November 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: SOF: ipc4: Move window offset configuration earlier
by Peter Ujfalusi 29 Nov '23

29 Nov '23

With the added exception handling support if the firmware fails to boot up we are trying to do a panic dump from the telemetry slot. The slot offsets would have been configured only after receiving the FW_READY message which makes this panic dump unusable for early boot failures. With IPC4 the DSP window offsets are at standard places unlike IPC3 where the offsets needs to be queried from the FW_READY message. Move the offset configuration to sof_ipc4_init from the fw_ready handler. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc4.c | 55 ++++++++++++++++++++++---------------------- 1 file changed, 27 insertions(+), 28 deletions(-) diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index 145d319e041f..ac5c6bc66d2a 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -576,40 +576,12 @@ EXPORT_SYMBOL(sof_ipc4_find_debug_slot_offset_by_type); static int ipc4_fw_ready(struct snd_sof_dev *sdev, struct sof_ipc4_msg *ipc4_msg) { - int inbox_offset, inbox_size, outbox_offset, outbox_size; - /* no need to re-check version/ABI for subsequent boots */ if (!sdev->first_boot) return 0; - /* Set up the windows for IPC communication */ - inbox_offset = snd_sof_dsp_get_mailbox_offset(sdev); - if (inbox_offset < 0) { - dev_err(sdev->dev, "%s: No mailbox offset\n", __func__); - return inbox_offset; - } - inbox_size = SOF_IPC4_MSG_MAX_SIZE; - outbox_offset = snd_sof_dsp_get_window_offset(sdev, SOF_IPC4_OUTBOX_WINDOW_IDX); - outbox_size = SOF_IPC4_MSG_MAX_SIZE; - - sdev->fw_info_box.offset = snd_sof_dsp_get_window_offset(sdev, SOF_IPC4_INBOX_WINDOW_IDX); - sdev->fw_info_box.size = sizeof(struct sof_ipc4_fw_registers); - sdev->dsp_box.offset = inbox_offset; - sdev->dsp_box.size = inbox_size; - sdev->host_box.offset = outbox_offset; - sdev->host_box.size = outbox_size; - - sdev->debug_box.offset = snd_sof_dsp_get_window_offset(sdev, - SOF_IPC4_DEBUG_WINDOW_IDX); - sof_ipc4_create_exception_debugfs_node(sdev); - dev_dbg(sdev->dev, "mailbox upstream 0x%x - size 0x%x\n", - inbox_offset, inbox_size); - dev_dbg(sdev->dev, "mailbox downstream 0x%x - size 0x%x\n", - outbox_offset, outbox_size); - dev_dbg(sdev->dev, "debug box 0x%x\n", sdev->debug_box.offset); - return sof_ipc4_init_msg_memory(sdev); } @@ -796,11 +768,38 @@ static const struct sof_ipc_pm_ops ipc4_pm_ops = { static int sof_ipc4_init(struct snd_sof_dev *sdev) { struct sof_ipc4_fw_data *ipc4_data = sdev->private; + int inbox_offset; mutex_init(&ipc4_data->pipeline_state_mutex); xa_init_flags(&ipc4_data->fw_lib_xa, XA_FLAGS_ALLOC); + /* Set up the windows for IPC communication */ + inbox_offset = snd_sof_dsp_get_mailbox_offset(sdev); + if (inbox_offset < 0) { + dev_err(sdev->dev, "%s: No mailbox offset\n", __func__); + return inbox_offset; + } + + sdev->dsp_box.offset = inbox_offset; + sdev->dsp_box.size = SOF_IPC4_MSG_MAX_SIZE; + sdev->host_box.offset = snd_sof_dsp_get_window_offset(sdev, + SOF_IPC4_OUTBOX_WINDOW_IDX); + sdev->host_box.size = SOF_IPC4_MSG_MAX_SIZE; + + sdev->debug_box.offset = snd_sof_dsp_get_window_offset(sdev, + SOF_IPC4_DEBUG_WINDOW_IDX); + + sdev->fw_info_box.offset = snd_sof_dsp_get_window_offset(sdev, + SOF_IPC4_INBOX_WINDOW_IDX); + sdev->fw_info_box.size = sizeof(struct sof_ipc4_fw_registers); + + dev_dbg(sdev->dev, "mailbox upstream %#x - size %#x\n", + sdev->dsp_box.offset, SOF_IPC4_MSG_MAX_SIZE); + dev_dbg(sdev->dev, "mailbox downstream %#x - size %#x\n", + sdev->host_box.offset, SOF_IPC4_MSG_MAX_SIZE); + dev_dbg(sdev->dev, "debug box %#x\n", sdev->debug_box.offset); + return 0; } -- 2.43.0

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[PATCH] ASoC: SOF: ipc4-topology: Add module ID print during module set up
by Peter Ujfalusi 29 Nov '23

29 Nov '23

From: Baofeng Tian <baofeng.tian(a)intel.com> This module ID will be used for module performance automatic analysis for different modules, module name, module ID and module instance ID will be combined as a new generated ID for current module, this ID will be further used by analysis tools to identify current module. Take below case as example: 0x030006 gain.11.1 3 is module instance ID, 6 is module ID and gain.11.1 is module name. For pipeline widget print, keep as it is. Signed-off-by: Baofeng Tian <baofeng.tian(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index b89065abb511..01f3f56495cd 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -2392,6 +2392,8 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget } if (swidget->id != snd_soc_dapm_scheduler) { + int module_id = msg->primary & SOF_IPC4_MOD_ID_MASK; + ret = sof_ipc4_widget_assign_instance_id(sdev, swidget); if (ret < 0) { dev_err(sdev->dev, "failed to assign instance id for %s\n", @@ -2407,9 +2409,15 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget msg->extension &= ~SOF_IPC4_MOD_EXT_PPL_ID_MASK; msg->extension |= SOF_IPC4_MOD_EXT_PPL_ID(pipe_widget->instance_id); + + dev_dbg(sdev->dev, "Create widget %s (pipe %d) - ID %d, instance %d, core %d\n", + swidget->widget->name, swidget->pipeline_id, module_id, + swidget->instance_id, swidget->core); + } else { + dev_dbg(sdev->dev, "Create pipeline %s (pipe %d) - instance %d, core %d\n", + swidget->widget->name, swidget->pipeline_id, + swidget->instance_id, swidget->core); } - dev_dbg(sdev->dev, "Create widget %s instance %d - pipe %d - core %d\n", - swidget->widget->name, swidget->instance_id, swidget->pipeline_id, swidget->core); msg->data_size = ipc_size; msg->data_ptr = ipc_data; -- 2.43.0

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[PATCH] ASoC: SOF: ipc4: check return value of snd_sof_ipc_msg_data
by Peter Ujfalusi 29 Nov '23

29 Nov '23

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> snd_sof_ipc_msg_data could return error. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc4.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index a9d9800d2fcc..145d319e041f 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -713,7 +713,14 @@ static void sof_ipc4_rx_msg(struct snd_sof_dev *sdev) return; ipc4_msg->data_size = data_size; - snd_sof_ipc_msg_data(sdev, NULL, ipc4_msg->data_ptr, ipc4_msg->data_size); + err = snd_sof_ipc_msg_data(sdev, NULL, ipc4_msg->data_ptr, ipc4_msg->data_size); + if (err < 0) { + dev_err(sdev->dev, "failed to read IPC notification data: %d\n", err); + kfree(ipc4_msg->data_ptr); + ipc4_msg->data_ptr = NULL; + ipc4_msg->data_size = 0; + return; + } } /* Handle notifications with payload */ -- 2.43.0

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[PATCH 0/5] ASoC: codecs: add support for WCD939x Codec
by Neil Armstrong 29 Nov '23

29 Nov '23

Add the main WCD9390/WCD9395 Audio Codec driver to support: - 4 ADC inputs for up to 5 Analog Microphones - 4 DMIC inputs for up to 8 Digital Microphones - 4 Microphone BIAS - Stereo Headphone output - Mono EAR output - MBHC engine for Headset Detection This adds: - bindings - MBHC changes to support Type-C muc - Soundwire Slave driver - Code driver The USB Mux subsystem support will be submitted separalety since it's a functionally separate subsystem connected over I2C with it's own reset signal. Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- Neil Armstrong (5): ASoC: dt-bindings: qcom,wcd938x: move out common properties ASoC: dt-bindings: document WCD939x Audio Codec ASoC: codec: wcd-mbhc-v2: add support when connected behind an USB-C audio mux ASoC: codecs: Add WCD939x Soundwire slave driver ASoC: codecs: Add WCD939x Codec driver .../devicetree/bindings/sound/qcom,wcd938x.yaml | 81 +- .../bindings/sound/qcom,wcd939x-sdw.yaml | 70 + .../devicetree/bindings/sound/qcom,wcd939x.yaml | 93 + .../bindings/sound/qcom,wcd93xx-common.yaml | 95 + sound/soc/codecs/Kconfig | 18 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/wcd-clsh-v2.h | 1 + sound/soc/codecs/wcd-mbhc-v2.c | 95 +- sound/soc/codecs/wcd-mbhc-v2.h | 3 + sound/soc/codecs/wcd939x-sdw.c | 1591 +++++++++ sound/soc/codecs/wcd939x.c | 3635 ++++++++++++++++++++ sound/soc/codecs/wcd939x.h | 936 +++++ 12 files changed, 6530 insertions(+), 95 deletions(-) --- base-commit: 07b677953b9dca02928be323e2db853511305fa9 change-id: 20231123-topic-sm8650-upstream-wcd939x-codec-c46f621b55d4 Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

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[PATCH] soundwire: qcom: allow multi-link on newer devices
by Krzysztof Kozlowski 29 Nov '23

29 Nov '23

Newer Qualcomm SoCs like X1E80100 might come with four speakers spread over two Soundwire controllers, thus they need a multi-link Soundwire stream runtime. Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- This is an entirely different approach than my previous try here: https://lore.kernel.org/all/20231025144601.268645-1-krzysztof.kozlowski@lin… --- drivers/soundwire/qcom.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 412b8e663a0a..57943724f0eb 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -920,6 +920,18 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) return 0; } +static int qcom_swrm_read_prop(struct sdw_bus *bus) +{ + struct qcom_swrm_ctrl *ctrl = to_qcom_sdw(bus); + + if (ctrl->version >= SWRM_VERSION_2_0_0) { + bus->multi_link = true; + bus->hw_sync_min_links = 3; + } + + return 0; +} + static enum sdw_command_response qcom_swrm_xfer_msg(struct sdw_bus *bus, struct sdw_msg *msg) { @@ -1078,6 +1090,7 @@ static const struct sdw_master_port_ops qcom_swrm_port_ops = { }; static const struct sdw_master_ops qcom_swrm_ops = { + .read_prop = qcom_swrm_read_prop, .xfer_msg = qcom_swrm_xfer_msg, .pre_bank_switch = qcom_swrm_pre_bank_switch, .post_bank_switch = qcom_swrm_post_bank_switch, @@ -1196,6 +1209,15 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, mutex_lock(&ctrl->port_lock); list_for_each_entry(m_rt, &stream->master_list, stream_node) { + /* + * For streams with multiple masters: + * Allocate ports only for devices connected to this master. + * Such devices will have ports allocated by their own master + * and its qcom_swrm_stream_alloc_ports() call. + */ + if (ctrl->bus.id != m_rt->bus->id) + continue; + if (m_rt->direction == SDW_DATA_DIR_RX) { maxport = ctrl->num_dout_ports; port_mask = &ctrl->dout_port_mask; -- 2.34.1

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Add ALC257 support for ChromeOS
by Kailang 29 Nov '23

29 Nov '23

Hi Takashi, Chromebook want to support ALC257. BR, Kailang

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[PATCH] ASoC: SOF: icp3-dtrace: Fix wrong kfree() usage
by Kamil Duljas 28 Nov '23

28 Nov '23

trace_filter_parse() allocs memory for *out and when -ENOMEM is returned, caller function, dfsentry_trace_filter_write() trying to freed this memory. After this patch, the memory is freed in trace_filter_parse() before -EINVAL returned. In caller function removed kfree(elms) from error label Signed-off-by: Kamil Duljas <kamil.duljas(a)gmail.com> --- sound/soc/sof/ipc3-dtrace.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index 0dca139322f3..93b189c2d2ee 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -137,6 +137,7 @@ static int trace_filter_parse(struct snd_sof_dev *sdev, char *string, dev_err(sdev->dev, "Parsing filter entry '%s' failed with %d\n", entry, entry_len); + kfree(*out); return -EINVAL; } } @@ -208,13 +209,13 @@ static ssize_t dfsentry_trace_filter_write(struct file *file, const char __user ret = ipc3_trace_update_filter(sdev, num_elems, elems); if (ret < 0) { dev_err(sdev->dev, "Filter update failed: %d\n", ret); + kfree(elems); goto error; } } ret = count; error: kfree(string); - kfree(elems); return ret; } -- 2.42.0.windows.2

