ctx variable to log an error. Fixes: 6668610b4d8c ("ASoC: Intel: sst: Support LPE0F28 ACPI HID") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202410261106.EBx49ssy-lkp@intel.com/ Signed-off-by: Hans de Goede Link: https://patch.msgid.link/20241026143615.171821-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index f4c4774249ee..257180630475 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -308,7 +308,7 @@ static int sst_acpi_probe(struct platform_device *pdev) rsrc = platform_get_resource(pdev, IORESOURCE_MEM, pdata->res_info->acpi_lpe_res_index); if (!rsrc) { - dev_err(ctx->dev, "Invalid SHIM base\n"); + dev_err(dev, "Invalid SHIM base\n"); return -EIO; } rsrc->start -= pdata->res_info->shim_offset; From 2ef9439f7a19fd3d43b288d38b1c6e55b668a4fe Mon Sep 17 00:00:00 2001 From: Aleksei Vetrov Date: Mon, 28 Oct 2024 22:50:30 +0000 Subject: [PATCH 41/52] ASoC: dapm: fix bounds checker error in dapm_widget_list_create The widgets array in the snd_soc_dapm_widget_list has a __counted_by attribute attached to it, which points to the num_widgets variable. This attribute is used in bounds checking, and if it is not set before the array is filled, then the bounds sanitizer will issue a warning or a kernel panic if CONFIG_UBSAN_TRAP is set. This patch sets the size of the widgets list calculated with list_for_each as the initial value for num_widgets as it is used for allocating memory for the array. It is updated with the actual number of added elements after the array is filled. Signed-off-by: Aleksei Vetrov Fixes: 80e698e2df5b ("ASoC: soc-dapm: Annotate struct snd_soc_dapm_widget_list with __counted_by") Link: https://patch.msgid.link/20241028-soc-dapm-bounds-checker-fix-v1-1-262b0394e89e@google.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c34934c31ffe..99521c784a9b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1147,6 +1147,8 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, if (*list == NULL) return -ENOMEM; + (*list)->num_widgets = size; + list_for_each_entry(w, widgets, work_list) (*list)->widgets[i++] = w; From ef5fbdf732a158ec27eeba69d8be851351f29f73 Mon Sep 17 00:00:00 2001 From: Piyush Raj Chouhan Date: Mon, 28 Oct 2024 15:55:16 +0000 Subject: [PATCH 42/52] ALSA: hda/realtek: Add subwoofer quirk for Infinix ZERO BOOK 13 Infinix ZERO BOOK 13 has a 2+2 speaker system which isn't probed correctly. This patch adds a quirk with the proper pin connections. Also The mic in this laptop suffers too high gain resulting in mostly fan noise being recorded, This patch Also limit mic boost. HW Probe for device; https://linux-hardware.org/?probe=a2e892c47b Test: All 4 speaker works, Mic has low noise. Signed-off-by: Piyush Raj Chouhan Link: https://patch.msgid.link/20241028155516.15552-1-piyuschouhan1598@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 784ac058418f..7f4926194e50 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7552,6 +7552,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13, ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -7998,6 +7999,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_U7x7_headset_mic, }, + [ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170151 }, /* use as internal speaker (LFE) */ + { 0x1b, 0x90170152 }, /* use as internal speaker (back) */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, [ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -11003,6 +11014,7 @@ static const struct snd_pci_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(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), SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), From cc8475a07cf34891bf11a63025659d3537b638ef Mon Sep 17 00:00:00 2001 From: Dmitry Yashin Date: Tue, 29 Oct 2024 02:33:12 +0500 Subject: [PATCH 43/52] ASoC: dt-bindings: rockchip,rk3308-codec: add port property Fix DTB warnings when rk3308-codec used with audio-graph-card by documenting port property: codec@ff560000: 'port' does not match any of the regexes: 'pinctrl-[0-9]+' Signed-off-by: Dmitry Yashin Reviewed-by: Luca Ceresoli Link: https://patch.msgid.link/20241028213314.476776-2-dmt.yashin@gmail.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/rockchip,rk3308-codec.