From 9822b4c90d77e3c6555fb21c459c4a61c6a8619f Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:07 +0800 Subject: [PATCH 01/34] ASoC: SOF: ipc4-topology: Do not set ALH node_id for aggregated DAIs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For aggregated DAIs, the node ID is set to the group_id during the DAI widget's ipc_prepare op. With the current logic, setting the dai_index for node_id in the dai_config is redundant as it will be overwritten with the group_id anyway. Removing it will also prevent any accidental clearing/resetting of the group_id for aggregated DAIs due to the dai_config calls could that happen before the allocated group_id is freed. Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao All: stable@vger.kernel.org # 6.10.x 6.11.x Link: https://patch.msgid.link/20241016032910.14601-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 87be7f16e8c2..240fee2166d1 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -3129,9 +3129,20 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * * group_id during copier's ipc_prepare op. */ if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { + struct sof_ipc4_alh_configuration_blob *blob; + + blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; ipc4_copier->dai_index = data->dai_node_id; - copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; - copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_node_id); + + /* + * no need to set the node_id for aggregated DAI's. These will be assigned + * a group_id during widget ipc_prepare + */ + if (blob->alh_cfg.device_count == 1) { + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= + SOF_IPC4_NODE_INDEX(data->dai_node_id); + } } break; From 6e38a7e098d32d128b00b42a536151de9ea1340b Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:08 +0800 Subject: [PATCH 02/34] ASoC: SOF: Intel: hda: Handle prepare without close for non-HDA DAI's MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When a PCM is restarted after a snd_pcm_drain/snd_pcm_drop(), the prepare callback will be invoked and the hw_params will be set again. For the HDA DAI's, the hw_params function handles this case already but not for the non-HDA DAI's. So, add the check for link_prepared to verify if the hw_params should be done again or not. Additionally, for SDW DAI's reset the PCMSyCM registers as would be done in the case of a start after a hw_free. Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao All: stable@vger.kernel.org # 6.10.x 6.11.x Link: https://patch.msgid.link/20241016032910.14601-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 36 +++++++++++++++++++++++++++++++---- 1 file changed, 32 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 1c823f9eea57..8cccf38967e7 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -370,6 +370,13 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream, return -EINVAL; } + sdev = widget_to_sdev(w); + hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream); + + /* nothing more to do if the link is already prepared */ + if (hext_stream && hext_stream->link_prepared) + return 0; + /* use HDaudio stream handling */ ret = hda_dai_hw_params_data(substream, params, cpu_dai, data, flags); if (ret < 0) { @@ -377,7 +384,6 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream, return ret; } - sdev = widget_to_sdev(w); if (sdev->dspless_mode_selected) return 0; @@ -482,6 +488,31 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, int ret; int i; + ops = hda_dai_get_ops(substream, cpu_dai); + if (!ops) { + dev_err(cpu_dai->dev, "DAI widget ops not set\n"); + return -EINVAL; + } + + sdev = widget_to_sdev(w); + hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream); + + /* nothing more to do if the link is already prepared */ + if (hext_stream && hext_stream->link_prepared) + return 0; + + /* + * reset the PCMSyCM registers to handle a prepare callback when the PCM is restarted + * due to xruns or after a call to snd_pcm_drain/drop() + */ + ret = hdac_bus_eml_sdw_map_stream_ch(sof_to_bus(sdev), link_id, cpu_dai->id, + 0, 0, substream->stream); + if (ret < 0) { + dev_err(cpu_dai->dev, "%s: hdac_bus_eml_sdw_map_stream_ch failed %d\n", + __func__, ret); + return ret; + } + data.dai_index = (link_id << 8) | cpu_dai->id; data.dai_node_id = intel_alh_id; ret = non_hda_dai_hw_params_data(substream, params, cpu_dai, &data, flags); @@ -490,10 +521,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, return ret; } - ops = hda_dai_get_ops(substream, cpu_dai); - sdev = widget_to_sdev(w); hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream); - if (!hext_stream) return -ENODEV; From c78f1e15e46ac82607eed593b22992fd08644d96 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:09 +0800 Subject: [PATCH 03/34] soundwire: intel_ace2x: Send PDI stream number during prepare MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In the case of a prepare callback after an xrun or when the PCM is restarted after a call to snd_pcm_drain/snd_pcm_drop, avoid reprogramming the SHIM registers but send the PDI stream number so that the link DMA data can be set. This is needed for the case that the DMA data is cleared when the PCM is stopped and restarted without being closed. Link: https://github.com/thesofproject/sof/issues/9502 Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao Acked-by: Vinod Koul All: stable@vger.kernel.org # 6.10.x 6.11.x Link: https://patch.msgid.link/20241016032910.14601-4-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- drivers/soundwire/intel_ace2x.c | 19 ++++++------------- 1 file changed, 6 insertions(+), 13 deletions(-) diff --git a/drivers/soundwire/intel_ace2x.c b/drivers/soundwire/intel_ace2x.c index fff312c6968d..4f3dd70d6a1a 100644 --- a/drivers/soundwire/intel_ace2x.c +++ b/drivers/soundwire/intel_ace2x.c @@ -376,11 +376,12 @@ static int intel_hw_params(struct snd_pcm_substream *substream, static int intel_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_cdns_dai_runtime *dai_runtime; + struct snd_pcm_hw_params *hw_params; int ch, dir; - int ret = 0; dai_runtime = cdns->dai_runtime_array[dai->id]; if (!dai_runtime) { @@ -389,12 +390,8 @@ static int intel_prepare(struct snd_pcm_substream *substream, return -EIO; } + hw_params = &rtd->dpcm[substream->stream].hw_params; if (dai_runtime->suspended) { - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_pcm_hw_params *hw_params; - - hw_params = &rtd->dpcm[substream->stream].hw_params; - dai_runtime->suspended = false; /* @@ -415,15 +412,11 @@ static int intel_prepare(struct snd_pcm_substream *substream, /* the SHIM will be configured in the callback functions */ sdw_cdns_config_stream(cdns, ch, dir, dai_runtime->pdi); - - /* Inform DSP about PDI stream number */ - ret = intel_params_stream(sdw, substream, dai, - hw_params, - sdw->instance, - dai_runtime->pdi->intel_alh_id); } - return ret; + /* Inform DSP about PDI stream number */ + return intel_params_stream(sdw, substream, dai, hw_params, sdw->instance, + dai_runtime->pdi->intel_alh_id); } static int From ab5593793e9088abcddce30ba8e376e31b7285fd Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 16 Oct 2024 11:29:10 +0800 Subject: [PATCH 04/34] ASoC: SOF: Intel: hda: Always clean up link DMA during stop MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This is required to reset the DMA read/write pointers when the stream is prepared and restarted after a call to snd_pcm_drain()/snd_pcm_drop(). Also, now that the stream is reset during stop, do not save LLP registers in the case of STOP/suspend to avoid erroneous delay reporting. Link: https://github.com/thesofproject/sof/issues/9502 Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao All: stable@vger.kernel.org # 6.10.x 6.11.x Link: https://patch.msgid.link/20241016032910.14601-5-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai-ops.c | 23 ++++++++++------------- sound/soc/sof/intel/hda-dai.c | 1 + 2 files changed, 11 insertions(+), 13 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 484c76147885..92681ca7f24d 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -346,20 +346,21 @@ static int hda_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: snd_hdac_ext_stream_start(hext_stream); break; - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - snd_hdac_ext_stream_clear(hext_stream); - /* - * Save the LLP