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[PATCH] ASoC: codecs: lpass-tx-macro: set active_decimator correct default value
by Neil Armstrong 28 Nov '23

28 Nov '23

The -1 value for active_decimator[dai_id] is considered as "not set", but at probe the table is initialized a 0, this prevents enabling the DEC0 Mixer since it will be considered as already set. Initialize the table entries as -1 to fix tx_macro_tx_mixer_put(). Fixes: 1c6a7f5250ce ("ASoC: codecs: tx-macro: fix active_decimator array") Fixes: c1057a08af43 ("ASoC: codecs: tx-macro: fix kcontrol put") Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- sound/soc/codecs/lpass-tx-macro.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 82f9873ffada..124c2e144f33 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -2021,6 +2021,11 @@ static int tx_macro_probe(struct platform_device *pdev) tx->dev = dev; + /* Set active_decimator default value */ + tx->active_decimator[TX_MACRO_AIF1_CAP] = -1; + tx->active_decimator[TX_MACRO_AIF2_CAP] = -1; + tx->active_decimator[TX_MACRO_AIF3_CAP] = -1; + /* set MCLK and NPL rates */ clk_set_rate(tx->mclk, MCLK_FREQ); clk_set_rate(tx->npl, MCLK_FREQ); --- base-commit: 5ba73bec5e7b0494da7fdca3e003d8b97fa932cd change-id: 20231116-topic-sm8x50-upstream-tx-macro-fix-active-decimator-set-93d363bfb369 Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

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[RESEND] ASoC: amd: yc: Add DMI entry to support System76 Pangolin 13
by Tim Crawford 28 Nov '23

28 Nov '23

From: Jeremy Soller <jeremy(a)system76.com> Add pang13 quirk to enable the internal microphone. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- 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 3ec15b46fa35..fd5b7d6c6eec 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -353,6 +353,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "System76"), + DMI_MATCH(DMI_PRODUCT_VERSION, "pang13"), + } + }, {} }; -- 2.41.0

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tlv320aic3x: Failed to init class D
by Marc Kleine-Budde 28 Nov '23

28 Nov '23

Hello, we are using an I2C connected ti,tlv320aic3007. During probe, regmap_register_patch() returns -6, which leads to the error message: "tlv320aic3x 2-0018: Failed to init class D: -6". I think this is because the codec is still in reset and does not respond to the I2C transfers. How to get rid of the error message? Power on the codec, just to write the register sequence or don't print the error message on "-6"? regards, Marc -- Pengutronix e.K. | Marc Kleine-Budde | Embedded Linux | https://www.pengutronix.de | Vertretung Nürnberg | Phone: +49-5121-206917-129 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-9 |

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[PATCH v2] ASoC: hdac_hda: Conditionally register dais for HDMI and Analog
by Peter Ujfalusi 28 Nov '23

28 Nov '23

The current driver is registering the same dais for each hdev found in the system which results duplicated widgets to be registered and the kernel log contains similar prints: snd_hda_codec_realtek ehdaudio0D0: ASoC: sink widget AIF1TX overwritten snd_hda_codec_realtek ehdaudio0D0: ASoC: source widget AIF1RX overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi3 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi2 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Codec Output Pin1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Codec Input Pin1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Analog Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Digital Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Alt Analog Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Analog Codec Capture overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Digital Codec Capture overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Alt Analog Codec Capture overwritten To avoid such issue, split the dai array into HDMI and non HDMI array and register them conditionally: for HDMI hdev only register the dais needed for HDMI for non HDMI hdev do not register the HDMI dais. Depends-on: 3d1dc8b1030d ("ASoC: Intel: skl_hda_dsp_generic: Drop HDMI routes when HDMI is not available") Link: https://github.com/thesofproject/linux/issues/4509 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, Changes since v1: - Drop the patch for patch_hdmi to export a funtion to match the device - Use the struct hdac_hda_priv.need_display_power boolean as indication that the device is a HDMI/DP audio codec Regards, Peter sound/soc/codecs/hdac_hda.c | 23 ++++++++++++++++++++--- 1 file changed, 20 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 355f30779a34..b075689db2dc 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -132,6 +132,9 @@ static struct snd_soc_dai_driver hdac_hda_dais[] = { .sig_bits = 24, }, }, +}; + +static struct snd_soc_dai_driver hdac_hda_hdmi_dais[] = { { .id = HDAC_HDMI_0_DAI_ID, .name = "intel-hdmi-hifi1", @@ -607,8 +610,16 @@ static const struct snd_soc_component_driver hdac_hda_codec = { .endianness = 1, }; +static const struct snd_soc_component_driver hdac_hda_hdmi_codec = { + .probe = hdac_hda_codec_probe, + .remove = hdac_hda_codec_remove, + .idle_bias_on = false, + .endianness = 1, +}; + static int hdac_hda_dev_probe(struct hdac_device *hdev) { + struct hdac_hda_priv *hda_pvt = dev_get_drvdata(&hdev->dev); struct hdac_ext_link *hlink; int ret; @@ -621,9 +632,15 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) snd_hdac_ext_bus_link_get(hdev->bus, hlink); /* ASoC specific initialization */ - ret = devm_snd_soc_register_component(&hdev->dev, - &hdac_hda_codec, hdac_hda_dais, - ARRAY_SIZE(hdac_hda_dais)); + if (hda_pvt->need_display_power) + ret = devm_snd_soc_register_component(&hdev->dev, + &hdac_hda_hdmi_codec, hdac_hda_hdmi_dais, + ARRAY_SIZE(hdac_hda_hdmi_dais)); + else + ret = devm_snd_soc_register_component(&hdev->dev, + &hdac_hda_codec, hdac_hda_dais, + ARRAY_SIZE(hdac_hda_dais)); + if (ret < 0) { dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret); return ret; -- 2.43.0

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[PATCH] ASoC: SOF: acp: Add prevent against NULL
by Kamil Duljas 28 Nov '23

28 Nov '23

When kasprintf function fail, then NULL is returned. The callers dereference them without null checked. Signed-off-by: Kamil Duljas <kamil.duljas(a)gmail.com> --- sound/soc/sof/amd/acp.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 603ea5fc0d0d..c60a3e28b5f3 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -548,11 +548,15 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) dmi_id = dmi_first_match(acp_sof_quirk_table); if (dmi_id && dmi_id->driver_data) { adata->fw_code_bin = kasprintf(GFP_KERNEL, "%s/sof-%s-code.bin", - plat_data->fw_filename_prefix, - chip->name); + plat_data->fw_filename_prefix, + chip->name); + if (!adata->fw_code_bin) + return -ENOMEM; adata->fw_data_bin = kasprintf(GFP_KERNEL, "%s/sof-%s-data.bin", - plat_data->fw_filename_prefix, - chip->name); + plat_data->fw_filename_prefix, + chip->name); + if (!adata->fw_data_bin) + return -ENOMEM; adata->signed_fw_image = dmi_id->driver_data; dev_dbg(sdev->dev, "fw_code_bin:%s, fw_data_bin:%s\n", adata->fw_code_bin, -- 2.42.0.windows.2

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[PATCH 00/27] ASoC: Intel: Link handling rework and fixes
by Peter Ujfalusi 28 Nov '23

28 Nov '23

Hi, SOF board updates for 6.8 including few small fix and the majority is to add generic helpers for codec, amp, BT offload, HDMI-In and DAI link generation among various machine drivers. Regards, Peter --- Bard Liao (2): ASoC: Intel: sof_maxim_common: add else between 2 if test ASoC: Intel: sof_maxim_common: check return value Brent Lu (24): ASoC: Intel: sof_ssp_amp: remove dead code ASoC: Intel: ssp-common: get codec name function ASoC: Intel: board_helpers: support codec link initialization ASoC: Intel: sof_cs42l42: use common module for codec link ASoC: Intel: sof_nau8825: use common module for codec link ASoC: Intel: sof_rt5682: use common module for codec link ASoC: Intel: board_helpers: support amp link initialization ASoC: Intel: sof_cs42l42: use common module for amp link ASoC: Intel: sof_nau8825: use common module for amp link ASoC: Intel: sof_rt5682: use common module for amp link ASoC: Intel: sof_ssp_amp: use common module for amp link ASoC: Intel: sof_ssp_amp: rename function parameter ASoC: Intel: board_helpers: support BT offload link initialization ASoC: Intel: sof_cs42l42: use common module for BT offload link ASoC: Intel: sof_nau8825: use common module for BT offload link ASoC: Intel: sof_rt5682: use common module for BT offload link ASoC: Intel: sof_ssp_amp: use common module for BT offload link ASoC: Intel: sof_ssp_amp: simplify HDMI-In quirks ASoC: Intel: board_helpers: support HDMI-In link initialization ASoC: Intel: sof_rt5682: use common module for HDMI-In link ASoC: Intel: sof_ssp_amp: use common module for HDMI-In link ASoC: Intel: board_helpers: support DAI link array generation ASoC: Intel: sof_nau8825: use common module for DAI link generation ASoC: Intel: sof_rt5682: use common module for DAI link generation Chao Song (1): ASoC: Intel: cht_bsw_rt5672: check return value sound/soc/intel/boards/cht_bsw_rt5672.c | 8 +- sound/soc/intel/boards/sof_board_helpers.c | 330 +++++++++++++++++++++ sound/soc/intel/boards/sof_board_helpers.h | 31 ++ sound/soc/intel/boards/sof_cs42l42.c | 229 ++++---------- sound/soc/intel/boards/sof_maxim_common.c | 13 +- sound/soc/intel/boards/sof_nau8825.c | 224 ++++---------- sound/soc/intel/boards/sof_rt5682.c | 328 ++++++-------------- sound/soc/intel/boards/sof_ssp_amp.c | 160 ++++------ sound/soc/intel/boards/sof_ssp_common.c | 21 ++ sound/soc/intel/boards/sof_ssp_common.h | 1 + 10 files changed, 653 insertions(+), 692 deletions(-) -- 2.43.0

2 28

[PATCH 0/2] ALSA/ASoC: hdmi/hdac_hda: Conditionally register dais
by Peter Ujfalusi 28 Nov '23

28 Nov '23

Hi, Each instance of the hdac_hda registers DAIs to be used for analog and HDMI audio. When the system have more than one HDA codec - like most devices then the kernel log will include the following warning prints: snd_hda_codec_realtek ehdaudio0D0: ASoC: sink widget AIF1TX overwritten snd_hda_codec_realtek ehdaudio0D0: ASoC: source widget AIF1RX overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi3 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi2 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Codec Output Pin1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Codec Input Pin1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Analog Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Digital Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Alt Analog Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Analog Codec Capture overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Digital Codec Capture overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Alt Analog Codec Capture overwritten These are printed when the second instance of the hdac_hda is probing since it is re-registering the DAIs with the same name. Using some clever way of prefixing might solve the issue but that would need non backwards compatible changes to topology files. Pragmatically the hdac_hda instance fro the normal codec should not register DAIs for HDMI and the hdac_hda instance for the HDMI should not register DAIs for the analog codec - for example on a SDW device where we have HDA-HDMI and sdw codecs. To keep the hdac_hda vendor neutral (altrough it is only used on Intel platforms afaik) add a helper function to answer the question: is this codec HDMI? Use the new helper to decide which sets of DAIs to register for the probing hdac_hda instance. Regards, Peter --- Peter Ujfalusi (2): ALSA: hda/hdmi: Add helper function to check if a device is HDMI codec ASoC: hdac_hda: Conditionally register dais for HDMI and Analog include/sound/hdaudio.h | 10 ++++++++++ sound/pci/hda/patch_hdmi.c | 13 +++++++++++++ sound/soc/codecs/hdac_hda.c | 22 +++++++++++++++++++--- 3 files changed, 42 insertions(+), 3 deletions(-) -- 2.43.0

2 18

[PATCH] soundwire: stream: fix NULL pointer dereference for multi_link
by Krzysztof Kozlowski 28 Nov '23

