From 49da1463c9e3d2082276c3e0e2a8b65a88711cd2 Mon Sep 17 00:00:00 2001 From: Zichen Xie Date: Sun, 6 Oct 2024 15:57:37 -0500 Subject: [PATCH 01/17] 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 02/17] 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 03/17] 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 04/17] 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 05/17] 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 06/17] 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 07/17] 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 08/17] 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 09/17] 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 10/17] 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 11/17] 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 12/17] 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 13/17] 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 14/17] 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 @@ audio_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 15/17] 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 16/17] 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 17/17] 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