registers in case the stream is - * restarting due PAUSE_RELEASE, or START without a pcm - * close/open since in this case the LLP register is not reset - * to 0 and the delay calculation will return with invalid - * results. + * Save the LLP registers since in case of PAUSE the LLP + * register are not reset to 0, the delay calculation will use + * the saved offsets for compensating the delay calculation. */ hext_stream->pplcllpl = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); hext_stream->pplcllpu = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); + snd_hdac_ext_stream_clear(hext_stream); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + hext_stream->pplcllpl = 0; + hext_stream->pplcllpu = 0; + snd_hdac_ext_stream_clear(hext_stream); break; default: dev_err(sdev->dev, "unknown trigger command %d\n", cmd); @@ -512,7 +513,6 @@ static const struct hda_dai_widget_dma_ops sdw_ipc4_chain_dma_ops = { static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, struct snd_pcm_substream *substream, int cmd) { - struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream); struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); switch (cmd) { @@ -527,9 +527,6 @@ static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *c if (ret < 0) return ret; - if (cmd == SNDRV_PCM_TRIGGER_STOP) - return hda_link_dma_cleanup(substream, hext_stream, cpu_dai); - break; } case SNDRV_PCM_TRIGGER_PAUSE_PUSH: diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 8cccf38967e7..ac505c7ad342 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -302,6 +302,7 @@ static int __maybe_unused hda_dai_trigger(struct snd_pcm_substream *substream, i } switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: ret = hda_link_dma_cleanup(substream, hext_stream, dai); if (ret < 0) { From b0867999e3282378a0b26a7ad200233044d31eca Mon Sep 17 00:00:00 2001 From: Ilya Dudikov Date: Wed, 16 Oct 2024 10:40:37 +0700 Subject: [PATCH 05/34] ASoC: amd: yc: Fix non-functional mic on ASUS E1404FA ASUS Vivobook E1404FA needs a quirks-table entry for the internal microphone to function properly. Signed-off-by: Ilya Dudikov Link: https://patch.msgid.link/20241016034038.13481-1-ilyadud25@gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 98f9237b7ad7..438865d5e376 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -325,6 +325,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M6500RC"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "E1404FA"), + } + }, { .driver_data = &acp6x_card, .matches = { From 6924565a04e5f424c95e6d894584e3059f257373 Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Wed, 16 Oct 2024 11:07:03 +0800 Subject: [PATCH 06/34] ASoC: Intel: soc-acpi: lnl: Add match entry for TM2 laptops MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add a new match table entry on Lunarlake for the TM2 laptops with rt713 and rt1318. Signed-off-by: Derek Fang Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Link: https://patch.msgid.link/20241016030703.13669-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-lnl-match.c | 38 +++++++++++++++++++ 1 file changed, 38 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index 3c4e0c7ca8ee..094ed4b27cb0 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -225,6 +225,15 @@ static const struct snd_soc_acpi_adr_device rt1316_3_group1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1318_1_adr[] = { + { + .adr = 0x000133025D131801ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt1318-1" + } +}; + static const struct snd_soc_acpi_adr_device rt1318_1_group1_adr[] = { { .adr = 0x000130025D131801ull, @@ -243,6 +252,15 @@ static const struct snd_soc_acpi_adr_device rt1318_2_group1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt713_0_adr[] = { + { + .adr = 0x000031025D071301ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt713" + } +}; + static const struct snd_soc_acpi_adr_device rt714_0_adr[] = { { .adr = 0x000030025D071401ull, @@ -378,6 +396,20 @@ static const struct snd_soc_acpi_link_adr lnl_sdw_rt1318_l12_rt714_l0[] = { {} }; +static const struct snd_soc_acpi_link_adr lnl_sdw_rt713_l0_rt1318_l1[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt713_0_adr), + .adr_d = rt713_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1318_1_adr), + .adr_d = rt1318_1_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { /* mockup tests need to be first */ @@ -447,6 +479,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-lnl-rt1318-l12-rt714-l0.tplg" }, + { + .link_mask = BIT(0) | BIT(1), + .links = lnl_sdw_rt713_l0_rt1318_l1, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-lnl-rt713-l0-rt1318-l1.tplg" + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_lnl_sdw_machines); From 740883fa6c7262036769aa54b50609c8043977e0 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin Date: Wed, 16 Oct 2024 23:58:10 +0200 Subject: [PATCH 07/34] ASoC: Change my e-mail to gmail Change my contact e-mail in pcm3060 driver and MAINTAINERS Signed-off-by: Kirill Marinushkin Cc: Kirill Marinushkin Cc: Liam Girdwood Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: linux-kernel@vger.kernel.org Cc: linux-sound@vger.kernel.org Link: https://patch.msgid.link/20241016215810.1544222-1-k.marinushkin@gmail.com Signed-off-by: Mark Brown --- MAINTAINERS | 2 +- sound/soc/codecs/pcm3060-i2c.c | 4 ++-- sound/soc/codecs/pcm3060-spi.c | 4 ++-- sound/soc/codecs/pcm3060.c | 4 ++-- sound/soc/codecs/pcm3060.h | 2 +- 5 files changed, 8 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index c1a2c296446c..9d6272c00fbd 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23290,7 +23290,7 @@ F: Documentation/devicetree/bindings/iio/adc/ti,lmp92064.yaml F: drivers/iio/adc/ti-lmp92064.c TI PCM3060 ASoC CODEC DRIVER -M: Kirill Marinushkin +M: Kirill Marinushkin L: linux-sound@vger.kernel.org S: Maintained F: Documentation/devicetree/bindings/sound/pcm3060.txt diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c index 5330cf46b127..3816b25a8ead 100644 --- a/sound/soc/codecs/pcm3060-i2c.c +++ b/sound/soc/codecs/pcm3060-i2c.c @@ -2,7 +2,7 @@ // // PCM3060 I2C driver // -// Copyright (C) 2018 Kirill Marinushkin +// Copyright (C) 2018 Kirill Marinushkin #include #include @@ -55,5 +55,5 @@ static struct i2c_driver pcm3060_i2c_driver = { module_i2c_driver(pcm3060_i2c_driver); MODULE_DESCRIPTION("PCM3060 I2C driver"); -MODULE_AUTHOR("Kirill Marinushkin "); +MODULE_AUTHOR("Kirill Marinushkin "); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c index 3b79734b832b..6095841f2f56 100644 --- a/sound/soc/codecs/pcm3060-spi.c +++ b/sound/soc/codecs/pcm3060-spi.c @@ -2,7 +2,7 @@ // // PCM3060 SPI driver // -// Copyright (C) 2018 Kirill Marinushkin +// Copyright (C) 2018 Kirill Marinushkin #include #include @@ -55,5 +55,5 @@ static struct spi_driver pcm3060_spi_driver = { module_spi_driver(pcm3060_spi_driver); MODULE_DESCRIPTION("PCM3060 SPI driver"); -MODULE_AUTHOR("Kirill Marinushkin "); +MODULE_AUTHOR("Kirill Marinushkin "); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 586ec8c7246c..8974200652e7 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -2,7 +2,7 @@ // // PCM3060 codec driver // -// Copyright (C) 2018 Kirill Marinushkin +// Copyright (C) 2018 Kirill Marinushkin #include #include @@ -343,5 +343,5 @@ int pcm3060_probe(struct device *dev) EXPORT_SYMBOL(pcm3060_probe); MODULE_DESCRIPTION("PCM3060 codec driver"); -MODULE_AUTHOR("Kirill Marinushkin "); +MODULE_AUTHOR("Kirill Marinushkin "); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index 5e1185e7b03d..1b96835600b4 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -2,7 +2,7 @@ /* * PCM3060 codec driver * - * Copyright (C) 2018 Kirill Marinushkin + * Copyright (C) 2018 Kirill Marinushkin */ #ifndef _SND_SOC_PCM3060_H From 9fc9ef05727ccb45fd881770f2aa5c3774b2e8e2 Mon Sep 17 00:00:00 2001 From: Alexey Klimov Date: Wed, 16 Oct 2024 23:10:49 +0100 Subject: [PATCH 08/34] ASoC: codecs: lpass-rx-macro: fix RXn(rx,n) macro for DSM_CTL and SEC7 regs Turns out some registers of pre-2.5 version of rxmacro codecs are not located at the expected offsets but 0xc further away in memory. So far the detected registers are CDC_RX_RX2_RX_PATH_SEC7 and CDC_RX_RX2_RX_PATH_DSM_CTL. CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macro incorrectly generates the address 0x540 for RX2 but it should be 0x54C and it also overwrites CDC_RX_RX2_RX_PATH_SEC7 which is located at 0x540. The same goes for CDC_RX_RXn_RX_PATH_SEC7(rx, n). Fix this by introducing additional rxn_reg_stride2 offset. For 2.5 version and above this offset will be equal to 0. With such change the corresponding RXn() macros will generate the same values for 2.5 codec version for all RX paths and the same old values for pre-2.5 version for RX0 and RX1. However for the latter case with RX2 path it will also add rxn_reg_stride2 on top. While at this, also remove specific if-check for INTERP_AUX from rx_macro_digital_mute() and rx_macro_enable_interp_clk(). These if-check was used to handle such special offset for AUX interpolator but since CDC_RX_RXn_RX_PATH_SEC7(rx, n) and CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macros will generate the correst addresses of dsm register, they are no longer needed. Cc: Srinivas Kandagatla Cc: Krzysztof Kozlowski Signed-off-by: Alexey Klimov Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20241016221049.1145101-1-alexey.klimov@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-rx-macro.