28 Nov '23

If bus is marked as multi_link, but number of masters in the stream is not higher than bus->hw_sync_min_links (bus->multi_link && m_rt_count >= bus->hw_sync_min_links), bank switching should not happen. The first part of do_bank_switch() code properly takes these conditions into account, but second part (sdw_ml_sync_bank_switch()) relies purely on bus->multi_link property. This is not balanced and leads to NULL pointer dereference: Unable to handle kernel NULL pointer dereference at virtual address 0000000000000000 ... Call trace: wait_for_completion_timeout+0x124/0x1f0 do_bank_switch+0x370/0x6f8 sdw_prepare_stream+0x2d0/0x438 qcom_snd_sdw_prepare+0xa0/0x118 sm8450_snd_prepare+0x128/0x148 snd_soc_link_prepare+0x5c/0xe8 __soc_pcm_prepare+0x28/0x1ec dpcm_be_dai_prepare+0x1e0/0x2c0 dpcm_fe_dai_prepare+0x108/0x28c snd_pcm_do_prepare+0x44/0x68 snd_pcm_action_single+0x54/0xc0 snd_pcm_action_nonatomic+0xe4/0xec snd_pcm_prepare+0xc4/0x114 snd_pcm_common_ioctl+0x1154/0x1cc0 snd_pcm_ioctl+0x54/0x74 Fixes: ce6e74d008ff ("soundwire: Add support for multi link bank switch") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/stream.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 9dc6399f206a..f9c0adc0738d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -742,14 +742,15 @@ static int sdw_bank_switch(struct sdw_bus *bus, int m_rt_count) * sdw_ml_sync_bank_switch: Multilink register bank switch * * @bus: SDW bus instance + * @multi_link: whether this is a multi-link stream with hardware-based sync * * Caller function should free the buffers on error */ -static int sdw_ml_sync_bank_switch(struct sdw_bus *bus) +static int sdw_ml_sync_bank_switch(struct sdw_bus *bus, bool multi_link) { unsigned long time_left; - if (!bus->multi_link) + if (!multi_link) return 0; /* Wait for completion of transfer */ @@ -847,7 +848,7 @@ static int do_bank_switch(struct sdw_stream_runtime *stream) bus->bank_switch_timeout = DEFAULT_BANK_SWITCH_TIMEOUT; /* Check if bank switch was successful */ - ret = sdw_ml_sync_bank_switch(bus); + ret = sdw_ml_sync_bank_switch(bus, multi_link); if (ret < 0) { dev_err(bus->dev, "multi link bank switch failed: %d\n", ret); -- 2.34.1

3 2

Set Mic VREF to 100
by Kailang 28 Nov '23

28 Nov '23

Hi Takashi, This platform need to set MIC vref to 100%. BR, Kailang

2 1

[PATCH RFC] ALSA: scarlett2: Add ioctls for user-space access
by Geoffrey D. Bennett 27 Nov '23

27 Nov '23

In order to support functions such as firmware upgrade from user-space, add ioctls for submitting arbitrary proprietary requests through scarlett2_usb() and requesting/releasing exclusive access. --- Hi Takashi, I recently figured how to update the firmware on Scarlett Gen 2+ devices. I think the best way to implement this is with an ioctl giving access to the scarlett2_usb() function from user-space, plus two ioctls to request/release exclusive access. Does something like this seem reasonable? What ioctl magic/command values should I use? Should lock/unlock be separate, or one ioctl takes an argument for which operation to do? Thanks, Geoffrey. MAINTAINERS | 1 + include/uapi/sound/scarlett_gen2.h | 28 +++++++++ sound/usb/mixer_scarlett_gen2.c | 94 +++++++++++++++++++++++++++++- 3 files changed, 121 insertions(+), 2 deletions(-) create mode 100644 include/uapi/sound/scarlett_gen2.h diff --git a/MAINTAINERS b/MAINTAINERS index 4cc6bf79fdd8..25e5e40f7118 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -8066,6 +8066,7 @@ M: Geoffrey D. Bennett <g(a)b4.vu> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git +F: include/uapi/sound/scarlett_gen2.h F: sound/usb/mixer_scarlett_gen2.c FORCEDETH GIGABIT ETHERNET DRIVER diff --git a/include/uapi/sound/scarlett_gen2.h b/include/uapi/sound/scarlett_gen2.h new file mode 100644 index 000000000000..0b51a9754ba2 --- /dev/null +++ b/include/uapi/sound/scarlett_gen2.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ +/* + * Focusrite Scarlett Gen 2/3 and Clarett USB/Clarett+ Driver for ALSA + * + * Copyright (c) 2023 by Geoffrey D. Bennett <g at b4.vu> + */ +#ifndef __UAPI_SOUND_SCARLETT_GEN2_H +#define __UAPI_SOUND_SCARLETT_GEN2_H + +#include <linux/types.h> +#include <linux/ioctl.h> + +/* Send a proprietary format request to the interface */ +struct snd_scarlett_gen2_usb_cmd { + u32 cmd; + void *req_data; + u16 req_size; + void *resp_data; + u16 resp_size; +}; + +#define SCARLETT2_IOCTL_USB_CMD _IOWR('S', 0x60, struct snd_scarlett_gen2_usb_cmd) + +/* Request/release exclusive access */ +#define SCARLETT2_IOCTL_LOCK _IO('S', 0x61) +#define SCARLETT2_IOCTL_UNLOCK _IO('S', 0x62) + +#endif /* __UAPI_SOUND_SCARLETT_GEN2_H */ diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index ffd398f26d2c..f1ef87b54813 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -144,6 +144,9 @@ #include <sound/control.h> #include <sound/tlv.h> +#include <sound/hwdep.h> + +#include <uapi/sound/scarlett_gen2.h> #include "usbaudio.h" #include "mixer.h" @@ -4262,10 +4265,84 @@ static int snd_scarlett_gen2_controls_create( return 0; } +static int scarlett_gen2_ioctl_usb_cmd(struct snd_hwdep *hw, unsigned long arg) +{ + struct scarlett2_data *private = hw->private_data; + struct snd_scarlett_gen2_usb_cmd usb_cmd; + int err; + void *req_data = NULL; + void *resp_data = NULL; + + // get cmd & req/resp buffers + if (copy_from_user(&usb_cmd, (void __user *)arg, sizeof(usb_cmd))) + return -EFAULT; + + // allocate request buffer, copy data from user + if (usb_cmd.req_size > 0) { + req_data = kmalloc(usb_cmd.req_size, GFP_KERNEL); + if (!req_data) { + err = -ENOMEM; + goto exit; + } + if (copy_from_user(req_data, usb_cmd.req_data, + usb_cmd.req_size)) { + err = -EFAULT; + goto exit; + } + } + + // allocate response buffer + if (usb_cmd.resp_size > 0) { + resp_data = kmalloc(usb_cmd.resp_size, GFP_KERNEL); + if (!resp_data) { + err = -ENOMEM; + goto exit; + } + } + + // send request, get response + err = scarlett2_usb(private->mixer, usb_cmd.cmd, + req_data, usb_cmd.req_size, + resp_data, usb_cmd.resp_size); + if (err < 0) + goto exit; + + // copy response to user + if (usb_cmd.resp_size > 0) + if (copy_to_user(usb_cmd.resp_data, resp_data, + usb_cmd.resp_size)) + err = -EFAULT; + +exit: + kfree(req_data); + kfree(resp_data); + + return err; +} + +static int scarlett_gen2_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, + unsigned int cmd, unsigned long arg) +{ + switch (cmd) { + + case SCARLETT2_IOCTL_USB_CMD: + return scarlett_gen2_ioctl_usb_cmd(hw, arg); + + // TODO + case SCARLETT2_IOCTL_LOCK: + case SCARLETT2_IOCTL_UNLOCK: + return -EINVAL; + + default: + return -ENOIOCTLCMD; + } +} + int snd_scarlett_gen2_init(struct usb_mixer_interface *mixer) { struct snd_usb_audio *chip = mixer->chip; const struct scarlett2_device_entry *entry; + struct snd_hwdep *hw; int err; /* only use UAC_VERSION_2 */ @@ -4302,11 +4379,24 @@ int snd_scarlett_gen2_init(struct usb_mixer_interface *mixer) USB_ID_PRODUCT(chip->usb_id)); err = snd_scarlett_gen2_controls_create(mixer, entry); - if (err < 0) + if (err < 0) { usb_audio_err(mixer->chip, "Error initialising %s Mixer Driver: %d", entry->series_name, err); + return err; + } + + err = snd_hwdep_new(mixer->chip->card, "Focusrite Control", 0, &hw); + if (err < 0) { + usb_audio_err(mixer->chip, + "Error creating hwdep device: %d", + err); + return err; + } - return err; + hw->private_data = mixer->private_data; + hw->ops.ioctl = scarlett_gen2_hwdep_ioctl; + + return 0; } -- 2.41.0

3 8

[PATCH] ASoC: SOF: Intel: mtl: call dsp dump when boot retry fails
by Peter Ujfalusi 27 Nov '23

27 Nov '23

From: Yong Zhi <yong.zhi(a)intel.com> Call snd_sof_dsp_dbg_dump() with the same flags/dump_msg as used in function hda_loader.c/cl_dsp_init(). Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/mtl.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 254dbbeac1d0..3ef9e5c37028 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -440,7 +440,8 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; unsigned int status; - u32 ipc_hdr; + u32 ipc_hdr, flags; + char *dump_msg; int ret; /* step 1: purge FW request */ @@ -493,8 +494,18 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) return 0; err: - snd_sof_dsp_dbg_dump(sdev, "MTL DSP init fail", 0); + flags = SOF_DBG_DUMP_PCI | SOF_DBG_DUMP_MBOX | SOF_DBG_DUMP_OPTIONAL; + + /* after max boot attempts make sure that the dump is printed */ + if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) + flags &= ~SOF_DBG_DUMP_OPTIONAL; + + dump_msg = kasprintf(GFP_KERNEL, "Boot iteration failed: %d/%d", + hda->boot_iteration, HDA_FW_BOOT_ATTEMPTS); + snd_sof_dsp_dbg_dump(sdev, dump_msg, flags); mtl_dsp_core_power_down(sdev, SOF_DSP_PRIMARY_CORE); + + kfree(dump_msg); return ret; } -- 2.43.0

2 1

[PATCH] ASoC: SOF: Intel: Use existing helpers to change GPROCEN and PIE bits
by Peter Ujfalusi 27 Nov '23

27 Nov '23

Instead of directly changing the GPROCEN/PIE bits in PPCTL we should use the existing helper hda_dsp_ctrl_ppcap_enable() and hda_dsp_ctrl_ppcap_int_enable() helpers for clarity. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-loader.c | 5 ++--- sound/soc/sof/intel/hda.c | 6 ++---- 2 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 46fb2d1425e9..1805cf754beb 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -510,9 +510,8 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) return chip_info->init_core_mask; /* disable DSP */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, - SOF_HDA_REG_PP_PPCTL, - SOF_HDA_PPCTL_GPROCEN, 0); + hda_dsp_ctrl_ppcap_enable(sdev, false); + return ret; } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 744c0dd5766d..fe4ae349dad5 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1350,8 +1350,7 @@ void hda_dsp_remove(struct snd_sof_dev *sdev) if (!sdev->dspless_mode_selected) { /* disable DSP IRQ */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - SOF_HDA_PPCTL_PIE, 0); + hda_dsp_ctrl_ppcap_int_enable(sdev, false); } /* disable CIE and GIE interrupts */ @@ -1366,8 +1365,7 @@ void hda_dsp_remove(struct snd_sof_dev *sdev) chip->power_down_dsp(sdev); /* disable DSP */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - SOF_HDA_PPCTL_GPROCEN, 0); + hda_dsp_ctrl_ppcap_enable(sdev, false); skip_disable_dsp: free_irq(sdev->ipc_irq, sdev); -- 2.43.0

2 1

[PATCH v5 00/16] ALSA/ASoC: hda: Address format selection limitations and ambiguity
by Cezary Rojewski 27 Nov '23