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml index ecf3d7d968c8..2cf229a076f0 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml @@ -48,6 +48,10 @@ properties: - const: mclk_rx - const: hclk + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + resets: maxItems: 1 From 041db4bbe04e8e0b48350b3bbbd9a799794d5c1e Mon Sep 17 00:00:00 2001 From: Alexey Klimov Date: Tue, 22 Oct 2024 04:31:30 +0100 Subject: [PATCH 44/52] ASoC: codecs: wcd937x: add missing LO Switch control The wcd937x supports also AUX input but the control that sets correct soundwire port for this is missing. This control is required for audio playback, for instance, on qrb4210 RB2 board as well as on other SoCs. Reported-by: Adam Skladowski Reported-by: Prasad Kumpatla Suggested-by: Adam Skladowski Suggested-by: Prasad Kumpatla Cc: Srinivas Kandagatla Cc: Mohammad Rafi Shaik Signed-off-by: Alexey Klimov Link: https://patch.msgid.link/20241022033132.787416-2-alexey.klimov@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd937x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index 45f32d281908..0f0d2537d322 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -2049,6 +2049,8 @@ static const struct snd_kcontrol_new wcd937x_snd_controls[] = { wcd937x_get_swr_port, wcd937x_set_swr_port), SOC_SINGLE_EXT("HPHR Switch", WCD937X_HPH_R, 0, 1, 0, wcd937x_get_swr_port, wcd937x_set_swr_port), + SOC_SINGLE_EXT("LO Switch", WCD937X_LO, 0, 1, 0, + wcd937x_get_swr_port, wcd937x_set_swr_port), SOC_SINGLE_EXT("ADC1 Switch", WCD937X_ADC1, 1, 1, 0, wcd937x_get_swr_port, wcd937x_set_swr_port), From 107a5c853eef5336a9846e7dd2f9184b6e3c07c7 Mon Sep 17 00:00:00 2001 From: Alexey Klimov Date: Tue, 22 Oct 2024 04:31:31 +0100 Subject: [PATCH 45/52] ASoC: codecs: wcd937x: relax the AUX PDM watchdog On a system with wcd937x, rxmacro and Qualcomm audio DSP, which is pretty common set of devices on Qualcomm platforms, and due to the order of how DAPM widgets are powered on (they are sorted), there is a small time window when wcd937x chip is online and expects the flow of incoming data but rxmacro is not yet online. When wcd937x is programmed to receive data via AUX port then its AUX PDM watchdog is enabled in wcd937x_codec_enable_aux_pa(). If due to some reasons the rxmacro and soundwire machinery are delayed to start streaming data, then there is a chance for this AUX PDM watchdog to reset the wcd937x codec. Such event is not logged as a message and only wcd937x IRQ counter is increased however there could be a lot of other reasons for that IRQ. There is a similar opportunity for such delay during DAPM widgets power down sequence. If wcd937x codec reset happens on the start of the playback, then there will be no sound and if such reset happens at the end of a playback then it may generate additional clicks and pops noises. On qrb4210 RB2 board without any debugging bits the wcd937x resets are sometimes observed at the end of a playback though not always. With some debugging messages or with some tracing enabled the AUX PDM watchdog resets the wcd937x codec at the start of a playback and there is no sound output at all. In this patch: - TIMEOUT_SEL bit in PDM_WD_CTL2 register is set to increase the watchdog reset delay to 100ms which eliminates the AUX PDM watchdog IRQs on qrb4210 RB2 board completely and decreases the number of unwanted clicks noises; - HOLD_OFF bit postpones triggering such watchdog IRQ till wcd937x codec reset which usually happens at the end of a playback. This allows to actually output some sound in case of debugging. Cc: Adam Skladowski Cc: Mohammad Rafi Shaik Cc: Prasad Kumpatla Cc: Srinivas Kandagatla Signed-off-by: Alexey Klimov Link: https://patch.msgid.link/20241022033132.787416-3-alexey.klimov@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd937x.c | 10 ++++++++-- sound/soc/codecs/wcd937x.h | 4 ++++ 2 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index 0f0d2537d322..08fb13a334a4 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -715,12 +715,17 @@ static int wcd937x_codec_enable_aux_pa(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct wcd937x_priv *wcd937x = snd_soc_component_get_drvdata(component); int hph_mode = wcd937x->hph_mode; + u8 val; switch (event) { case SND_SOC_DAPM_PRE_PMU: + val = WCD937X_DIGITAL_PDM_WD_CTL2_EN | + WCD937X_DIGITAL_PDM_WD_CTL2_TIMEOUT_SEL | + WCD937X_DIGITAL_PDM_WD_CTL2_HOLD_OFF; snd_soc_component_update_bits(component, WCD937X_DIGITAL_PDM_WD_CTL2, - BIT(0), BIT(0)); + WCD937X_DIGITAL_PDM_WD_CTL2_MASK, + val); break; case SND_SOC_DAPM_POST_PMU: usleep_range(1000, 1010); @@ -741,7 +746,8 @@ static int wcd937x_codec_enable_aux_pa(struct snd_soc_dapm_widget *w, hph_mode); snd_soc_component_update_bits(component, WCD937X_DIGITAL_PDM_WD_CTL2, - BIT(0), 0x00); + WCD937X_DIGITAL_PDM_WD_CTL2_MASK, + 0x00); break; } diff --git a/sound/soc/codecs/wcd937x.h b/sound/soc/codecs/wcd937x.h index 35f3d48bd7dd..4afa48dcaf74 100644 --- a/sound/soc/codecs/wcd937x.h +++ b/sound/soc/codecs/wcd937x.h @@ -391,6 +391,10 @@ #define WCD937X_DIGITAL_PDM_WD_CTL0 0x3465 #define WCD937X_DIGITAL_PDM_WD_CTL1 0x3466 #define WCD937X_DIGITAL_PDM_WD_CTL2 0x3467 +#define WCD937X_DIGITAL_PDM_WD_CTL2_HOLD_OFF BIT(2) +#define WCD937X_DIGITAL_PDM_WD_CTL2_TIMEOUT_SEL BIT(1) +#define WCD937X_DIGITAL_PDM_WD_CTL2_EN BIT(0) +#define WCD937X_DIGITAL_PDM_WD_CTL2_MASK GENMASK(2, 0) #define WCD937X_DIGITAL_INTR_MODE 0x346A #define WCD937X_DIGITAL_INTR_MASK_0 0x346B #define WCD937X_DIGITAL_INTR_MASK_1 0x346C From 4413665dd6c528b31284119e3571c25f371e1c36 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jan=20Sch=C3=A4r?