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index ef7a70fa6966..febbbe073962 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -202,12 +202,14 @@ #define CDC_RX_RXn_RX_PATH_SEC3(rx, n) (0x042c + rx->rxn_reg_stride * n) #define CDC_RX_RX0_RX_PATH_SEC4 (0x0430) #define CDC_RX_RX0_RX_PATH_SEC7 (0x0434) -#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) (0x0434 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) \ + (0x0434 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0)) #define CDC_RX_DSM_OUT_DELAY_SEL_MASK GENMASK(2, 0) #define CDC_RX_DSM_OUT_DELAY_TWO_SAMPLE 0x2 #define CDC_RX_RX0_RX_PATH_MIX_SEC0 (0x0438) #define CDC_RX_RX0_RX_PATH_MIX_SEC1 (0x043C) -#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) (0x0440 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) \ + (0x0440 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0)) #define CDC_RX_RXn_DSM_CLK_EN_MASK BIT(0) #define CDC_RX_RX0_RX_PATH_DSM_CTL (0x0440) #define CDC_RX_RX0_RX_PATH_DSM_DATA1 (0x0444) @@ -645,6 +647,7 @@ struct rx_macro { int rx_mclk_cnt; enum lpass_codec_version codec_version; int rxn_reg_stride; + int rxn_reg_stride2; bool is_ear_mode_on; bool hph_pwr_mode; bool hph_hd2_mode; @@ -1929,9 +1932,6 @@ static int rx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) CDC_RX_PATH_PGA_MUTE_MASK, 0x0); } - if (j == INTERP_AUX) - dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2); - int_mux_cfg0 = CDC_RX_INP_MUX_RX_INT0_CFG0 + j * 8; int_mux_cfg1 = int_mux_cfg0 + 4; int_mux_cfg0_val = snd_soc_component_read(component, int_mux_cfg0); @@ -2702,9 +2702,6 @@ static int rx_macro_enable_interp_clk(struct snd_soc_component *component, main_reg = CDC_RX_RXn_RX_PATH_CTL(rx, interp_idx); dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, interp_idx); - if (interp_idx == INTERP_AUX) - dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2); - rx_cfg2_reg = CDC_RX_RXn_RX_PATH_CFG2(rx, interp_idx); if (SND_SOC_DAPM_EVENT_ON(event)) { @@ -3821,6 +3818,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_0: case LPASS_CODEC_VERSION_2_1: rx->rxn_reg_stride = 0x80; + rx->rxn_reg_stride2 = 0xc; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_pre_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) @@ -3834,6 +3832,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_7: case LPASS_CODEC_VERSION_2_8: rx->rxn_reg_stride = 0xc0; + rx->rxn_reg_stride2 = 0x0; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) From da95e891dd5d5de6c5ebc010bd028a2e028de093 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Thu, 17 Oct 2024 16:15:07 +0900 Subject: [PATCH 09/34] ASoC: fsl_micfil: Add a flag to distinguish with different volume control types On i.MX8MM the register of volume control has positive and negative values. It is different from other platforms like i.MX8MP and i.MX93 which only have positive values. Add a volume_sx flag to use SX_TLV volume control for this kind of platform. Use common TLV volume control for other platforms. Fixes: cdfa92eb90f5 ("ASoC: fsl_micfil: Correct the number of steps on SX controls") Signed-off-by: Chancel Liu Reviewed-by: Daniel Baluta Link: https://patch.msgid.link/20241017071507.2577786-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 43 +++++++++++++++++++++++++++++++++++++- 1 file changed, 42 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 193be098fa5e..84638c1a2863 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -67,6 +67,7 @@ struct fsl_micfil_soc_data { bool imx; bool use_edma; bool use_verid; + bool volume_sx; u64 formats; }; @@ -76,6 +77,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mm = { .fifo_depth = 8, .dataline = 0xf, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .volume_sx = true, }; static struct fsl_micfil_soc_data fsl_micfil_imx8mp = { @@ -84,6 +86,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mp = { .fifo_depth = 32, .dataline = 0xf, .formats = SNDRV_PCM_FMTBIT_S32_LE, + .volume_sx = false, }; static struct fsl_micfil_soc_data fsl_micfil_imx93 = { @@ -94,6 +97,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx93 = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .use_edma = true, .use_verid = true, + .volume_sx = false, }; static const struct of_device_id fsl_micfil_dt_ids[] = { @@ -317,7 +321,26 @@ static int hwvad_detected(struct snd_kcontrol *kcontrol, return 0; } -static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { +static const struct snd_kcontrol_new fsl_micfil_volume_controls[] = { + SOC_SINGLE_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(0), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(1), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH2 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(2), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH3 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(3), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH4 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(4), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH5 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(5), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH6 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(6), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(7), 0xF, 0, gain_tlv), +}; + +static const struct snd_kcontrol_new fsl_micfil_volume_sx_controls[] = { SOC_SINGLE_SX_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, MICFIL_OUTGAIN_CHX_SHIFT(0), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, @@ -334,6 +357,9 @@ static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { MICFIL_OUTGAIN_CHX_SHIFT(6), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, MICFIL_OUTGAIN_CHX_SHIFT(7), 0x8, 0xF, gain_tlv), +}; + +static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { SOC_ENUM_EXT("MICFIL Quality Select", fsl_micfil_quality_enum, micfil_quality_get, micfil_quality_set), @@ -801,6 +827,20 @@ static int fsl_micfil_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } +static int fsl_micfil_component_probe(struct snd_soc_component *component) +{ + struct fsl_micfil *micfil = snd_soc_component_get_drvdata(component); + + if (micfil->soc->volume_sx) + snd_soc_add_component_controls(component, fsl_micfil_volume_sx_controls, + ARRAY_SIZE(fsl_micfil_volume_sx_controls)); + else + snd_soc_add_component_controls(component, fsl_micfil_volume_controls, + ARRAY_SIZE(fsl_micfil_volume_controls)); + + return 0; +} + static const struct snd_soc_dai_ops fsl_micfil_dai_ops = { .probe = fsl_micfil_dai_probe, .startup = fsl_micfil_startup, @@ -821,6 +861,7 @@ static struct snd_soc_dai_driver fsl_micfil_dai = { static const struct snd_soc_component_driver fsl_micfil_component = { .name = "fsl-micfil-dai", + .probe = fsl_micfil_component_probe, .controls = fsl_micfil_snd_controls, .num_controls = ARRAY_SIZE(fsl_micfil_snd_controls), .legacy_dai_naming = 1, From 038fa6ddf5d22694f61ff7a7a53c8887c6b08c45 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Mon, 21 Oct 2024 06:15:44 +0000 Subject: [PATCH 10/34] ASoC: rt722-sdca: increase clk_stop_timeout to fix clock stop issue clk_stop_timeout should be increased to 900ms to fix clock stop issue. Signed-off-by: Jack Yu Link: https://patch.msgid.link/cd26275d9fc54374a18dc016755cb72d@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index 87354bb1564e..d5c985ff5ac5 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -253,7 +253,7 @@ static int rt722_sdca_read_prop(struct sdw_slave *slave) } /* set the timeout values */ - prop->clk_stop_timeout = 200; + prop->clk_stop_timeout = 900; /* wake-up event */ prop->wake_capable = 1; From b9a8ecf81066e01e8a3de35517481bc5aa0439e5 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 14 Oct 2024 13:38:33 +0800 Subject: [PATCH 11/34] ASoC: fsl_micfil: Add sample rate constraint On some platforms, for example i.MX93, there is only one audio PLL source, so some sample rate can't be supported. If the PLL source is used for 8kHz series rates, then 11kHz series rates can't be supported. So add constraints according to the frequency of available clock sources, then alsa-lib will help to convert the unsupported rate for the driver. Signed-off-by: Shengjiu Wang Link: https://patch.msgid.link/1728884313-6778-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 38 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 84638c1a2863..0c71a73476df 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -28,6 +28,13 @@ #define MICFIL_OSR_DEFAULT 16 +#define MICFIL_NUM_RATES 7 +#define MICFIL_CLK_SRC_NUM 3 +/* clock source ids */ +#define MICFIL_AUDIO_PLL1 0 +#define MICFIL_AUDIO_PLL2 1 +#define MICFIL_CLK_EXT3 2 + enum quality { QUALITY_HIGH, QUALITY_MEDIUM, @@ -45,9 +52,12 @@ struct fsl_micfil { struct clk *mclk; struct clk *pll8k_clk; struct clk *pll11k_clk; + struct clk *clk_src[MICFIL_CLK_SRC_NUM]; struct snd_dmaengine_dai_dma_data dma_params_rx; struct sdma_peripheral_config sdmacfg; struct snd_soc_card *card; + struct snd_pcm_hw_constraint_list constraint_rates; + unsigned int constraint_rates_list[MICFIL_NUM_RATES]; unsigned int dataline; char name[32]; int irq[MICFIL_IRQ_LINES]; @@ -475,12 +485,34 @@ static int fsl_micfil_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_micfil *micfil = snd_soc_dai_get_drvdata(dai); + unsigned int rates[MICFIL_NUM_RATES] = {8000, 11025, 16000, 22050, 32000, 44100, 48000}; + int i, j, k = 0; + u64 clk_rate; if (!micfil) { dev_err(dai->dev, "micfil dai priv_data not set\n"); return -EINVAL; } + micfil->constraint_rates.list = micfil->constraint_rates_list; + micfil->constraint_rates.count = 0; + + for (j = 0; j < MICFIL_NUM_RATES; j++) { + for (i = 0; i < MICFIL_CLK_SRC_NUM; i++) { + clk_rate = clk_get_rate(micfil->clk_src[i]); + if (clk_rate != 0 && do_div(clk_rate, rates[j]) == 0) { + micfil->constraint_rates_list[k++] = rates[j]; + micfil->constraint_rates.count++; + break; + } + } + } + + if (micfil->constraint_rates.count > 0) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &micfil->constraint_rates); + return 0; } @@ -1175,6 +1207,12 @@ static int fsl_micfil_probe(struct platform_device *pdev) fsl_asoc_get_pll_clocks(&pdev->dev, &micfil->pll8k_clk, &micfil->pll11k_clk); + micfil->clk_src[MICFIL_AUDIO_PLL1] = micfil->pll8k_clk; + micfil->clk_src[MICFIL_AUDIO_PLL2] = micfil->pll11k_clk; + micfil->clk_src[MICFIL_CLK_EXT3] = devm_clk_get(&pdev->dev, "clkext3"); + if (IS_ERR(micfil->clk_src[MICFIL_CLK_EXT3])) + micfil->clk_src[MICFIL_CLK_EXT3] = NULL; + /* init regmap */ regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(regs)) From db7e59e6a39a4d3d54ca8197c796557e6d480b0d Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sat, 12 Oct 2024 12:11:08 +0200 Subject: [PATCH 12/34] ASoC: qcom: sc7280: Fix missing Soundwire runtime stream alloc Commit 15c7fab0e047 ("ASoC: qcom: Move Soundwire runtime stream alloc to soundcards") moved the allocation of Soundwire stream runtime from the Qualcomm Soundwire driver to each individual machine sound card driver, except that it forgot to update SC7280 card. Just like for other Qualcomm sound cards using Soundwire, the card driver should allocate and release the runtime. Otherwise sound playback will result in a NULL pointer dereference or other effect of uninitialized memory accesses (which was confirmed on SDM845 having similar issue). Cc: stable@vger.kernel.org Cc: Alexey Klimov Cc: Steev Klimaszewski Fixes: 15c7fab0e047 ("ASoC: qcom: Move Soundwire runtime stream alloc to soundcards") Link: https://lore.kernel.org/r/20241010054109.16938-1-krzysztof.kozlowski@linaro.org Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20241012101108.129476-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/sc7280.c | 10 +++++++++- 2 files changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 3687b9db5ed4..ca7a30ebd26a 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -209,6 +209,7 @@ config SND_SOC_SC7280 tristate "SoC Machine driver for SC7280 boards" depends on I2C && SOUNDWIRE select SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_SDW select SND_SOC_LPASS_SC7280 select SND_SOC_MAX98357A select SND_SOC_WCD938X_SDW diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c index 207ac5da4dd4..230af8d7b205 100644 --- a/sound/soc/qcom/sc7280.c +++ b/sound/soc/qcom/sc7280.c @@ -23,6 +23,7 @@ #include "common.h" #include "lpass.h" #include "qdsp6/q6afe.h" +#include "sdw.h" #define DEFAULT_MCLK_RATE 19200000 #define RT5682_PLL_FREQ (48000 * 512) @@ -316,6 +317,7 @@ static void sc7280_snd_shutdown(struct snd_pcm_substream *substream) struct snd_soc_card *card = rtd->card; struct sc7280_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; switch (cpu_dai->id) { case MI2S_PRIMARY: @@ -333,6 +335,9 @@ static void sc7280_snd_shutdown(struct snd_pcm_substream *substream) default: break; } + + data->sruntime[cpu_dai->id] = NULL; + sdw_release_stream(sruntime); } static int sc7280_snd_startup(struct snd_pcm_substream *substream) @@ -347,6 +352,8 @@ static int sc7280_snd_startup(struct snd_pcm_substream *substream) switch (cpu_dai->id) { case MI2S_PRIMARY: ret = sc7280_rt5682_init(rtd); + if (ret) + return ret; break; case SECONDARY_MI2S_RX: codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; @@ -360,7 +367,8 @@ static int sc7280_snd_startup(struct snd_pcm_substream *substream) default: break; } - return ret; + + return qcom_snd_sdw_startup(substream); } static const struct snd_soc_ops sc7280_ops = { From 032532f91a1d06d0750f16c49a9698ef5374a68f Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 24 Oct 2024 23:56:12 +0200 Subject: [PATCH 13/34] ASoC: codecs: rt5640: Always disable IRQs from rt5640_cancel_work() Disable IRQs from rt5640_cancel_work(), this fixes a crash caused by the IRQ never getting freed when the driver is unbound from the i2c_client with jack-detection active: [ 193.138780] rt5640 i2c-rt5640: ASoC: unknown pin LDO2 [ 193.138830] rt5640 i2c-rt5640: ASoC: unknown pin MICBIAS1 [ 193.671218] BUG: kernel NULL pointer dereference, address: 0000000000000078 [ 193.671239] #PF: supervisor read access in kernel mode [ 193.671248] #PF: error_code(0x0000) - not-present page ... [ 193.671531] ? asm_exc_page_fault+0x22/0x30 [ 193.671551] ? rt5640_jack_inserted+0x10/0x80 [snd_soc_rt5640] [ 193.671574] rt5640_detect_headset+0x93/0x130 [snd_soc_rt5640] [ 193.671596] rt5640_jack_work+0x93/0x355 [snd_soc_rt5640] Signed-off-by: Hans de Goede Link: https://patch.msgid.link/20241024215612.92147-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 16f3425a3e35..855139348edb 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2419,10 +2419,20 @@ static irqreturn_t rt5640_jd_gpio_irq(int irq, void *data) return IRQ_HANDLED; } -static void rt5640_cancel_work(void *data) +static void rt5640_disable_irq_and_cancel_work(void *data) { struct rt5640_priv *rt5640 = data; + if (rt5640->jd_gpio_irq_requested) { + free_irq(rt5640->jd_gpio_irq, rt5640); + rt5640->jd_gpio_irq_requested = false; + } + + if (rt5640->irq_requested) { + free_irq(rt5640->irq, rt5640); + rt5640->irq_requested = false; + } + cancel_delayed_work_sync(&rt5640->jack_work); cancel_delayed_work_sync(&rt5640->bp_work); } @@ -2463,13 +2473,7 @@ static void rt5640_disable_jack_detect(struct snd_soc_component *component) if (!rt5640->jack) return; - if (rt5640->jd_gpio_irq_requested) - free_irq(rt5640->jd_gpio_irq, rt5640); - - if (rt5640->irq_requested) - free_irq(rt5640->irq, rt5640); - - rt5640_cancel_work(rt5640); + rt5640_disable_irq_and_cancel_work(rt5640); if (rt5640->jack->status & SND_JACK_MICROPHONE) { rt5640_disable_micbias1_ovcd_irq(component); @@ -2477,8 +2481,6 @@ static void rt5640_disable_jack_detect(struct snd_soc_component *component) snd_soc_jack_report(rt5640->jack, 0, SND_JACK_BTN_0); } - rt5640->jd_gpio_irq_requested = false; - rt5640->irq_requested = false; rt5640->jd_gpio = NULL; rt5640->jack = NULL; } @@ -2798,7 +2800,8 @@ static int rt5640_suspend(struct snd_soc_component *component) if (rt5640->jack) { /* disable jack interrupts during system suspend */ disable_irq(rt5640->irq); - rt5640_cancel_work(rt5640); + cancel_delayed_work_sync(&rt5640->jack_work); + cancel_delayed_work_sync(&rt5640->bp_work); } snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); @@ -3032,7 +3035,7 @@ static int rt5640_i2c_probe(struct i2c_client *i2c) INIT_DELAYED_WORK(&rt5640->jack_work, rt5640_jack_work); /* Make sure work is stopped on probe-error / remove */ - ret = devm_add_action_or_reset(&i2c->dev, rt5640_cancel_work, rt5640); + ret = devm_add_action_or_reset(&i2c->dev, rt5640_disable_irq_and_cancel_work, rt5640); if (ret) return ret; From d48696b915527b5bcdd207a299aec03fb037eb17 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 24 Oct 2024 23:16:14 +0200 Subject: [PATCH 14/34] ASoC: Intel: bytcr_rt5640: Add support for non ACPI instantiated codec MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit On some x86 Bay Trail tablets which shipped with Android as factory OS, the DSDT is so broken that the codec needs to be manually instantatiated by the special x86-android-tablets.ko "fixup" driver for cases like this. This means that the codec-dev cannot be retrieved through its ACPI fwnode, add support to the bytcr_rt5640 machine driver for such manually instantiated rt5640 i2c_clients. An example of a tablet which needs this is the Vexia EDU ATLA 10 tablet, which has been distributed to schools in the Spanish Andalucía region. Signed-off-by: Hans de Goede Link: https://patch.msgid.link/20241024211615.79518-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 33 ++++++++++++++++++++++++--- 1 file changed, 30 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 2ed49acb4e36..8dfd91cc3668 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -32,6 +33,8 @@ #include "../atom/sst-atom-controls.h" #include "../common/soc-intel-quirks.h" +#define BYT_RT5640_FALLBACK_CODEC_DEV_NAME "i2c-rt5640" + enum { BYT_RT5640_DMIC1_MAP, BYT_RT5640_DMIC2_MAP, @@ -1698,9 +1701,33 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) codec_dev = acpi_get_first_physical_node(adev); acpi_dev_put(adev); - if (!codec_dev) - return -EPROBE_DEFER; - priv->codec_dev = get_device(codec_dev); + + if (codec_dev) { + priv->codec_dev = get_device(codec_dev); + } else { + /* + * Special case for Android tablets where the codec i2c_client + * has been manually instantiated by x86_android_tablets.ko due + * to a broken DSDT. + */ + codec_dev = bus_find_device_by_name(&i2c_bus_type, NULL, + BYT_RT5640_FALLBACK_CODEC_DEV_NAME); + if (!codec_dev) + return -EPROBE_DEFER; + + if (!i2c_verify_client(codec_dev)) { + dev_err(dev, "Error '%s' is not an i2c_client\n", + BYT_RT5640_FALLBACK_CODEC_DEV_NAME); + put_device(codec_dev); + } + + /* fixup codec name */ + strscpy(byt_rt5640_codec_name, BYT_RT5640_FALLBACK_CODEC_DEV_NAME, + sizeof(byt_rt5640_codec_name)); + + /* bus_find_device() returns a reference no need to get() */ + priv->codec_dev = codec_dev; + } /* * swap SSP0 if bytcr is detected From 0107f28f135231da22a9ad5756bb16bd5cada4d5 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 24 Oct 2024 23:16:15 +0200 Subject: [PATCH 15/34] ASoC: Intel: bytcr_rt5640: Add DMI quirk for Vexia Edu Atla 10 tablet The Vexia Edu Atla 10 tablet mostly uses the BYTCR tablet defaults, but as happens on more models it is using IN1 instead of IN3 for its internal mic and JD_SRC_JD2_IN4N instead of JD_SRC_JD1_IN4P for jack-detection. Add a DMI quirk for this to fix the internal-mic and jack-detection. Signed-off-by: Hans de Goede Link: https://patch.msgid.link/20241024211615.79518-2-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 8dfd91cc3668..54f77f57ec8e 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1132,6 +1132,21 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { /* Vexia Edu Atla 10 tablet */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* Above strings are too generic, also match on BIOS date */ + DMI_MATCH(DMI_BIOS_DATE, "08/25/2014"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { /* Voyo Winpad A15 */ .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), From 6668610b4d8ce9a3ee3ed61a9471f62fb5f05bf9 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Fri, 25 Oct 2024 11:02:21 +0200 Subject: [PATCH 16/34] ASoC: Intel: sst: Support LPE0F28 ACPI HID MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Some old Bay Trail tablets which shipped with Android as factory OS have the SST/LPE audio engine described by an ACPI device with a HID (Hardware-ID) of LPE0F28 instead of 80860F28. Add support for this. Note this uses a new sst_res_info for just the LPE0F28 case because it has a different layout for the IO-mem ACPI resources then the 80860F28. An example of a tablet which needs this is the Vexia EDU ATLA 10 tablet, which has been distributed to schools in the Spanish Andalucía region. Signed-off-by: Hans de Goede Link: https://patch.msgid.link/20241025090221.52198-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/hda/intel-dsp-config.c | 4 ++ sound/soc/intel/atom/sst/sst_acpi.c | 64 +++++++++++++++++++++++++---- 2 files changed, 59 insertions(+), 9 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index f018bd779862..9f849e05ce79 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -721,6 +721,10 @@ static const struct config_entry acpi_config_table[] = { #if IS_ENABLED(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) || \ IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) /* BayTrail */ + { + .flags = FLAG_SST_OR_SOF_BYT, + .acpi_hid = "LPE0F28", + }, { .flags = FLAG_SST_OR_SOF_BYT, .acpi_hid = "80860F28", diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 9956dc63db74..f4c4774249ee 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -125,6 +125,28 @@ static const struct sst_res_info bytcr_res_info = { .acpi_ipc_irq_index = 0 }; +/* For "LPE0F28" ACPI device found on some Android factory OS models */ +static const struct sst_res_info lpe8086_res_info = { + .shim_offset = 0x140000, + .shim_size = 0x000100, + .shim_phy_addr = SST_BYT_SHIM_PHY_ADDR, + .ssp0_offset = 0xa0000, + .ssp0_size = 0x1000, + .dma0_offset = 0x98000, + .dma0_size = 0x4000, + .dma1_offset = 0x9c000, + .dma1_size = 0x4000, + .iram_offset = 0x0c0000, + .iram_size = 0x14000, + .dram_offset = 0x100000, + .dram_size = 0x28000, + .mbox_offset = 0x144000, + .mbox_size = 0x1000, + .acpi_lpe_res_index = 1, + .acpi_ddr_index = 0, + .acpi_ipc_irq_index = 0 +}; + static struct sst_platform_info byt_rvp_platform_data = { .probe_data = &byt_fwparse_info, .ipc_info = &byt_ipc_info, @@ -268,10 +290,38 @@ static int sst_acpi_probe(struct platform_device *pdev) mach->pdata = &chv_platform_data; pdata = mach->pdata; - ret = kstrtouint(id->id, 16, &dev_id); - if (ret < 0) { - dev_err(dev, "Unique device id conversion error: %d\n", ret); - return ret; + if (!strcmp(id->id, "LPE0F28")) { + struct resource *rsrc; + + /* Use regular BYT SST PCI VID:PID */ + dev_id = 0x80860F28; + byt_rvp_platform_data.res_info = &lpe8086_res_info; + + /* + * The "LPE0F28" ACPI device has separate IO-mem resources for: + * DDR, SHIM, MBOX, IRAM, DRAM, CFG + * None of which covers the entire LPE base address range. + * lpe8086_res_info.acpi_lpe_res_index points to the SHIM. + * Patch this to cover the entire base address range as expected + * by sst_platform_get_resources(). + */ + rsrc = platform_get_resource(pdev, IORESOURCE_MEM, + pdata->res_info->acpi_lpe_res_index); + if (!rsrc) { + dev_err(ctx->dev, "Invalid SHIM base\n"); + return -EIO; + } + rsrc->start -= pdata->res_info->shim_offset; + rsrc->end = rsrc->start + 0x200000 - 1; + } else { + ret = kstrtouint(id->id, 16, &dev_id); + if (ret < 0) { + dev_err(dev, "Unique device id conversion error: %d\n", ret); + return ret; + } + + if (soc_intel_is_byt_cr(pdev)) + byt_rvp_platform_data.res_info = &bytcr_res_info; } dev_dbg(dev, "ACPI device id: %x\n", dev_id); @@ -280,11 +330,6 @@ static int sst_acpi_probe(struct platform_device *pdev) if (ret < 0) return ret; - if (soc_intel_is_byt_cr(pdev)) { - /* override resource info */ - byt_rvp_platform_data.res_info = &bytcr_res_info; - } - /* update machine parameters */ mach->mach_params.acpi_ipc_irq_index = pdata->res_info->acpi_ipc_irq_index; @@ -344,6 +389,7 @@ static void sst_acpi_remove(struct platform_device *pdev) } static const struct acpi_device_id sst_acpi_ids[] = { + { "LPE0F28", (unsigned long)&snd_soc_acpi_intel_baytrail_machines}, { "80860F28", (unsigned long)&snd_soc_acpi_intel_baytrail_machines}, { "808622A8", (unsigned long)&snd_soc_acpi_intel_cherrytrail_machines}, { }, From d221b844ee79823ffc29b7badc4010bdb0960224 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Sat, 26 Oct 2024 22:46:34 +0200 Subject: [PATCH 17/34] ASoC: cs42l51: Fix some error handling paths in cs42l51_probe() If devm_gpiod_get_optional() fails, we need to disable previously enabled regulators, as done in the other error handling path of the function. Also, gpiod_set_value_cansleep(, 1) needs to be called to undo a potential gpiod_set_value_cansleep(, 0). If the "reset" gpio is not defined, this additional call is just a no-op. This behavior is the same as the one already in the .remove() function. Fixes: 11b9cd748e31 ("ASoC: cs42l51: add reset management") Signed-off-by: Christophe JAILLET Reviewed-by: Charles Keepax Link: https://patch.msgid.link/a5e5f4b9fb03f46abd2c93ed94b5c395972ce0d1.1729975570.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index e4827b8c2bde..6e51954bdb1e 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -747,8 +747,10 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) cs42l51->reset_gpio = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); - if (IS_ERR(cs42l51->reset_gpio)) - return PTR_ERR(cs42l51->reset_gpio); + if (IS_ERR(cs42l51->reset_gpio)) { + ret = PTR_ERR(cs42l51->reset_gpio); + goto error; + } if (cs42l51->reset_gpio) { dev_dbg(dev, "Release reset gpio\n"); @@ -780,6 +782,7 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) return 0; error: + gpiod_set_value_cansleep(cs42l51->reset_gpio, 1); regulator_bulk_disable(ARRAY_SIZE(cs42l51->supplies), cs42l51->supplies); return ret; From c1895ba181e560144601fafe46aeedbafdf4dbc4 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sat, 26 Oct 2024 16:36:15 +0200 Subject: [PATCH 18/34] ASoC: Intel: sst: Fix used of uninitialized ctx to log an error Fix the new "LPE0F28" code path using the uninitialized ctx variable to log an error. Fixes: 6668610b4d8c ("ASoC: Intel: sst: Support LPE0F28 ACPI HID") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202410261106.EBx49ssy-lkp@intel.com/ Signed-off-by: Hans de Goede Link: https://patch.msgid.link/20241026143615.171821-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index f4c4774249ee..257180630475 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -308,7 +308,7 @@ static int sst_acpi_probe(struct platform_device *pdev) rsrc = platform_get_resource(pdev, IORESOURCE_MEM, pdata->res_info->acpi_lpe_res_index); if (!rsrc) { - dev_err(ctx->dev, "Invalid SHIM base\n"); + dev_err(dev, "Invalid SHIM base\n"); return -EIO; } rsrc->start -= pdata->res_info->shim_offset; From 2ef9439f7a19fd3d43b288d38b1c6e55b668a4fe Mon Sep 17 00:00:00 2001 From: Aleksei Vetrov Date: Mon, 28 Oct 2024 22:50:30 +0000 Subject: [PATCH 19/34] ASoC: dapm: fix bounds checker error in dapm_widget_list_create The widgets array in the snd_soc_dapm_widget_list has a __counted_by attribute attached to it, which points to the num_widgets variable. This attribute is used in bounds checking, and if it is not set before the array is filled, then the bounds sanitizer will issue a warning or a kernel panic if CONFIG_UBSAN_TRAP is set. This patch sets the size of the widgets list calculated with list_for_each as the initial value for num_widgets as it is used for allocating memory for the array. It is updated with the actual number of added elements after the array is filled. Signed-off-by: Aleksei Vetrov Fixes: 80e698e2df5b ("ASoC: soc-dapm: Annotate struct snd_soc_dapm_widget_list with __counted_by") Link: https://patch.msgid.link/20241028-soc-dapm-bounds-checker-fix-v1-1-262b0394e89e@google.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c34934c31ffe..99521c784a9b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1147,6 +1147,8 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, if (*list == NULL) return -ENOMEM; + (*list)->num_widgets = size; + list_for_each_entry(w, widgets, work_list) (*list)->widgets[i++] = w; From cc8475a07cf34891bf11a63025659d3537b638ef Mon Sep 17 00:00:00 2001 From: Dmitry Yashin Date: Tue, 29 Oct 2024 02:33:12 +0500 Subject: [PATCH 20/34] ASoC: dt-bindings: rockchip,rk3308-codec: add port property Fix DTB warnings when rk3308-codec used with audio-graph-card by documenting port property: codec@ff560000: 'port' does not match any of the regexes: 'pinctrl-[0-9]+' Signed-off-by: Dmitry Yashin Reviewed-by: Luca Ceresoli Link: https://patch.msgid.link/20241028213314.476776-2-dmt.yashin@gmail.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/rockchip,rk3308-codec.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml index ecf3d7d968c8..2cf229a076f0 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml @@ -48,6 +48,10 @@ properties: - const: mclk_rx - const: hclk + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + resets: maxItems: 1 From 041db4bbe04e8e0b48350b3bbbd9a799794d5c1e Mon Sep 17 00:00:00 2001 From: Alexey Klimov Date: Tue, 22 Oct 2024 04:31:30 +0100 Subject: [PATCH 21/34] ASoC: codecs: wcd937x: add missing LO Switch control The wcd937x supports also AUX input but the control that sets correct soundwire port for this is missing. This control is required for audio playback, for instance, on qrb4210 RB2 board as well as on other SoCs. Reported-by: Adam Skladowski Reported-by: Prasad Kumpatla Suggested-by: Adam Skladowski Suggested-by: Prasad Kumpatla Cc: Srinivas Kandagatla Cc: Mohammad Rafi Shaik Signed-off-by: Alexey Klimov Link: https://patch.msgid.link/20241022033132.787416-2-alexey.klimov@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd937x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index 45f32d281908..0f0d2537d322 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -2049,6 +2049,8 @@ static const struct snd_kcontrol_new wcd937x_snd_controls[] = { wcd937x_get_swr_port, wcd937x_set_swr_port), SOC_SINGLE_EXT("HPHR Switch", WCD937X_HPH_R, 0, 1, 0, wcd937x_get_swr_port, wcd937x_set_swr_port), + SOC_SINGLE_EXT("LO Switch", WCD937X_LO, 0, 1, 0, + wcd937x_get_swr_port, wcd937x_set_swr_port), SOC_SINGLE_EXT("ADC1 Switch", WCD937X_ADC1, 1, 1, 0, wcd937x_get_swr_port, wcd937x_set_swr_port), From 107a5c853eef5336a9846e7dd2f9184b6e3c07c7 Mon Sep 17 00:00:00 2001 From: Alexey Klimov Date: Tue, 22 Oct 2024 04:31:31 +0100 Subject: [PATCH 22/34] ASoC: codecs: wcd937x: relax the AUX PDM watchdog On a system with wcd937x, rxmacro and Qualcomm audio DSP, which is pretty common set of devices on Qualcomm platforms, and due to the order of how DAPM widgets are powered on (they are sorted), there is a small time window when wcd937x chip is online and expects the flow of incoming data but rxmacro is not yet online. When wcd937x is programmed to receive data via AUX port then its AUX PDM watchdog is enabled in wcd937x_codec_enable_aux_pa(). If due to some reasons the rxmacro and soundwire machinery are delayed to start streaming data, then there is a chance for this AUX PDM watchdog to reset the wcd937x codec. Such event is not logged as a message and only wcd937x IRQ counter is increased however there could be a lot of other reasons for that IRQ. There is a similar opportunity for such delay during DAPM widgets power down sequence. If wcd937x codec reset happens on the start of the playback, then there will be no sound and if such reset happens at the end of a playback then it may generate additional clicks and pops noises. On qrb4210 RB2 board without any debugging bits the wcd937x resets are sometimes observed at the end of a playback though not always. With some debugging messages or with some tracing enabled the AUX PDM watchdog resets the wcd937x codec at the start of a playback and there is no sound output at all. In this patch: - TIMEOUT_SEL bit in PDM_WD_CTL2 register is set to increase the watchdog reset