27 Nov '23

Patchset aims to address format selection restrictions present currently in the HDAudio library. Formats which we are concerned about are 20 and 24 valid bits per sample within 32 bit depth container. One may identify them as S20_LE and S24_LE except that those, according to comments found in include/uapi/sound/asound.h, are for LSB-aligned scenarios. HDAudio streams expect MSB-aligned data, no matter if we are speaking of HOST (SDxFMT) or LINK (PPLCxFMT) side - chapter 4.5.1 of the public HDAudio specification. In short, S20_LE and S24_LE are invalid options. Right now, given the implementation of snd_hdac_query_supported_pcm() within sound/hda/hdac_device.c, even if a codec responds with: "I support all possible formats specified within HDAudio specification", there will be no option to open a 20/32 or 24/32 stream. The kernel will force the stream to be opened using the highest available bit depth. After discussing subject initially with Jaroslav and Takashi, suggestion was made to utilize 'subformat' option to address the problem. The eye-opening discussion begun much earlier though, in 2019 [1]. Paired with PRs for alsa-utils [2] and alsa-lib [3]. Flow of changes: The very first patch adds MSBITS subformat options to allow for granular 20/32, 24/32 and 32/32 format selection. The next two make sure subformat is actually honored during runtime. Most of that code is based on format-related API. Follow up is upgrade to the hda stream-format interface - several functions are added to make the granular format selection simple in the HDAudio world. Core of the implementation is based on the existing snd_hdac_calc_stream_format(). The next ten patches are straightforward switch from one interface to another with cleanup of now-unsed function as a finishing touch. Last but not least - the avs-driver, on which the problem analyzed and debugged, is updated to no longer acknowledge S24_LE as a valid format option. Results with skylake-driver and snd_hda_intel show status quo on our RVPs. PR filed on SOF github shows promising results too [4]. Changes in v5: - reworded 'bps' to 'bits' in all occurrences. - fixed an issue of MSBITS_MAX not being reported for S32_LE as reported by Jaroslav - snd_pcm_subformat_width() has been inlined into its only user: snd_pcm_hw_params_bits() as suggested by Jaroslav. - updated commit messege for patch 01 as it was out of date given the recent updates. Changes in v4: - fixed compilation issues in sof-driver, patch 12/16, reported by ikp - fixed sparse warnings in patch 01/16, reported by ikp - updated commit message for patch 03: "ASoC pcm: Honor subformat when configuring runtime", as snd_pcm_hw_copy() is gone since revision v3. Changes in v3: - merged the first two patches as suggested by Jaroslav - re-authored patch 01 to Jaroslav, added my Co-developed-by. - added Jaroslav' Co-developed-by to patch 02. - 'subformats' field now S32_LE-specific. Given the fact that it is the only format currently requiring subformat-intervention, functionality is narrowed to reduce amount of memory allocations and cleanup. Suggested by Jaroslav. - note to the above: the hdaudio part converted 1:1 as requested, patch 02/16 - note #2: alsa part converted to S32_LE-specific yet without addition of the chicken bit. Instead, struct snd_pcm_hardware is updated with u32 subformat mask to do the job. - ALSA-core additions in form of snd_pcm_subformat_width() and snd_pcm_hw_params_bps() relocated from 01/16 to the user, patch 05. Changes in v2: - patch 01/17, introduced struct snd_pcm_subformat which task is to represent subformat-mask on per format basis. Expectation is that manipulated arrays of subformats always end with a sentinel entry - patch 01/17, added snd_pcm_hw_copy() as the copying snd_pcm_hardware becomes non-trivial - patch 02/17, added hw_rule that produces final subformat mask based on provided formats as suggested by Jaroslav - patch 04/17, soc_pcm_hw_update_subformat() refactored as the subformat intersection becomes non-trivial - relevant functions releasing resources occupied by hda_pcm and snd_pcm_runtime updated to also kfree() subformats - except for 16/17, no changes to patches past 04/17, retaining acks for these Changes in v1: - fixed UBSAN due to missing snd_pcm_subformat_names[] entries for new subformats - as HDMI stream capabilities are assigned on PCM open, patch 16/17 has been updated to ignore such codecs for now. A separate patchset will take care of this case - params_bps() reworded to snd_pcm_hw_params_bps() - fixed compilation issues in sof-driver, patch 13/17 [1]: https://lore.kernel.org/alsa-devel/20190905053302.9262-1-pawel.harlozinski@… [2]: https://github.com/alsa-project/alsa-utils/pull/228 [3]: https://github.com/alsa-project/alsa-lib/pull/342 [4]: https://github.com/thesofproject/linux/pull/4539 Cezary Rojewski (15): ALSA: hda: Honor subformat when querying PCMs ASoC: pcm: Honor subformat when configuring runtime ALSA: hda: Upgrade stream-format infrastructure ALSA: hda: Switch to new stream-format interface ALSA: hda/hdmi: Switch to new stream-format interface ALSA: hda/ca0132: Switch to new stream-format interface ASoC: codecs: hda: Switch to new stream-format interface ASoC: codecs: hdac_hda: Switch to new stream-format interface ASoC: codecs: hdac_hdmi: Switch to new stream-format interface ASoC: Intel Skylake: Switch to new stream-format interface ASoC: SOF: Intel: Switch to new stream-format interface ASoC: Intel: avs: Switch to new stream-format interface ALSA: hda: Drop snd_hdac_calc_stream_format() ASoC: Intel: avs: Kill S24_LE format ASoC: Intel: avs: Unhardcode HDAudio BE DAI drivers description Jaroslav Kysela (1): ALSA: pcm: Introduce MSBITS subformat interface include/sound/hda_codec.h | 5 +- include/sound/hdaudio.h | 13 +-- include/sound/pcm.h | 7 ++ include/sound/pcm_params.h | 2 + include/sound/soc.h | 1 + include/uapi/sound/asound.h | 7 +- sound/core/pcm.c | 3 + sound/core/pcm_lib.c | 34 +++++++ sound/core/pcm_native.c | 55 ++++++++++- sound/hda/hdac_device.c | 156 +++++++++++++++++++++--------- sound/pci/hda/hda_codec.c | 2 + sound/pci/hda/hda_controller.c | 10 +- sound/pci/hda/patch_ca0132.c | 3 +- sound/pci/hda/patch_hdmi.c | 6 +- sound/soc/codecs/hda-dai.c | 7 +- sound/soc/codecs/hda.c | 2 + sound/soc/codecs/hdac_hda.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 10 +- sound/soc/intel/avs/loader.c | 4 +- sound/soc/intel/avs/path.c | 2 +- sound/soc/intel/avs/pcm.c | 58 +++++++++-- sound/soc/intel/avs/probes.c | 3 +- sound/soc/intel/avs/topology.c | 13 ++- sound/soc/intel/skylake/skl-pcm.c | 13 ++- sound/soc/soc-pcm.c | 10 ++ sound/soc/sof/intel/hda-dai-ops.c | 21 ++-- tools/include/uapi/sound/asound.h | 7 +- 27 files changed, 352 insertions(+), 110 deletions(-) -- 2.25.1

3 19

ASoC: amd: yc: Fix non-functional mic on ASUS E1504FA
by Malcolm Hart 27 Nov '23

27 Nov '23

Like other ASUS models the Asus Vivobook E1504FA requires an entry in the quirk list to enable the internal microphone. Showing with 7 additions and 0 deletions. 7 changes: 7 additions & 0 deletions 7 sound/soc/amd/yc/acp6x-mach.c @@ -283,6 +283,13 @@ 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, "E1504FA"), } }, { I have this laptop and I have tested this patch successfully. Malcolm

5 8

[PATCH] ALSA: hda: intel-nhlt: Ignore vbps when looking for DMIC 32 bps format
by Peter Ujfalusi 27 Nov '23

27 Nov '23

When looking up DMIC blob from the NHLT table and the format is 32 bits, ignore the vbps matching for 32 bps for DMIC since some NHLT table have the vbps as 24, some have it as 32. The DMIC hardware supports only one type of 32 bit sample size, which is 24 bit sampling on the MSB side and bits[1:0] is used for indicating the channel number. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/hda/intel-nhlt.c | 33 +++++++++++++++++++++++++++++---- 1 file changed, 29 insertions(+), 4 deletions(-) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 2c4dfc0b7e34..696a958d93e9 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -238,7 +238,7 @@ EXPORT_SYMBOL(intel_nhlt_ssp_mclk_mask); static struct nhlt_specific_cfg * nhlt_get_specific_cfg(struct device *dev, struct nhlt_fmt *fmt, u8 num_ch, - u32 rate, u8 vbps, u8 bps) + u32 rate, u8 vbps, u8 bps, bool ignore_vbps) { struct nhlt_fmt_cfg *cfg = fmt->fmt_config; struct wav_fmt *wfmt; @@ -255,8 +255,12 @@ nhlt_get_specific_cfg(struct device *dev, struct nhlt_fmt *fmt, u8 num_ch, dev_dbg(dev, "Endpoint format: ch=%d fmt=%d/%d rate=%d\n", wfmt->channels, _vbps, _bps, wfmt->samples_per_sec); + /* + * When looking for exact match of configuration ignore the vbps + * from NHLT table when ignore_vbps is true + */ if (wfmt->channels == num_ch && wfmt->samples_per_sec == rate && - vbps == _vbps && bps == _bps) + (ignore_vbps || vbps == _vbps) && bps == _bps) return &cfg->config; cfg = (struct nhlt_fmt_cfg *)(cfg->config.caps + cfg->config.size); @@ -289,6 +293,7 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, { struct nhlt_specific_cfg *cfg; struct nhlt_endpoint *epnt; + bool ignore_vbps = false; struct nhlt_fmt *fmt; int i; @@ -298,7 +303,26 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, dev_dbg(dev, "Looking for configuration:\n"); dev_dbg(dev, " vbus_id=%d link_type=%d dir=%d, dev_type=%d\n", bus_id, link_type, dir, dev_type); - dev_dbg(dev, " ch=%d fmt=%d/%d rate=%d\n", num_ch, vbps, bps, rate); + if (link_type == NHLT_LINK_DMIC && bps == 32 && (vbps == 24 || vbps == 32)) { + /* + * The DMIC hardware supports only one type of 32 bits sample + * size, which is 24 bit sampling on the MSB side and bits[1:0] + * are used for indicating the channel number. + * It has been observed that some NHLT tables have the vbps + * specified as 32 while some uses 24. + * The format these variations describe are identical, the + * hardware is configured and behaves the same way. + * Note: when the samples assumed to be vbps=32 then the 'noise' + * introduced by the lower two bits (channel number) have no + * real life implication on audio quality. + */ + dev_dbg(dev, + " ch=%d fmt=%d rate=%d (vbps is ignored for DMIC 32bit format)\n", + num_ch, bps, rate); + ignore_vbps = true; + } else { + dev_dbg(dev, " ch=%d fmt=%d/%d rate=%d\n", num_ch, vbps, bps, rate); + } dev_dbg(dev, "Endpoint count=%d\n", nhlt->endpoint_count); epnt = (struct nhlt_endpoint *)nhlt->desc; @@ -307,7 +331,8 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, if (nhlt_check_ep_match(dev, epnt, bus_id, link_type, dir, dev_type)) { fmt = (struct nhlt_fmt *)(epnt->config.caps + epnt->config.size); - cfg = nhlt_get_specific_cfg(dev, fmt, num_ch, rate, vbps, bps); + cfg = nhlt_get_specific_cfg(dev, fmt, num_ch, rate, + vbps, bps, ignore_vbps); if (cfg) return cfg; } -- 2.43.0

