From 49da1463c9e3d2082276c3e0e2a8b65a88711cd2 Mon Sep 17 00:00:00 2001 From: Zichen Xie Date: Sun, 6 Oct 2024 15:57:37 -0500 Subject: 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, -- cgit From 8380dbf1b9ef66e3ce6c1d660fd7259637c2a929 Mon Sep 17 00:00:00 2001 From: Miquel Raynal Date: Thu, 3 Oct 2024 10:36:11 +0200 Subject: 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 --- .../devicetree/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 -- cgit From 0dbb186c3510cad4e9f443e801bf2e6ab5770c00 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Tue, 8 Oct 2024 10:37:58 +0200 Subject: 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 -- cgit From 0a5c40393b123f3f08e428143985ab0c5ddb4d28 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 8 Oct 2024 14:43:44 +0530 Subject: 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; } -- cgit From 494ddacd4a2ae5fd1c46ea49364eaab4fc1e5461 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 8 Oct 2024 14:43:45 +0530 Subject: 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); -- cgit From 9814c1447f9cc67c9e88e0a4423de3a496078360 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Tue, 8 Oct 2024 09:07:10 +0300 Subject: 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, -- cgit From 3fe9f5882cf71573516749b0bb687ef88f470d1d Mon Sep 17 00:00:00 2001 From: Benjamin Bara Date: Tue, 8 Oct 2024 13:36:14 +0200 Subject: 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); -- cgit From a6134e7b4d4a14e0942f113a6df1d518baa2a0a4 Mon Sep 17 00:00:00 2001 From: Binbin Zhou Date: Wed, 9 Oct 2024 15:52:27 +0800 Subject: 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) { -- cgit From d0e806b0cc6260b59c65e606034a63145169c04c Mon Sep 17 00:00:00 2001 From: Alexey Klimov Date: Wed, 9 Oct 2024 22:39:22 +0100 Subject: 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) -- cgit From 251ce34a446ef0e1d6acd65cf5947abd5d10b8b6 Mon Sep 17 00:00:00 2001 From: Zhu Jun Date: Wed, 9 Oct 2024 00:39:38 -0700 Subject: 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; } -- cgit From 9eb2142a2ae8c8fdfce2aaa4c110f5a6f6b0b56e Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Wed, 9 Oct 2024 10:12:30 +0200 Subject: 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 -- cgit From 182fff3a2aafe4e7f3717a0be9df2fe2ed1a77de Mon Sep 17 00:00:00 2001 From: Christian Heusel Date: Thu, 10 Oct 2024 15:32:11 +0200 Subject: 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 = { -- cgit From ca2803fadfd239abf155ef4a563b22a9507ee4b2 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Thu, 10 Oct 2024 19:20:32 +0100 Subject: 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; } -- cgit From 9b064d200aa8fee9d1d7ced05d8a617e45966715 Mon Sep 17 00:00:00 2001 From: Lad Prabhakar Date: Thu, 10 Oct 2024 15:14:32 +0100 Subject: 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); -- cgit From 54c805c1eb264c839fa3027d0073bb7f323b0722 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 11 Oct 2024 12:53:53 +0800 Subject: 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"); -- cgit From b930d8647869802a0d430aae6b1b05c3acb24a41 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sat, 12 Oct 2024 12:09:57 +0200 Subject: 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 -- cgit From 3692a4ccacf3c44249e584aea3ae8568f953e7e4 Mon Sep 17 00:00:00 2001 From: Andrei Simion Date: Mon, 14 Oct 2024 12:28:31 +0300 Subject: 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 -- cgit From 9822b4c90d77e3c6555fb21c459c4a61c6a8619f Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:07 +0800 Subject: 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; -- cgit From 6e38a7e098d32d128b00b42a536151de9ea1340b Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:08 +0800 Subject: 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; -- cgit From c78f1e15e46ac82607eed593b22992fd08644d96 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:09 +0800 Subject: 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 -- cgit From ab5593793e9088abcddce30ba8e376e31b7285fd Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:10 +0800 Subject: 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) { -- cgit From b0867999e3282378a0b26a7ad200233044d31eca Mon Sep 17 00:00:00 2001 From: Ilya Dudikov Date: Wed, 16 Oct 2024 10:40:37 +0700 Subject: 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 = { -- cgit From 6924565a04e5f424c95e6d894584e3059f257373 Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Wed, 16 Oct 2024 11:07:03 +0800 Subject: 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 --- sound/soc/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); -- cgit From 740883fa6c7262036769aa54b50609c8043977e0 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin Date: Wed, 16 Oct 2024 23:58:10 +0200 Subject: 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 -- cgit From 9fc9ef05727ccb45fd881770f2aa5c3774b2e8e2 Mon Sep 17 00:00:00 2001 From: Alexey Klimov Date: Wed, 16 Oct 2024 23:10:49 +0100 Subject: 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) -- cgit From da95e891dd5d5de6c5ebc010bd028a2e028de093 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Thu, 17 Oct 2024 16:15:07 +0900 Subject: 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, -- cgit From 038fa6ddf5d22694f61ff7a7a53c8887c6b08c45 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Mon, 21 Oct 2024 06:15:44 +0000 Subject: 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; -- cgit From b9a8ecf81066e01e8a3de35517481bc5aa0439e5 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 14 Oct 2024 13:38:33 +0800 Subject: 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)) -- cgit From db7e59e6a39a4d3d54ca8197c796557e6d480b0d Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sat, 12 Oct 2024 12:11:08 +0200 Subject: 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 = { -- cgit