delay to 100ms which eliminates the AUX PDM watchdog IRQs on qrb4210 RB2 board completely and decreases the number of unwanted clicks noises; - HOLD_OFF bit postpones triggering such watchdog IRQ till wcd937x codec reset which usually happens at the end of a playback. This allows to actually output some sound in case of debugging. Cc: Adam Skladowski Cc: Mohammad Rafi Shaik Cc: Prasad Kumpatla Cc: Srinivas Kandagatla Signed-off-by: Alexey Klimov Link: https://patch.msgid.link/20241022033132.787416-3-alexey.klimov@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd937x.c | 10 ++++++++-- sound/soc/codecs/wcd937x.h | 4 ++++ 2 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index 0f0d2537d322..08fb13a334a4 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -715,12 +715,17 @@ static int wcd937x_codec_enable_aux_pa(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct wcd937x_priv *wcd937x = snd_soc_component_get_drvdata(component); int hph_mode = wcd937x->hph_mode; + u8 val; switch (event) { case SND_SOC_DAPM_PRE_PMU: + val = WCD937X_DIGITAL_PDM_WD_CTL2_EN | + WCD937X_DIGITAL_PDM_WD_CTL2_TIMEOUT_SEL | + WCD937X_DIGITAL_PDM_WD_CTL2_HOLD_OFF; snd_soc_component_update_bits(component, WCD937X_DIGITAL_PDM_WD_CTL2, - BIT(0), BIT(0)); + WCD937X_DIGITAL_PDM_WD_CTL2_MASK, + val); break; case SND_SOC_DAPM_POST_PMU: usleep_range(1000, 1010); @@ -741,7 +746,8 @@ static int wcd937x_codec_enable_aux_pa(struct snd_soc_dapm_widget *w, hph_mode); snd_soc_component_update_bits(component, WCD937X_DIGITAL_PDM_WD_CTL2, - BIT(0), 0x00); + WCD937X_DIGITAL_PDM_WD_CTL2_MASK, + 0x00); break; } diff --git a/sound/soc/codecs/wcd937x.h b/sound/soc/codecs/wcd937x.h index 35f3d48bd7dd..4afa48dcaf74 100644 --- a/sound/soc/codecs/wcd937x.h +++ b/sound/soc/codecs/wcd937x.h @@ -391,6 +391,10 @@ #define WCD937X_DIGITAL_PDM_WD_CTL0 0x3465 #define WCD937X_DIGITAL_PDM_WD_CTL1 0x3466 #define WCD937X_DIGITAL_PDM_WD_CTL2 0x3467 +#define WCD937X_DIGITAL_PDM_WD_CTL2_HOLD_OFF BIT(2) +#define WCD937X_DIGITAL_PDM_WD_CTL2_TIMEOUT_SEL BIT(1) +#define WCD937X_DIGITAL_PDM_WD_CTL2_EN BIT(0) +#define WCD937X_DIGITAL_PDM_WD_CTL2_MASK GENMASK(2, 0) #define WCD937X_DIGITAL_INTR_MODE 0x346A #define WCD937X_DIGITAL_INTR_MASK_0 0x346B #define WCD937X_DIGITAL_INTR_MASK_1 0x346C From fe09de2db2365eed8b44b572cff7d421eaf1754a Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Mon, 4 Nov 2024 18:00:55 +0800 Subject: [PATCH 23/34] ASoC: tas2781: Add new driver version for tas2563 & tas2781 qfn chip Add new driver version to support tas2563 & tas2781 qfn chip Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20241104100055.48-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-fmwlib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index ae360c97fe1e..0aeb88abbf52 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1992,6 +1992,7 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, break; case 0x202: case 0x400: + case 0x401: tas_priv->fw_parse_variable_header = fw_parse_variable_header_git; tas_priv->fw_parse_program_data = From 08a3b241adfd90361c16c3e92f5275b816a73f04 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 5 Nov 2024 01:00:00 +0000 Subject: [PATCH 24/34] MAINTAINERS: Generic Sound Card section ALSA SoC Sound has Generic Sound Card (Simple-Card, Audio-Graph-Card, Audio-Graph-Card2). Adds its Maintainer. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87ikt2a41c.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- MAINTAINERS | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 9d6272c00fbd..388626c3df1f 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -21697,6 +21697,15 @@ S: Supported W: https://github.com/thesofproject/linux/ F: sound/soc/sof/ +SOUND - GENERIC SOUND CARD (Simple-Audio-Card, Audio-Graph-Card) +M: Kuninori Morimoto +S: Supported +L: linux-sound@vger.kernel.org +F: sound/soc/generic/ +F: include/sound/simple_card* +F: Documentation/devicetree/bindings/sound/simple-card.yaml +F: Documentation/devicetree/bindings/sound/audio-graph*.yaml + SOUNDWIRE SUBSYSTEM M: Vinod Koul M: Bard Liao From 9bb4af400c386374ab1047df44c508512c08c31f Mon Sep 17 00:00:00 2001 From: Amelie Delaunay Date: Tue, 5 Nov 2024 15:02:42 +0100 Subject: [PATCH 25/34] ASoC: stm32: spdifrx: fix dma channel release in stm32_spdifrx_remove In case of error when requesting ctrl_chan DMA channel, ctrl_chan is not null. So the release of the dma channel leads to the following issue: [ 4.879000] st,stm32-spdifrx 500d0000.audio-controller: dma_request_slave_channel error -19 [ 4.888975] Unable to handle kernel NULL pointer dereference at virtual address 000000000000003d [...] [ 5.096577] Call trace: [ 5.099099] dma_release_channel+0x24/0x100 [ 5.103235] stm32_spdifrx_remove+0x24/0x60 [snd_soc_stm32_spdifrx] [ 5.109494] stm32_spdifrx_probe+0x320/0x4c4 [snd_soc_stm32_spdifrx] To avoid this issue, release channel only if the pointer is valid. Fixes: 794df9448edb ("ASoC: stm32: spdifrx: manage rebind issue") Signed-off-by: Amelie Delaunay Signed-off-by: Olivier Moysan Link: https://patch.msgid.link/20241105140242.527279-1-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_spdifrx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index d1b32ba1e1a2..9e30852de93c 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -939,7 +939,7 @@ static void stm32_spdifrx_remove(struct platform_device *pdev) { struct stm32_spdifrx_data *spdifrx = platform_get_drvdata(pdev); - if (spdifrx->ctrl_chan) + if (!IS_ERR(spdifrx->ctrl_chan)) dma_release_channel(spdifrx->ctrl_chan); if (spdifrx->dmab) From de156f3cf70e17dc6ff4c3c364bb97a6db961ffd Mon Sep 17 00:00:00 2001 From: Mingcong Bai Date: Wed, 6 Nov 2024 10:40:50 +0800 Subject: [PATCH 26/34] ASoC: amd: yc: fix internal mic on Xiaomi Book Pro 14 2022 Xiaomi Book Pro 14 2022 (MIA2210-AD) requires a quirk entry for its internal microphone to be enabled. This is likely due to similar reasons as seen previously on Redmi Book 14/15 Pro 2022 models (since they likely came with similar firmware): - commit dcff8b7ca92d ("ASoC: amd: yc: Add Xiaomi Redmi Book Pro 15 2022 into DMI table") - commit c1dd6bf61997 ("ASoC: amd: yc: Add Xiaomi Redmi Book Pro 14 2022 into DMI table") A quirk would likely be needed for Xiaomi Book Pro 15 2022 models, too. However, I do not have such device on hand so I will leave it for now. Signed-off-by: Mingcong Bai Link: https://patch.msgid.link/20241106024052.15748-1-jeffbai@aosc.io 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 438865d5e376..dc476bfb6da4 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -395,6 +395,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Redmi Book Pro 15 2022"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "TIMI"), + DMI_MATCH(DMI_PRODUCT_NAME, "Xiaomi Book Pro 14 2022"), + } + }, { .driver_data = &acp6x_card, .matches = { From 94debe5eaa0adaa24a6de4a8e5f138be7381eb9e Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 6 Nov 2024 19:56:57 +0530 Subject: [PATCH 27/34] ASoC: SOF: amd: Fix for incorrect DMA ch status register offset DMA ch status register offset change in acp7.0 platform Incorrect DMA channel status register offset check lead to firmware boot failure. [ 14.432497] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump start ]------------ [ 14.432533] snd_sof_amd_acp70 0000:c4:00.5: Firmware boot failure due to timeout [ 14.432549] snd_sof_amd_acp70 0000:c4:00.5: fw_state: SOF_FW_BOOT_IN_PROGRESS (3) [ 14.432610] snd_sof_amd_acp70 0000:c4:00.5: invalid header size 0x71c41000. FW oops is bogus [ 14.432626] snd_sof_amd_acp70 0000:c4:00.5: unexpected fault 0x71c40000 trace 0x71c40000 [ 14.432642] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump end ]------------ [ 14.432657] snd_sof_amd_acp70 0000:c4:00.5: error: failed to boot DSP firmware -5 [ 14.432672] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 3 -> 4 [ 14.433260] dmic-codec dmic-codec: ASoC: Unregistered DAI 'dmic-hifi' [ 14.433319] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 4 -> 0 [ 14.433358] snd_sof_amd_acp70 0000:c4:00.5: error: sof_probe_work failed err: -5 Update correct register offset for DMA ch status register. Fixes: 490be7ba2a01 ("ASoC: SOF: amd: add support for acp7.0 based platform") Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20241106142658.1240929-1-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index