2 1

Yamaha YIT-W12TX not working
by Julian Sikorski 27 Nov '23

27 Nov '23

Hi list, Yamaha YIT-W12TX is a USB dongle which allows to transmit audio from PC to the YSP-4300 sound bar [1][2]. While it is quite old and no longer sold, I recently tried to make it work under linux. Unfortunately, it did not work: [ 3676.366404] usb 2-1.1: new full-speed USB device number 5 using ehci-pci [ 3676.445995] usb 2-1.1: config 1 has an invalid interface number: 4 but max is 3 [ 3676.446001] usb 2-1.1: config 1 has no interface number 2 [ 3676.446664] usb 2-1.1: New USB device found, idVendor=0499, idProduct=3108, bcdDevice= 1.00 [ 3676.446666] usb 2-1.1: New USB device strings: Mfr=2, Product=1, SerialNumber=0 [ 3676.446679] usb 2-1.1: Product: YIT-W12TX [ 3676.446683] usb 2-1.1: Manufacturer: YAMAHA Corp. [ 3676.449508] hid-generic 0003:0499:3108.0001: hiddev96,hidraw0: USB HID v0.01 Device [YAMAHA Corp. YIT-W12TX] on usb-0000:00:1d.0-1.1/input3 [ 3676.450846] input: YAMAHA Corp. YIT-W12TX as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.1/2-1.1:1.4/0003:0499:3108.0002/input/input34 [ 3676.503844] hid-generic 0003:0499:3108.0002: input,hidraw1: USB HID v1.00 Device [YAMAHA Corp. YIT-W12TX] on usb-0000:00:1d.0-1.1/input4 [ 3676.671641] usb 2-1.1: 1:1: cannot get freq at ep 0x1 [ 3676.675786] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.676476] usbcore: registered new interface driver snd-usb-audio [ 3676.731911] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.735658] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.739399] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.743131] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.746920] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.750648] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.754445] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.758524] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.762274] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.789671] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.790049] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.790414] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.790773] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.791169] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.792134] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.792547] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.792899] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.793254] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.793634] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.794900] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.795255] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.795634] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.796008] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.796384] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.797543] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.797879] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.798254] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.798634] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.799007] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.813819] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.814281] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.814686] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.815281] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.815929] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.817807] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.818155] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.818592] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.819165] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.819801] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.822163] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.822802] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.823284] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.824161] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.824907] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.827683] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.828159] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.828675] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.829415] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.830157] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.841801] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.842155] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.842551] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.843158] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.843785] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.844909] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.845283] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.845673] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.846156] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.846800] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.848287] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.848799] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.849280] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.850031] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.850799] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.852677] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.853154] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.853674] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.854429] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.855155] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.858916] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.859277] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.859672] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.860280] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.860909] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.862033] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.862418] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.862797] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.863279] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.863923] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.865471] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.866033] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.866590] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.867410] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.868157] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.870162] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.870674] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.871155] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.871908] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.872676] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.900672] usb 2-1.1: USB disconnect, device number 5 [ 3676.900688] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 This is with kernel-5.9.13-200.fc33.x86_64 and alsa-lib-1.2.4-5.fc33.x86_64 on Fedora 33. Under Windows 10 the dongle works without any special drivers from Yamaha, everything appears to be downloaded by Windows automatically, so I was hoping the device could be made to work under Linux relatively easily. Best regards, Julian [1] https://europe.yamaha.com/en/products/audio_visual/accessories/yit-w12/down… [2] https://europe.yamaha.com/en/products/audio_visual/sound_bar/ysp-4300/downl…

2 7

[PATCH] sound: samsung: Add error handling to clk_set_rate in smdk_spdif
by Haoran Liu 27 Nov '23

27 Nov '23

This patch adds error handling for clk_set_rate calls in the set_audio_clock_rate function of the smdk_spdif.c file. Previously, the return value of clk_set_rate was not checked, which could potentially lead to unhandled errors in setting the audio clock rate. This change ensures that if clk_set_rate fails, the error is properly reported and handled. Signed-off-by: Haoran Liu <liuhaoran14(a)163.com> --- sound/soc/samsung/smdk_spdif.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index 2474eb619882..c31b82b89148 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -75,6 +75,7 @@ static int set_audio_clock_rate(unsigned long epll_rate, unsigned long audio_rate) { struct clk *fout_epll, *sclk_spdif; + int ret; fout_epll = clk_get(NULL, "fout_epll"); if (IS_ERR(fout_epll)) { @@ -82,7 +83,14 @@ static int set_audio_clock_rate(unsigned long epll_rate, return -ENOENT; } - clk_set_rate(fout_epll, epll_rate); + ret = clk_set_rate(fout_epll, epll_rate); + if (ret) { + printk(KERN_ERR "%s: failed to set rate for fout_epll\n", + __func__); + clk_put(fout_epll); + return ret; + } + clk_put(fout_epll); sclk_spdif = clk_get(NULL, "sclk_spdif"); @@ -91,7 +99,15 @@ static int set_audio_clock_rate(unsigned long epll_rate, return -ENOENT; } - clk_set_rate(sclk_spdif, audio_rate); + ret = clk_set_rate(sclk_spdif, audio_rate); + if (ret) { + printk(KERN_ERR "%s: failed to set rate for sclk_spdif\n", + __func__); + clk_put(sclk_spdif); + clk_put(fout_epll); + return ret; + } + clk_put(sclk_spdif); return 0; -- 2.17.1

1 0

[PATCH v6 00/11] Introduce STM32 Firewall framework
by Gatien Chevallier 27 Nov '23

27 Nov '23

Introduce STM32 Firewall framework for STM32MP1x and STM32MP2x platforms. STM32MP1x(ETZPC) and STM32MP2x(RIFSC) Firewall controllers register to the framework to offer firewall services such as access granting. This series of patches is a new approach on the previous STM32 system bus, history is available here: https://lore.kernel.org/lkml/20230127164040.1047583/ The need for such framework arises from the fact that there are now multiple hardware firewalls implemented across multiple products. Drivers are shared between different products, using the same code. When it comes to firewalls, the purpose mostly stays the same: Protect hardware resources. But the implementation differs, and there are multiple types of firewalls: peripheral, memory, ... Some hardware firewall controllers such as the RIFSC implemented on STM32MP2x platforms may require to take ownership of a resource before being able to use it, hence the requirement for firewall services to take/release the ownership of such resources. On the other hand, hardware firewall configurations are becoming more and more complex. These mecanisms prevent platform crashes or other firewall-related incoveniences by denying access to some resources. The stm32 firewall framework offers an API that is defined in firewall controllers drivers to best fit the specificity of each firewall. For every peripherals protected by either the ETZPC or the RIFSC, the firewall framework checks the firewall controlelr registers to see if the peripheral's access is granted to the Linux kernel. If not, the peripheral is configured as secure, the node is marked populated, so that the driver is not probed for that device. The firewall framework relies on the access-controller device tree binding. It is used by peripherals to reference a domain access controller. In this case a firewall controller. The bus uses the ID referenced by the access-controller property to know where to look in the firewall to get the security configuration for the peripheral. This allows a device tree description rather than a hardcoded peripheral table in the bus driver. The STM32 ETZPC device is responsible for filtering accesses based on security level, or co-processor isolation for any resource connected to it. The RIFSC is responsible for filtering accesses based on Compartment ID / security level / privilege level for any resource connected to it. STM32MP13/15/25 SoC device tree files are updated in this series to implement this mecanism. Changes in V6: - Rename access-controller to access-controllers - Remove access-controller-provider - Update device trees and other bindings accordingly - Rework ETZPC/RIFSC bindings to define what access-controllers cells contain inside #access-controller-cells - Some other minor fixes Changes in V5: - Integrate and rework the "feature-domains" binding patch in this patchset. The binding is renamed to "access-controller" - Rename every feature-domain* reference to access-control* ones - Correct loop bug and missing select STM32_FIREWALL in 32-bit platform Kconfig Changes in V4: - Fix typo in commit message and YAML check errors in "dt-bindings: Document common device controller bindings" Note: This patch should be ignored as stated in the cover letter. I've done this to avoid errors on this series of patch - Correct code syntax/style issues reported by Simon Horman - Added Jonathan's tag for IIO on the treewide patch Changes in V3: Change incorrect ordering for bindings commits leading to an error while running "make DT_CHECKER_FLAGS=-m dt_binding_check" Changes in V2: generic: - Add fw_devlink dependency for "feature-domains" property. bindings: - Corrected YAMLS errors highlighted by Rob's robot - Firewall controllers YAMLs no longer define the maxItems for the "feature-domains" property - Renamed st,stm32-rifsc.yaml to st,stm32mp25-rifsc.yaml - Fix examples in YAML files - Change feature-domains maxItems to 2 in firewall consumer files as there should not be more than 2 entries for now - Declare "feature-domain-names" as an optional property for firewall controllers child nodes. - Add missing "feature-domains" property declaration in bosch,m_can.yaml and st,stm32-cryp.yaml files firewall framework: - Support multiple entries for "feature-domains" property - Better handle the device-tree parsing using phandle+args APIs - Remove "resource firewall" type - Add a field for the name of the firewall entry - Fix licenses RIFSC: - Add controller name - Driver is now a module_platform_driver - Fix license ETZPC: - Add controller name - Driver is now a module_platform_driver - Fix license Device trees: - Fix rifsc node name - Move the "ranges" property under the "feature-domains" one Gatien Chevallier (10): dt-bindings: treewide: add access-controllers description dt-bindings: bus: document RIFSC dt-bindings: bus: document ETZPC firewall: introduce stm32_firewall framework of: property: fw_devlink: Add support for "access-controller" bus: rifsc: introduce RIFSC firewall controller driver arm64: dts: st: add RIFSC as an access controller for STM32MP25x boards bus: etzpc: introduce ETZPC firewall controller driver ARM: dts: stm32: add ETZPC as a system bus for STM32MP15x boards ARM: dts: stm32: add ETZPC as a system bus for STM32MP13x boards Oleksii Moisieiev (1): dt-bindings: document generic access controllers .../access-controllers.yaml | 84 + .../bindings/bus/st,stm32-etzpc.yaml | 87 + .../bindings/bus/st,stm32mp25-rifsc.yaml | 96 + .../bindings/crypto/st,stm32-cryp.yaml | 4 + .../bindings/crypto/st,stm32-hash.yaml | 4 + .../devicetree/bindings/dma/st,stm32-dma.yaml | 4 + .../bindings/dma/st,stm32-dmamux.yaml | 4 + .../devicetree/bindings/i2c/st,stm32-i2c.yaml | 4 + .../bindings/iio/adc/st,stm32-adc.yaml | 4 + .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 4 + .../bindings/iio/dac/st,stm32-dac.yaml | 4 + .../bindings/media/cec/st,stm32-cec.yaml | 4 + .../bindings/media/st,stm32-dcmi.yaml | 4 + .../memory-controllers/st,stm32-fmc2-ebi.yaml | 4 + .../bindings/mfd/st,stm32-lptimer.yaml | 4 + .../bindings/mfd/st,stm32-timers.yaml | 4 + .../devicetree/bindings/mmc/arm,pl18x.yaml | 4 + .../bindings/net/can/bosch,m_can.yaml | 4 + .../devicetree/bindings/net/stm32-dwmac.yaml | 4 + .../bindings/phy/phy-stm32-usbphyc.yaml | 4 + .../bindings/regulator/st,stm32-vrefbuf.yaml | 4 + .../devicetree/bindings/rng/st,stm32-rng.yaml | 4 + .../bindings/serial/st,stm32-uart.yaml | 4 + .../bindings/sound/st,stm32-i2s.yaml | 4 + .../bindings/sound/st,stm32-sai.yaml | 4 + .../bindings/sound/st,stm32-spdifrx.yaml | 4 + .../bindings/spi/st,stm32-qspi.yaml | 4 + .../devicetree/bindings/spi/st,stm32-spi.yaml | 4 + .../devicetree/bindings/usb/dwc2.yaml | 4 + MAINTAINERS | 7 + arch/arm/boot/dts/st/stm32mp131.dtsi | 1025 +++--- arch/arm/boot/dts/st/stm32mp133.dtsi | 51 +- arch/arm/boot/dts/st/stm32mp13xc.dtsi | 19 +- arch/arm/boot/dts/st/stm32mp13xf.dtsi | 19 +- arch/arm/boot/dts/st/stm32mp151.dtsi | 2756 +++++++++-------- arch/arm/boot/dts/st/stm32mp153.dtsi | 52 +- arch/arm/boot/dts/st/stm32mp15xc.dtsi | 19 +- arch/arm/mach-stm32/Kconfig | 1 + arch/arm64/Kconfig.platforms | 1 + arch/arm64/boot/dts/st/stm32mp251.dtsi | 6 +- drivers/bus/Kconfig | 9 + drivers/bus/Makefile | 1 + drivers/bus/stm32_etzpc.c | 141 + drivers/bus/stm32_firewall.c | 294 ++ drivers/bus/stm32_firewall.h | 83 + drivers/bus/stm32_rifsc.c | 252 ++ drivers/of/property.c | 2 + include/linux/bus/stm32_firewall_device.h | 141 + 48 files changed, 3331 insertions(+), 1919 deletions(-) create mode 100644 Documentation/devicetree/bindings/access-controllers/access-controllers.yaml create mode 100644 Documentation/devicetree/bindings/bus/st,stm32-etzpc.yaml create mode 100644 Documentation/devicetree/bindings/bus/st,stm32mp25-rifsc.yaml create mode 100644 drivers/bus/stm32_etzpc.c create mode 100644 drivers/bus/stm32_firewall.c create mode 100644 drivers/bus/stm32_firewall.h create mode 100644 drivers/bus/stm32_rifsc.c create mode 100644 include/linux/bus/stm32_firewall_device.h -- 2.35.3