= Date: Tue, 29 Oct 2024 23:12:49 +0100 Subject: [PATCH 46/52] ALSA: usb-audio: Add quirks for Dell WD19 dock MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The WD19 family of docks has the same audio chipset as the WD15. This change enables jack detection on the WD19. We don't need the dell_dock_mixer_init quirk for the WD19. It is only needed because of the dell_alc4020_map quirk for the WD15 in mixer_maps.c, which disables the volume controls. Even for the WD15, this quirk was apparently only needed when the dock firmware was not updated. Signed-off-by: Jan Schär Cc: Link: https://patch.msgid.link/20241029221249.15661-1-jan@jschaer.ch Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 2a9594f34dac..6456e87e2f39 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -4042,6 +4042,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; err = dell_dock_mixer_init(mixer); break; + case USB_ID(0x0bda, 0x402e): /* Dell WD19 dock */ + err = dell_dock_mixer_create(mixer); + break; case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */ case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */ From 0b04fbe886b4274c8e5855011233aaa69fec6e75 Mon Sep 17 00:00:00 2001 From: Christoffer Sandberg Date: Tue, 29 Oct 2024 16:16:52 +0100 Subject: [PATCH 47/52] ALSA: hda/realtek: Fix headset mic on TUXEDO Gemini 17 Gen3 Quirk is needed to enable headset microphone on missing pin 0x19. Signed-off-by: Christoffer Sandberg Signed-off-by: Werner Sembach Cc: Link: https://patch.msgid.link/20241029151653.80726-1-wse@tuxedocomputers.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 7f4926194e50..e06a6fdc0bab 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10750,6 +10750,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x2624, "Clevo L240TU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x28c1, "Clevo V370VND", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), From e49370d769e71456db3fbd982e95bab8c69f73e8 Mon Sep 17 00:00:00 2001 From: Christoffer Sandberg Date: Tue, 29 Oct 2024 16:16:53 +0100 Subject: [PATCH 48/52] ALSA: hda/realtek: Fix headset mic on TUXEDO Stellaris 16 Gen6 mb1 Quirk is needed to enable headset microphone on missing pin 0x19. Signed-off-by: Christoffer Sandberg Signed-off-by: Werner Sembach Cc: Link: https://patch.msgid.link/20241029151653.80726-2-wse@tuxedocomputers.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 e06a6fdc0bab..571fa8a6c9e1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11008,6 +11008,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x1387, "TongFang GMxIXxx", ALC2XX_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1d05, 0x1409, "TongFang GMxIXxx", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1d17, 0x3288, "Haier Boyue G42", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), From c9363bbb0f68dd1ddb8be7bbfe958cdfcd38d851 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jaros=C5=82aw=20Janik?= Date: Wed, 30 Oct 2024 18:18:12 +0100 Subject: [PATCH 49/52] Revert "ALSA: hda/conexant: Mute speakers at suspend / shutdown" MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Commit 4f61c8fe3520 ("ALSA: hda/conexant: Mute speakers at suspend / shutdown") mutes speakers on system shutdown or whenever HDA controller is suspended by PM; this however interacts badly with Thinkpad's ACPI firmware behavior which uses beeps to signal various events (enter/leave suspend or hibernation, AC power connect/disconnect, low battery, etc.); now those beeps are either muted altogether (for suspend/hibernate/ shutdown related events) or work more or less randomly (eg. AC plug/unplug is only audible when you are playing music at the moment, because HDA device is likely in suspend mode otherwise). Since the original bug report mentioned in 4f61c8fe3520 complained about Lenovo's Thinkpad laptop - revert this commit altogether. Fixes: 4f61c8fe3520 ("ALSA: hda/conexant: Mute speakers at suspend / shutdown") Signed-off-by: Jarosław Janik