de3001f5b9bb..95d4762c9d93 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -342,11 +342,19 @@ int acp_dma_status(struct acp_dev_data *adata, unsigned char ch) { struct snd_sof_dev *sdev = adata->dev; unsigned int val; + unsigned int acp_dma_ch_sts; int ret = 0; + switch (adata->pci_rev) { + case ACP70_PCI_ID: + acp_dma_ch_sts = ACP70_DMA_CH_STS; + break; + default: + acp_dma_ch_sts = ACP_DMA_CH_STS; + } val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_DMA_CNTL_0 + ch * sizeof(u32)); if (val & ACP_DMA_CH_RUN) { - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_DMA_CH_STS, val, !val, + ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, acp_dma_ch_sts, val, !val, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); if (ret < 0) From 8c21e40e1e481f7fef6e570089e317068b972c45 Mon Sep 17 00:00:00 2001 From: Markus Petri Date: Thu, 7 Nov 2024 10:40:20 +0100 Subject: [PATCH 28/34] ASoC: amd: yc: Support dmic on another model of Lenovo Thinkpad E14 Gen 6 Another model of Thinkpad E14 Gen 6 (21M4) needs a quirk entry for the dmic to be detected. Signed-off-by: Markus Petri Link: https://patch.msgid.link/20241107094020.1050935-1-mp@localhost 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 dc476bfb6da4..2436e8deb2be 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -227,6 +227,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21M3"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21M4"), + } + }, { .driver_data = &acp6x_card, .matches = { From 63c1c87993e0e5bb11bced3d8224446a2bc62338 Mon Sep 17 00:00:00 2001 From: Luo Yifan Date: Wed, 6 Nov 2024 09:46:54 +0800 Subject: [PATCH 29/34] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate() This patch checks if div is less than or equal to zero (div <= 0). If div is zero or negative, the function returns -EINVAL, ensuring the division operation (*prate / div) is safe to perform. Signed-off-by: Luo Yifan Link: https://patch.msgid.link/20241106014654.206860-1-luoyifan@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 7bc4a96b7503..2570daa3e225 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -378,8 +378,8 @@ static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, int div; div = stm32_sai_get_clk_div(sai, *prate, rate); - if (div < 0) - return div; + if (div <= 0) + return -EINVAL; mclk->freq = *prate / div; From 23569c8b314925bdb70dd1a7b63cfe6100868315 Mon Sep 17 00:00:00 2001 From: Luo Yifan Date: Thu, 7 Nov 2024 09:59:36 +0800 Subject: [PATCH 30/34] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div() This patch checks if div is less than or equal to zero (div <= 0). If div is zero or negative, the function returns -EINVAL, ensuring the division operation is safe to perform. Signed-off-by: Luo Yifan Reviewed-by: Olivier Moysan Link: https://patch.msgid.link/20241107015936.211902-1-luoyifan@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 2570daa3e225..5828f9dd866e 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -317,7 +317,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } From 48b86532c10128cf50c854a90c2d5b1410f4012d Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 7 Nov 2024 15:28:40 +0200 Subject: [PATCH 31/34] ASoC: SOF: sof-client-probes-ipc4: Set param_size extension bits Write the size of the optional payload of SOF_IPC4_MOD_INIT_INSTANCE message to extension param_size-bits. The previous IPC4 version does not set these bits that should indicate the size of the optional payload (struct sof_ipc4_probe_cfg). The old firmware side component code works well without these bits, but when the probes are converted to use the generic module API, this does not work anymore. Fixes: f5623593060f ("ASoC: SOF: IPC4: probes: Implement IPC4 ops for probes client device") Signed-off-by: Jyri Sarha Reviewed-by: Ranjani Sridharan Reviewed-by: Liam Girdwood Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://patch.msgid.link/20241107132840.17386-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-client-probes-ipc4.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/sof-client-probes-ipc4.c b/sound/soc/sof/sof-client-probes-ipc4.c index 796eac0a2e74..603aed222480 100644 --- a/sound/soc/sof/sof-client-probes-ipc4.c +++ b/sound/soc/sof/sof-client-probes-ipc4.c @@ -125,6 +125,7 @@ static int ipc4_probes_init(struct sof_client_dev *cdev, u32 stream_tag, msg.primary |= SOF_IPC4_MSG_TARGET(SOF_IPC4_MODULE_MSG); msg.extension = SOF_IPC4_MOD_EXT_DST_MOD_INSTANCE(INVALID_PIPELINE_ID); msg.extension |= SOF_IPC4_MOD_EXT_CORE_ID(0); + msg.extension |= SOF_IPC4_MOD_EXT_PARAM_SIZE(sizeof(cfg) / sizeof(uint32_t)); msg.data_size = sizeof(cfg); msg.data_ptr = &cfg; From f8da001ae7af0abd9f6250c02c01a1121074ca60 Mon Sep 17 00:00:00 2001 From: John Watts Date: Fri, 8 Nov 2024 12:37:15 +1100 Subject: [PATCH 32/34] ASoC: audio-graph-card2: Purge absent supplies for device tree nodes The audio graph card doesn't mark its subnodes such as multi {}, dpcm {} and c2c {} as not requiring any suppliers. This causes a hang as Linux waits for these phantom suppliers to show up on boot. Make it clear these nodes have no suppliers. Example error message: [ 15.208558] platform 2034000.i2s: deferred probe pending: platform: wait for supplier /sound/multi [ 15.208584] platform sound: deferred probe pending: asoc-audio-graph-card2: parse error Signed-off-by: John Watts Acked-by: Kuninori Morimoto Link: https://patch.msgid.link/20241108-graph_dt_fix-v1-1-173e2f9603d6@jookia.org Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 4ad3d1b0714f..93eee40cec76 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -270,16 +270,19 @@ static enum graph_type __graph_get_type(struct device_node *lnk) if (of_node_name_eq(np, GRAPH_NODENAME_MULTI)) { ret = GRAPH_MULTI; + fw_devlink_purge_absent_suppliers(&np->fwnode); goto out_put; } if (of_node_name_eq(np, GRAPH_NODENAME_DPCM)) { ret = GRAPH_DPCM; + fw_devlink_purge_absent_suppliers(&np->fwnode); goto out_put; } if (of_node_name_eq(np, GRAPH_NODENAME_C2C)) { ret = GRAPH_C2C; + fw_devlink_purge_absent_suppliers(&np->fwnode); goto out_put; } From d859923faeca740ae9235e2b9328999836e681b9 Mon Sep 17 00:00:00 2001 From: Deep Harsora Date: Mon, 11 Nov 2024 15:06:18 +0800 Subject: [PATCH 33/34] ASoC: intel: sof_sdw: add quirk for Dell SKU MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch adds a quirk to include the codec amplifier function for this Dell SKU. Note: In this SKU '0CF1', the RT722 codec amplifier is excluded, and an external amplifier is used instead. Signed-off-by: Deep Harsora Reviewed-by: Liam Girdwood Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Link: https://patch.msgid.link/20241111070618.5414-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 35d707d3ae9c..4a0ab50d1e50 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -590,6 +590,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOC_SDW_CODEC_SPKR), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF1") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, { .callback = sof_sdw_quirk_cb, .matches = { From 2ae6da569e34e1d26c5275442d17ffd75fd343b3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Nov 2024 13:09:50 +0000 Subject: [PATCH 34/34] ASoC: max9768: Fix event generation for playback mute The max9768 has a custom control for playback mute which unconditionally returns 0 from the put() operation, rather than returning 1 on change to ensure notifications are generated to userspace. Check to see if the value has changed and return appropriately. Signed-off-by: Mark Brown Link: https://patch.msgid.link/20241112-asoc-max9768-event-v1-1-ba5d50599787@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/max9768.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index e4793a5d179e..8af3c7e5317f 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -54,10 +54,17 @@ static int max9768_set_gpio(struct snd_kcontrol *kcontrol, { struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); struct max9768 *max9768 = snd_soc_component_get_drvdata(c); + bool val = !ucontrol->value.integer.value[0]; + int ret; - gpiod_set_value_cansleep(max9768->mute, !ucontrol->value.integer.value[0]); + if (val != gpiod_get_value_cansleep(max9768->mute)) + ret = 1; + else + ret = 0; - return 0; + gpiod_set_value_cansleep(max9768->mute, val); + + return ret; } static const DECLARE_TLV_DB_RANGE(volume_tlv,