2 18

[RFC PATCH 0/2] ASoC: soc-dai: add flag to mute and unmute stream during trigger.
by Srinivas Kandagatla 27 Nov '23

27 Nov '23

Click/Pop Noise was a long pending issue with WSA Codecs which are prone to accumlate DC when ports are active but without any data streams. There are multiple places in the current setup, where this could happen in both startup as well as shutdown path. This patchset adds a new flag mute_unmute_on_trigger to dai_ops to let generic code do the mute/unmute on trigger. This patches help fix those issues by making sure the PA is Muted/Unmuted inline with the stream start/stop events. Srinivas Kandagatla (2): ASoC: soc-dai: add flag to mute and unmute stream during trigger ASoC: codecs: wsa883x: make use of new mute_unmute_on_trigger flag include/sound/soc-dai.h | 1 + sound/soc/codecs/wsa883x.c | 7 +------ sound/soc/soc-dai.c | 7 +++++++ sound/soc/soc-pcm.c | 12 ++++++++---- 4 files changed, 17 insertions(+), 10 deletions(-) -- 2.21.0

5 13

[PATCH v2 0/1] ASoC: soc-pcm.c: Make sure DAI parameters cleared if the DAI becomes inactive
by Chancel Liu 27 Nov '23

27 Nov '23

It's reasonable that DAI parameters should be cleared if current DAI becomes inactive. Only check DAI active status in soc_pcm_hw_free() is not enough since there's the risk that DAI parameters never be cleared if there're more than one stream [see A]. Only check DAI active status in soc_pcm_close() is also not enough since it will cause the cleanup just happening in soc_pcm_close() [see B]. [A] For example, we have stream1 and stream2 for DAI about to stop. stream2 executes soc_pcm_hw_free() before stream1 executes soc_pcm_close(). At the moment, stream2 should clear current DAI parameters because stream1 passed the cleanup stage in soc_pcm_hw_free() and stream2 in fact is the only active stream which has a chance to do clean up. Since DAI active status is not yet updated by stream1 in soc_pcm_close(), stream2 doesn't know itself should clear DAI parameters. In result both stream1 and stream2 don't clear the parameters. [B] Suppose a case: aplay -Dhw:0 44100.wav 48000.wav In this case, we call soc_pcm_open()->soc_pcm_hw_params()->soc_pcm_hw_free() ->soc_pcm_hw_params()->soc_pcm_hw_free()->soc_pcm_close() in order. The DAI parameters would be remained in the system even if the playback of 44100.wav is finished. In conclusion, it's better to check DAI active status in both soc_pcm_hw_free() and soc_pcm_close() which makes sure DAI parameters cleared if the DAI becomes inactive. changes in v2: - Patch v1 link: https://lore.kernel.org/all/20230112065834.580192-1-chancel.liu@nxp.com/ Patch v1 tries to introduce a usage count called hw_params_count to fix issue on DAI parameters cleanup. However it's not a good fix because not considering hw_params() and hw_free() are not symmetrical and hw_params() might be called multilpe times by user. - Both check DAI active status in soc_pcm_hw_free() and soc_pcm_close() which makes sure DAI parameters cleared if the DAI becomes inactive. Chancel Liu (1): ASoC: soc-pcm.c: Make sure DAI parameters cleared if the DAI becomes inactive sound/soc/soc-pcm.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) -- 2.25.1

3 4

[PATCH v3 0/3] Add audio device nodes for i.MX93 platform
by Chancel Liu 27 Nov '23

27 Nov '23

changes in v3 - drop unused compatible string in fsl,mqs.yaml changes in v2 - Move MQS device node under "/" root node - Convert fsl,mqs.txt to yaml Chancel Liu (3): arm64: dts: imx93: Add audio device nodes arm64: dts: imx93: Configure clock rate for audio PLL ASoC: dt-bindings: fsl,mqs: Convert format to json-schema .../devicetree/bindings/sound/fsl,mqs.txt | 36 ------ .../devicetree/bindings/sound/fsl,mqs.yaml | 105 ++++++++++++++++++ arch/arm64/boot/dts/freescale/imx93.dtsi | 89 +++++++++++++++ 3 files changed, 194 insertions(+), 36 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.yaml -- 2.42.0

3 6

[PATCH v7 resend 0/5] ASoC: makes CPU/Codec channel connection map more generic
by Kuninori Morimoto 27 Nov '23

27 Nov '23

Hi Mark Cc Bard, Pierre-Louis, Jerome, DT-ML This is v7 patch-set. 2 weeks past, this is resend patch-set. Current ASoC is supporting CPU/Codec = N:M (N < M) connection by using ch_map idea. This patch-set expands it that all connection uses this idea, and no longer N < M limit [1][2]. Link: https://lore.kernel.org/r/87fs6wuszr.wl-kuninori.morimoto.gx@renesas.com [1] Link: https://lore.kernel.org/r/878r7yqeo4.wl-kuninori.morimoto.gx@renesas.com [2] ASoC core code ([PATCH 1/5]) is same as v6 and it was tested by Pierre-Louis, and Jerome. Big change on v7 is basically for Audio-Graph-Card2. v6 -> v7 - use "endpoint" for N:M connection instead of ch-map-idx on Audio-Graph-Card2 v5 -> v6 - tidyup some warnings - rename "ch-map-idx" -> "channel-map-index" - Update "channel-map-index" description - add Tested-by from Pierre-Louis / Jerome v4 -> v5 - use cpu/codec index on ch_maps - separate card2 sample DT patch into prepare and new feature - ch-maps -> ch-map-idx on DT v3 -> v4 - add Jerome on To - add "description" on "ch-maps" v2 -> v3 - tidyup comment - use more clear connection image on DT - "ch_maps" -> "ch-maps" on DT - Add DT maintainer on "To:" for all patches v1 -> v2 - makes CPU/Codec connection relation clear on comment/sample - fixup type "connction" -> "connection" - makes error message clear Kuninori Morimoto (5): ASoC: makes CPU/Codec channel connection map more generic ASoC: audio-graph-card2: use better image for Multi connection ASoC: audio-graph-card2: add CPU:Codec = N:M support ASoC: audio-graph-card2-custom-sample: Add connection image ASoC: audio-graph-card2-custom-sample: add CPU/Codec = N:M sample include/sound/soc.h | 56 ++- .../audio-graph-card2-custom-sample.dtsi | 380 ++++++++++++++++-- sound/soc/generic/audio-graph-card2.c | 277 ++++++++++--- sound/soc/intel/boards/sof_sdw.c | 28 +- sound/soc/soc-core.c | 95 ++++- sound/soc/soc-dapm.c | 45 +-- sound/soc/soc-pcm.c | 44 +- 7 files changed, 754 insertions(+), 171 deletions(-) -- 2.25.1

1 5

[PATCH alsa-ucm-conf] chtrt5645: Add support for the components string
by Hans de Goede 26 Nov '23

26 Nov '23

Upcoming kernel versions will report which speaker and dmic config is used by the device in a components strings so that we don't need to duplicate the DMI quirks in both the kernel and the UCM profile. Add support for getting the speaker and dmic config from the components string. Note the old DMI matching is kept for support of older kernels, this means that on devices where the old DMI matching was used things like: Define.MonoSpeaker "Yes" Will now be done twice, this is harmless as long as the kernel and UCM profile DMI quirks are in sync, which they are. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/Intel/chtrt5645/HiFi.conf | 35 ++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/ucm2/Intel/chtrt5645/HiFi.conf b/ucm2/Intel/chtrt5645/HiFi.conf index 30fad50..f8dcbf5 100644 --- a/ucm2/Intel/chtrt5645/HiFi.conf +++ b/ucm2/Intel/chtrt5645/HiFi.conf @@ -2,6 +2,41 @@ Define.MonoSpeaker "" Define.AnalogMic "yes" Define.DigitalMic "" +If.components-dmic1 { + Condition { + Type String + Haystack "${CardComponents}" + Needle "cfg-mic:dmic1" + } + True { + Define.AnalogMic "" + Define.DigitalMic "DMIC1" + } +} + +If.components-dmic2 { + Condition { + Type String + Haystack "${CardComponents}" + Needle "cfg-mic:dmic2" + } + True { + Define.AnalogMic "" + Define.DigitalMic "DMIC2" + } +} + +If.components-mspk { + Condition { + Type String + Haystack "${CardComponents}" + Needle "cfg-spk:1" + } + True { + Define.MonoSpeaker "yes" + } +} + If.cfg-dmic1 { Condition { Type RegexMatch -- 2.41.0

1 0

[PATCH 0/4] ASoC: SOF: ipc4: Add support for control change notification
by Peter Ujfalusi 25 Nov '23

25 Nov '23

Hi, This series adds support for handling control (switch/enum) change notifications sent by the firmware. The use case is similar to what is already used by IPC3 version: the firmware can update the value of an enum or switch and sends notification to the kernel, which in turn will notify the user space of a change. Regards, Peter --- Peter Ujfalusi (4): ASoC: SOF: ipc4-topology: Helper to find an swidget by module/instance id ASoC: SOF: ipc4: Add data struct for module notification message from firmware ASoC: SOF: ipc4-control: Implement control update for switch/enum controls ASoC: SOF: ipc4: Handle ALSA kcontrol change notification from firmware include/sound/sof/ipc4/header.h | 29 ++++++ sound/soc/sof/ipc4-control.c | 179 ++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-priv.h | 3 + sound/soc/sof/ipc4-topology.c | 20 ++++ sound/soc/sof/ipc4.c | 57 ++++++++++ 5 files changed, 288 insertions(+) -- 2.42.1

2 5

[PATCH 0/2] ASoC: SOF: Extend the enabled DSP core handling
by Peter Ujfalusi 25 Nov '23

25 Nov '23

Hi, In the current code, we enable a widget core when it is set up and disable it when it is freed. This is problematic with IPC4 because widget free is essentially a NOP and all widgets are freed in the firmware when the pipeline is deleted. This results in a crash during pipeline deletion when one of it's widgets is scheduled to run on a secondary core and is powered off when widget is freed. So, change the logic to enable all cores needed by all the modules in a pipeline when the pipeline widget is set up and disable them after the pipeline widget is freed. Regards, Peter --- Ranjani Sridharan (2): ASoC: SOF: ipc4-topology: Add core_mask in struct snd_sof_pipeline ASoC: SOF: sof-audio: Modify logic for enabling/disabling topology cores sound/soc/sof/ipc3-topology.c | 2 ++ sound/soc/sof/ipc4-topology.c | 9 +++++ sound/soc/sof/sof-audio.c | 65 ++++++++++++++++++++++------------- sound/soc/sof/sof-audio.h | 2 ++ 4 files changed, 54 insertions(+), 24 deletions(-) -- 2.42.1

2 3

[PATCH] ASoC: dt-bindings: correct white-spaces in examples
by Krzysztof Kozlowski 25 Nov '23