Link: https://patch.msgid.link/20241030171813.18941-2-jaroslaw.janik@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c74f6742c359..b2bcdf76da30 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -205,8 +205,6 @@ static void cx_auto_shutdown(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - snd_hda_gen_shutup_speakers(codec); - /* Turn the problematic codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); From c2d188e137e77294323132a760a4608321a36a70 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Nov 2024 11:07:34 +0100 Subject: [PATCH 50/52] ALSA: ump: Don't enumeration invalid groups for legacy rawmidi The legacy rawmidi tries to enumerate all possible UMP groups belonging to the UMP endpoint. But currently it shows all 16 ports when the UMP endpoint is configured with static blocks, although most of them may be unused. There was already a fix for the sequencer client side to ignore such groups in the commit 3bfd7c0ba184 ("ALSA: seq: ump: Skip useless ports for static blocks"), and this commit is a similar fix for UMP rawmidi devices; it adds simply the check for the validity of each group that has been already parsed. (Note that the group info was moved to snd_ump_endpoint.groups[] by the commit 0642a3c5cacc0321c755 ("ALSA: ump: Update substream name from assigned FB names")). Link: https://patch.msgid.link/20241104100735.16127-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/ump.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/ump.c b/sound/core/ump.c index cf22a17e38dd..7d59a0a9b037 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -1233,7 +1233,7 @@ static int fill_legacy_mapping(struct snd_ump_endpoint *ump) num = 0; for (i = 0; i < SNDRV_UMP_MAX_GROUPS; i++) - if (group_maps & (1U << i)) + if ((group_maps & (1U << i)) && ump->groups[i].valid) ump->legacy_mapping[num++] = i; return num; From 8abbf1f01d6a2ef9f911f793e30f7382154b5a3a Mon Sep 17 00:00:00 2001 From: Murad Masimov Date: Fri, 1 Nov 2024 21:55:13 +0300 Subject: [PATCH 51/52] ALSA: firewire-lib: fix return value on fail in amdtp_tscm_init() If amdtp_stream_init() fails in amdtp_tscm_init(), the latter returns zero, though it's supposed to return error code, which is checked inside init_stream() in file tascam-stream.c. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 47faeea25ef3 ("ALSA: firewire-tascam: add data block processing layer") Signed-off-by: Murad Masimov Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20241101185517.1819-1-m.masimov@maxima.ru --- sound/firewire/tascam/amdtp-tascam.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c index 0b42d6559008..079afa4bd381 100644 --- a/sound/firewire/tascam/amdtp-tascam.c +++ b/sound/firewire/tascam/amdtp-tascam.c @@ -238,7 +238,7 @@ int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit, err = amdtp_stream_init(s, unit, dir, flags, fmt, process_ctx_payloads, sizeof(struct amdtp_tscm)); if (err < 0) - return 0; + return err; if (dir == AMDTP_OUT_STREAM) { // Use fixed value for FDF field. From dabc44c28f118910dea96244d903f0c270225669 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2024 13:02:17 +0100 Subject: [PATCH 52/52] ALSA: usb-audio: Add quirk for HP 320 FHD Webcam HP 320 FHD Webcam (03f0:654a) seems to have flaky firmware like other webcam devices that don't like the frequency inquiries. Also, Mic Capture Volume has an invalid resolution, hence fix it to be 16 (as a blind shot). Link: https://bugzilla.suse.com/show_bug.cgi?id=1232768 Cc: Link: https://patch.msgid.link/20241105120220.5740-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 1 + sound/usb/quirks.c | 2 ++ 2 files changed, 3 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 9945ae55b0d0..bd67027c7677 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1205,6 +1205,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } break; case USB_ID(0x1bcf, 0x2283): /* NexiGo N930AF FHD Webcam */ + case USB_ID(0x03f0, 0x654a): /* HP 320 FHD Webcam */ if (!strcmp(kctl->id.name, "Mic Capture Volume")) { usb_audio_info(chip, "set resolution quirk: cval->res = 16\n"); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e6278a245795..c5fd180357d1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2114,6 +2114,8 @@ struct usb_audio_quirk_flags_table { static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { /* Device matches */ + DEVICE_FLG(0x03f0, 0x654a, /* HP 320 FHD Webcam */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x041e, 0x3000, /* Creative SB Extigy */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x041e, 0x4080, /* Creative Live Cam VF0610 */