25 Nov '23

Use only one and exactly one space around '=' in DTS example. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Cc: Neil Armstrong <neil.armstrong(a)linaro.org> --- .../devicetree/bindings/sound/nuvoton,nau8821.yaml | 2 +- Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml | 4 ++-- .../devicetree/bindings/sound/qcom,wcd938x-sdw.yaml | 4 ++-- Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml | 4 ++-- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 8 ++++---- .../devicetree/bindings/sound/ti,tlv320aic32x4.yaml | 2 +- 6 files changed, 12 insertions(+), 12 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml index 3380b6aa9542..054b53954ac3 100644 --- a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -135,7 +135,7 @@ examples: nuvoton,jack-insert-debounce = <7>; nuvoton,jack-eject-debounce = <0>; nuvoton,dmic-clk-threshold = <3072000>; - nuvoton,dmic-slew-rate= <0>; + nuvoton,dmic-slew-rate = <0>; #sound-dai-cells = <0>; }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml index 4df59f3b7b01..beb0ff0245b0 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml @@ -201,9 +201,9 @@ examples: - | codec@1,0{ compatible = "slim217,250"; - reg = <1 0>; + reg = <1 0>; reset-gpios = <&tlmm 64 0>; - slim-ifc-dev = <&wcd9340_ifd>; + slim-ifc-dev = <&wcd9340_ifd>; #sound-dai-cells = <1>; interrupt-parent = <&tlmm>; interrupts = <54 4>; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml index b430dd3e1841..7b31bf93f1a1 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml @@ -51,7 +51,7 @@ examples: reg = <0x03210000 0x2000>; wcd938x_rx: codec@0,4 { compatible = "sdw20217010d00"; - reg = <0 4>; + reg = <0 4>; qcom,rx-port-mapping = <1 2 3 4 5>; }; }; @@ -62,7 +62,7 @@ examples: reg = <0x03230000 0x2000>; wcd938x_tx: codec@0,3 { compatible = "sdw20217010d00"; - reg = <0 3>; + reg = <0 3>; qcom,tx-port-mapping = <2 3 4 5>; }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml index 018565793a3e..adbfa67f88ed 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml @@ -137,7 +137,7 @@ examples: reg = <0x03210000 0x2000>; wcd938x_rx: codec@0,4 { compatible = "sdw20217010d00"; - reg = <0 4>; + reg = <0 4>; qcom,rx-port-mapping = <1 2 3 4 5>; }; }; @@ -148,7 +148,7 @@ examples: reg = <0x03230000 0x2000>; wcd938x_tx: codec@0,3 { compatible = "sdw20217010d00"; - reg = <0 3>; + reg = <0 3>; qcom,tx-port-mapping = <2 3 4 5>; }; }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 1174205286d4..0d7a6b576d88 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -497,19 +497,19 @@ examples: rcar_sound,dai { dai0 { playback = <&ssi5>, <&src5>; - capture = <&ssi6>; + capture = <&ssi6>; }; dai1 { playback = <&ssi3>; }; dai2 { - capture = <&ssi4>; + capture = <&ssi4>; }; dai3 { playback = <&ssi7>; }; dai4 { - capture = <&ssi8>; + capture = <&ssi8>; }; }; @@ -523,7 +523,7 @@ examples: frame-master = <&rsnd_endpoint0>; playback = <&ssi0>, <&src0>, <&dvc0>; - capture = <&ssi1>, <&src1>, <&dvc1>; + capture = <&ssi1>, <&src1>, <&dvc1>; }; }; }; diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml index a7cc9aa34468..4783e6dbb5c4 100644 --- a/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml @@ -90,7 +90,7 @@ examples: ldoin-supply = <&reg_3v3>; clocks = <&clks 201>; clock-names = "mclk"; - aic32x4-gpio-func= < + aic32x4-gpio-func = < 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ -- 2.34.1

2 1

[PATCH] ASoC: Intel: sof_sdw: Always register the HDMI dai links
by Peter Ujfalusi 24 Nov '23

24 Nov '23

The topology files for SDW devices require HDMI dai links to be present and this is granted under normal conditions but in case of special use cases the display (i915) driver might not be enabled due to deny-listing, booting with nomodeset or just not compiled at all. This should not block the non HDMI audio to be usable so register the dai links unconditionally. The code has been prepared for this and in case of no HDMI audio the link is created with dummy codec. Closes: https://github.com/thesofproject/linux/issues/4594 Closes: https://github.com/thesofproject/linux/issues/4648 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 3312ad8a563b..4e4284729773 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1546,7 +1546,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) { struct device *dev = card->dev; struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev); - int sdw_be_num = 0, ssp_num = 0, dmic_num = 0, hdmi_num = 0, bt_num = 0; + int sdw_be_num = 0, ssp_num = 0, dmic_num = 0, bt_num = 0; struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; const struct snd_soc_acpi_link_adr *adr_link = mach_params->links; @@ -1564,6 +1564,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) char *codec_name, *codec_dai_name; int i, j, be_id = 0; int codec_index; + int hdmi_num; int ret; ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num); @@ -1584,14 +1585,13 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) ssp_num = hweight_long(ssp_mask); } - if (mach_params->codec_mask & IDISP_CODEC_MASK) { + if (mach_params->codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) - hdmi_num = SOF_TGL_HDMI_COUNT; - else - hdmi_num = SOF_PRE_TGL_HDMI_COUNT; - } + if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) + hdmi_num = SOF_TGL_HDMI_COUNT; + else + hdmi_num = SOF_PRE_TGL_HDMI_COUNT; /* enable dmic01 & dmic16k */ if (sof_sdw_quirk & SOF_SDW_PCH_DMIC || mach_params->dmic_num) @@ -1601,7 +1601,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) bt_num = 1; dev_dbg(dev, "sdw %d, ssp %d, dmic %d, hdmi %d, bt: %d\n", - sdw_be_num, ssp_num, dmic_num, hdmi_num, bt_num); + sdw_be_num, ssp_num, dmic_num, + ctx->hdmi.idisp_codec ? hdmi_num : 0, bt_num); /* allocate BE dailinks */ num_links = sdw_be_num + ssp_num + dmic_num + hdmi_num + bt_num; -- 2.42.1

2 1

[PATCH] ASoC: Intel: skl_hda_dsp_generic: Drop HDMI routes when HDMI is not available
by Peter Ujfalusi 24 Nov '23

24 Nov '23

When the HDMI is not present due to disabled display support we will use dummy codec and the HDMI routes will refer to non existent DAPM widgets. Trim the route list from the HDMI routes to be able to probe the card even if the HDMI dais are not registered. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 6c6ef63cd5d9..6e172719c979 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -154,6 +154,8 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) card->dapm_widgets = skl_hda_widgets; card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); if (!ctx->idisp_codec) { + card->dapm_routes = &skl_hda_map[IDISP_ROUTE_COUNT]; + num_route -= IDISP_ROUTE_COUNT; for (i = 0; i < IDISP_DAI_COUNT; i++) { skl_hda_be_dai_links[i].codecs = &snd_soc_dummy_dlc; skl_hda_be_dai_links[i].num_codecs = 1; -- 2.42.1

2 1

[PATCH v3 0/2] Introduce playback-only/capture-only DAI link flags
by Daniel Baluta 24 Nov '23

24 Nov '23

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries allows users to specify a link only direction with audio-graph-card2. Changes since v1 - https://lore.kernel.org/lkml/20230801082433.548206-1-daniel.baluta@oss.nxp.… - used audio-graph-card2 instead of simple-card as we are deprecting simple-card. Changes since v2: - Fix device tree binding error reported by Rob Daniel Baluta (2): ASoC: audio-graph-card2: Introduce playback-only/capture only DAI link flags ASoC: dt-bindings: audio-graph-port: Document new DAI link flags playback-only/capture-only .../bindings/sound/audio-graph-port.yaml | 6 ++++++ include/sound/simple_card_utils.h | 3 +++ sound/soc/generic/audio-graph-card2.c | 6 ++++++ sound/soc/generic/simple-card-utils.c | 19 +++++++++++++++++++ 4 files changed, 34 insertions(+) -- 2.25.1

2 3

pcm: clarify documentation of poll descriptor usage
by GitHub pull_request - opened 24 Nov '23

24 Nov '23

alsa-project/alsa-lib pull request #370 was opened from z-s-e: This is based on my understanding of the intended behavior and the test/pcm.c example code. There needs to be more clarifiaction regarding the exact semantics of the value of the revents output parameter of snd_pcm_poll_descriptors_revents, since there are events that do not necessarily correspond to POLLIN or POLLOUT (such as period events), but I believe this is a lot less obvious and needs confirmation first. Request URL : https://github.com/alsa-project/alsa-lib/pull/370 Patch URL : https://github.com/alsa-project/alsa-lib/pull/370.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[RESEND] ASoC: cs43130: Allow configuration of bit clock and frame inversion
by Maciej Strozek 24 Nov '23

24 Nov '23

Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- Resending patch which previously broke x86 allmodconfig [1], now rebased on for-next branch [1]: https://lore.kernel.org/all/ZVfLMIbCpShVAYBf@finisterre.sirena.org.uk/ sound/soc/codecs/cs43130.c | 42 ++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/cs43130.h | 1 + 2 files changed, 41 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 4f16baf4eafb..0f3ead84665f 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -620,6 +620,27 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, return -EINVAL; } + switch (cs43130->dais[dai_id].dai_invert) { + case SND_SOC_DAIFMT_NB_NF: + sclk_edge = 1; + lrck_edge = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + sclk_edge = 0; + lrck_edge = 0; + break; + case SND_SOC_DAIFMT_NB_IF: + sclk_edge = 1; + lrck_edge = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + sclk_edge = 0; + lrck_edge = 1; + break; + default: + return -EINVAL; + } + switch (cs43130->dais[dai_id].dai_mode) { case SND_SOC_DAIFMT_CBS_CFS: dai_mode_val = 0; @@ -632,8 +653,6 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, } frm_size = bitwidth_sclk * params_channels(params); - sclk_edge = 1; - lrck_edge = 0; loc_ch1 = 0; loc_ch2 = bitwidth_sclk * (params_channels(params) - 1); @@ -1516,6 +1535,25 @@ static int cs43130_pcm_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_NB_NF; + break; + case SND_SOC_DAIFMT_IB_NF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_IB_NF; + break; + case SND_SOC_DAIFMT_NB_IF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_NB_IF; + break; + case SND_SOC_DAIFMT_IB_IF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_IB_IF; + break; + default: + dev_err(cs43130->dev, "Unsupported invert mode 0x%x\n", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_I2S; diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h index 694286b78d03..dbdb5b262f1b 100644 --- a/sound/soc/codecs/cs43130.h +++ b/sound/soc/codecs/cs43130.h @@ -497,6 +497,7 @@ struct cs43130_dai { unsigned int sclk; unsigned int dai_format; unsigned int dai_mode; + unsigned int dai_invert; }; struct cs43130_private { -- 2.34.1

3 2

[PATCH] soundwire: qcom: set controller id to hw master id
by srinivas.kandagatla＠linaro.org 24 Nov '23

24 Nov '23

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Qualcomm Soundwire Controllers IP version after 1.3 have a dedicated master id register which will provide a unique id value for each controller instance. Use this value instead of artificially generated value from idr. Versions 1.3 and below only have one instance of soundwire controller which does no have this register, so let them use value from idr. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- This patch depends on "[RFC PATCH 0/2] soundwire: introduce controller ID" https://lore.kernel.org/lkml/4815c8d6-635d-415c-9e02-4e39e92a3d6d@sirena.or… patchset. drivers/soundwire/qcom.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 8e027eee8b73..48291fbaf674 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1624,9 +1624,13 @@ static int qcom_swrm_probe(struct platform_device *pdev) } } - /* FIXME: is there a DT-defined value to use ? */ ctrl->bus.controller_id = -1; + if (ctrl->version > SWRM_VERSION_1_3_0) { + ctrl->reg_read(ctrl, SWRM_COMP_MASTER_ID, &val); + ctrl->bus.controller_id = val; + } + ret = sdw_bus_master_add(&ctrl->bus, dev, dev->fwnode); if (ret) { dev_err(dev, "Failed to register Soundwire controller (%d)\n", -- 2.42.0

3 2

[RFC PATCH 0/2] soundwire: introduce controller ID
by Pierre-Louis Bossart 24 Nov '23

24 Nov '23

This patchset is an alternate proposal to the solution suggested in [1], which breaks Intel machine drivers. The only difference is to use a known controller ID instead of an IDA, which wouldn't work with the hard-coded device name. This patchset was tested on Intel and AMD platforms, testing on Qualcomm platforms is required - hence the RFC status. [1] https://lore.kernel.org/alsa-devel/20231004130243.493617-1-krzysztof.kozlow… Krzysztof Kozlowski (1): soundwire: fix initializing sysfs for same devices on different buses Pierre-Louis Bossart (1): soundwire: bus: introduce controller_id drivers/soundwire/amd_manager.c | 8 ++++++++ drivers/soundwire/bus.c | 4 ++++ drivers/soundwire/debugfs.c | 2 +- drivers/soundwire/intel_auxdevice.c | 3 +++ drivers/soundwire/master.c | 2 +- drivers/soundwire/qcom.c | 3 +++ drivers/soundwire/slave.c | 12 ++++++------ include/linux/soundwire/sdw.h | 4 +++- sound/soc/intel/boards/sof_sdw.c | 4 ++-- 9 files changed, 31 insertions(+), 11 deletions(-) -- 2.39.2

5 8

[PATCH] ASoC: fsl_xcvr: refine the requested phy clock frequency
by Shengjiu Wang 23 Nov '23

23 Nov '23

As the input phy clock frequency will divided by 2 by default on i.MX8MP with the implementation of clk-imx8mp-audiomix driver, So the requested frequency need to be updated. The relation of phy clock is: sai_pll_ref_sel sai_pll sai_pll_bypass sai_pll_out sai_pll_out_div2 earc_phy_cg Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 77f8e2394bf9..f0fb33d719c2 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -358,7 +358,7 @@ static int fsl_xcvr_en_aud_pll(struct fsl_xcvr *xcvr, u32 freq) struct device *dev = &xcvr->pdev->dev; int ret; - freq = xcvr->soc_data->spdif_only ? freq / 10 : freq; + freq = xcvr->soc_data->spdif_only ? freq / 5 : freq; clk_disable_unprepare(xcvr->phy_clk); ret = clk_set_rate(xcvr->phy_clk, freq); if (ret < 0) { @@ -409,7 +409,7 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 m_ctl = 0, v_ctl = 0; u32 r = substream->runtime->rate, ch = substream->runtime->channels; - u32 fout = 32 * r * ch * 10 * 2; + u32 fout = 32 * r * ch * 10; int ret = 0; switch (xcvr->mode) { -- 2.34.1

3 2

[PATCH v5] ASoC: cs43130: Allow driver to work without IRQ connection
by Maciej Strozek 23 Nov '23

23 Nov '23

Add a polling mechanism that will keep the driver operational even in absence of physical IRQ connection. If IRQ line is detected, the driver will continue working as usual, in case of missing IRQ line it will fallback to the polling mechanism introduced in this change. This will support users which choose not to connect an IRQ line as it is not critical to part's operation. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- V4 -> V5: Do not use hpload interrupt with the new polling function V3 -> V4: - Removed unnecesary else and indentation - Changed cs43130_wait_for_completion return codes and updated callers accordingly - Changed approach for checking for presence of IRQ connection: see if client->irq equals 0 which seems to be a standard way of checking it used by other drivers (instead of inspecting devm_request_threaded_irq error code) V2 -> V3: Amended changelog message and subject line V1 -> V2: Add changelog message sound/soc/codecs/cs43130.c | 77 ++++++++++++++++++++++++++------------ sound/soc/codecs/cs43130.h | 1 + 2 files changed, 55 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 02e7f45b9208..0f3ead84665f 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -326,6 +326,34 @@ static int cs43130_set_pll(struct snd_soc_component *component, int pll_id, int return ret; } +static int cs43130_wait_for_completion(struct cs43130_private *cs43130, struct completion *to_poll, + int time) +{ + int stickies, offset, flag, ret; + + if (cs43130->has_irq_line) { + ret = wait_for_completion_timeout(to_poll, msecs_to_jiffies(time)); + if (ret == 0) + return -ETIMEDOUT; + else + return 0; // Discard number of jiffies left till timeout and return success + } + + if (to_poll == &cs43130->xtal_rdy) { + offset = 0; + flag = CS43130_XTAL_RDY_INT; + } else if (to_poll == &cs43130->pll_rdy) { + offset = 0; + flag = CS43130_PLL_RDY_INT; + } else { + return -EINVAL; + } + + return regmap_read_poll_timeout(cs43130->regmap, CS43130_INT_STATUS_1 + offset, + stickies, (stickies & flag), + 1000, time * 1000); +} + static int cs43130_change_clksrc(struct snd_soc_component *component, enum cs43130_mclk_src_sel src) { @@ -364,14 +392,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for XTAL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for XTAL_READY interrupt: %d\n", ret); + return ret; } } @@ -400,14 +427,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for XTAL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for XTAL_READY interrupt: %d\n", ret); + return ret; } } @@ -416,14 +442,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_PLL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_PLL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->pll_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->pll_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_PLL_RDY_INT_MASK, 1 << CS43130_PLL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for PLL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for PLL_READY interrupt: %d\n", ret); + return ret; } regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, @@ -2053,7 +2078,7 @@ static int cs43130_hpload_proc(struct cs43130_private *cs43130, regmap_read(cs43130->regmap, CS43130_INT_MASK_4, &msk); if (!ret) { dev_err(cs43130->dev, "Timeout waiting for HPLOAD interrupt\n"); - return -1; + return -ETIMEDOUT; } dev_dbg(cs43130->dev, "HP load stat: %x, INT_MASK_4: %x\n", @@ -2548,13 +2573,19 @@ static int cs43130_i2c_probe(struct i2c_client *client) init_completion(&cs43130->pll_rdy); init_completion(&cs43130->hpload_evt); - ret = devm_request_threaded_irq(cs43130->dev, client->irq, - NULL, cs43130_irq_thread, - IRQF_ONESHOT | IRQF_TRIGGER_LOW, - "cs43130", cs43130); - if (ret != 0) { - dev_err(cs43130->dev, "Failed to request IRQ: %d\n", ret); - goto err; + if (!client->irq) { + dev_dbg(cs43130->dev, "IRQ not found, will poll instead\n"); + cs43130->has_irq_line = 0; + } else { + ret = devm_request_threaded_irq(cs43130->dev, client->irq, + NULL, cs43130_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs43130", cs43130); + if (ret != 0) { + dev_err(cs43130->dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + cs43130->has_irq_line = 1; } cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO; diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h index d3f595bbd3ba..dbdb5b262f1b 100644 --- a/sound/soc/codecs/cs43130.h +++ b/sound/soc/codecs/cs43130.h @@ -508,6 +508,7 @@ struct cs43130_private { struct gpio_desc *reset_gpio; unsigned int dev_id; /* codec device ID */ int xtal_ibias; + bool has_irq_line; /* shared by both DAIs */ struct mutex clk_mutex; -- 2.34.1

3 4

[PATCH v2 0/2] Introduce playback-only/capture-only DAI link
by Daniel Baluta 23 Nov '23

23 Nov '23

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries allows users to specify a link only direction with audio-graph-card2. Changes since v1 - https://lore.kernel.org/lkml/20230801082433.548206-1-daniel.baluta@oss.nxp.… - used audio-graph-card2 instead of simple-card as we are deprecting simple-card. Daniel Baluta (2): ASoC: audio-graph-card2: Introduce playback-only/capture only DAI link flags ASoC: dt-bindings: audio-graph-port: Document new DAI link flags playback-only/capture-only .../bindings/sound/audio-graph-port.yaml | 6 ++++++ include/sound/simple_card_utils.h | 3 +++ sound/soc/generic/audio-graph-card2.c | 6 ++++++ sound/soc/generic/simple-card-utils.c | 19 +++++++++++++++++++ 4 files changed, 34 insertions(+) -- 2.25.1

2 3

[PATCH v7] dt-bindings: arm: mediatek: convert audsys and mt2701-afe-pcm to yaml
by Eugen Hristev 23 Nov '23

23 Nov '23

Convert the mediatek,audsys binding to YAML, together with the associated binding bindings/sound/mt2701-afe-pcm.yaml . Signed-off-by: Eugen Hristev <eugen.hristev(a)collabora.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- Changes in v7: - correctly renamed mt2701-audio.yaml to mediatek,mt2701-audio.yaml Changes in v6: - renamed mt2701-audio.yaml to mediatek,mt2701-audio.yaml - add Rb Rob Herring Changes in v5: - removed example for mt2701-audio - renamed mt2701-afe-pcm to mt2701-audio - added clock cells as required, removed reg from required, in audsys binding - removed reference comment in mt2701-audio - removed assigned clocks from mt2701-audio Changes in v4: - fix error reported by Rob's bot : ./Documentation/devicetree/bindings/sound/mt2701-afe-pcm.yaml:11:4: [error] missing starting space in comment (comments) Changes in v3: - not added Rb Conor Dooley since the patch was changed in a big essence - As per review by Krzysztof, also convert the mt2701-afe-pcm and reference the correct schema in the audsys binding. Changes in v2: - remove comment reference to inexistent binding .../bindings/arm/mediatek/mediatek,audsys.txt | 39 ----- .../arm/mediatek/mediatek,audsys.yaml | 153 ++++++++++++++++++ .../bindings/sound/mediatek,mt2701-audio.yaml | 117 ++++++++++++++ .../bindings/sound/mt2701-afe-pcm.txt | 146 ----------------- 4 files changed, 270 insertions(+), 185 deletions(-) delete mode 100644 Documentation/devicetree/bindings/arm/mediatek/mediatek,audsys.txt create mode 100644 Documentation/devicetree/bindings/arm/mediatek/mediatek,audsys.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt2701-audio.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt diff --git a/Documentation/devicetree/bindings/arm/mediatek/mediatek,audsys.txt b/Documentation/devicetree/bindings/arm/mediatek/mediatek,audsys.txt deleted file mode 100644 index 699776be1dd3..000000000000 --- a/Documentation/devicetree/bindings/arm/mediatek/mediatek,audsys.txt +++ /dev/null @@ -1,39 +0,0 @@ -MediaTek AUDSYS controller -============================ - -The MediaTek AUDSYS controller provides various clocks to the system. - -Required Properties: - -- compatible: Should be one of: - - "mediatek,mt2701-audsys", "syscon" - - "mediatek,mt6765-audsys", "syscon" - - "mediatek,mt6779-audio", "syscon" - - "mediatek,mt7622-audsys", "syscon" - - "mediatek,mt7623-audsys", "mediatek,mt2701-audsys", "syscon" - - "mediatek,mt8167-audiosys", "syscon" - - "mediatek,mt8183-audiosys", "syscon" - - "mediatek,mt8192-audsys", "syscon" - - "mediatek,mt8516-audsys", "syscon" -- #clock-cells: Must be 1 - -The AUDSYS controller uses the common clk binding from -Documentation/devicetree/bindings/clock/clock-bindings.txt -The available clocks are defined in dt-bindings/clock/mt*-clk.h. - -Required sub-nodes: -------- -For common binding part and usage, refer to -../sonud/mt2701-afe-pcm.txt. - -Example: - - audsys: clock-controller@11220000 { - compatible = "mediatek,mt7622-audsys", "syscon"; - reg = <0 0x11220000 0 0x2000>; - #clock-cells = <1>; - - afe: audio-controller { - ... - }; - }; diff --git a/Documentation/devicetree/bindings/arm/mediatek/mediatek,audsys.yaml b/Documentation/devicetree/bindings/arm/mediatek/mediatek,audsys.yaml new file mode 100644 index 000000000000..45d4a6620041 --- /dev/null +++ b/Documentation/devicetree/bindings/arm/mediatek/mediatek,audsys.yaml @@ -0,0 +1,153 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/arm/mediatek/mediatek,audsys.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek AUDSYS controller + +maintainers: + - Eugen Hristev <eugen.hristev(a)collabora.com> + +description: + The MediaTek AUDSYS controller provides various clocks to the system. + +properties: + compatible: + oneOf: + - items: + - enum: + - mediatek,mt2701-audsys + - mediatek,mt6765-audsys + - mediatek,mt6779-audsys + - mediatek,mt7622-audsys + - mediatek,mt8167-audsys + - mediatek,mt8173-audsys + - mediatek,mt8183-audsys + - mediatek,mt8186-audsys + - mediatek,mt8192-audsys + - mediatek,mt8516-audsys + - const: syscon + - items: + # Special case for mt7623 for backward compatibility + - const: mediatek,mt7623-audsys + - const: mediatek,mt2701-audsys + - const: syscon + + reg: + maxItems: 1 + + '#clock-cells': + const: 1 + + audio-controller: + $ref: /schemas/sound/mediatek,mt2701-audio.yaml# + type: object + +required: + - compatible + - '#clock-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt2701-power.h> + #include <dt-bindings/clock/mt2701-clk.h> + soc { + #address-cells = <2>; + #size-cells = <2>; + audsys: clock-controller@11220000 { + compatible = "mediatek,mt7622-audsys", "syscon"; + reg = <0 0x11220000 0 0x2000>; + #clock-cells = <1>; + + afe: audio-controller { + compatible = "mediatek,mt2701-audio"; + interrupts = <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>, + <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>; + interrupt-names = "afe", "asys"; + power-domains = <&scpsys MT2701_POWER_DOMAIN_IFR_MSC>; + + clocks = <&infracfg CLK_INFRA_AUDIO>, + <&topckgen CLK_TOP_AUD_MUX1_SEL>, + <&topckgen CLK_TOP_AUD_MUX2_SEL>, + <&topckgen CLK_TOP_AUD_48K_TIMING>, + <&topckgen CLK_TOP_AUD_44K_TIMING>, + <&topckgen CLK_TOP_AUD_K1_SRC_SEL>, + <&topckgen CLK_TOP_AUD_K2_SRC_SEL>, + <&topckgen CLK_TOP_AUD_K3_SRC_SEL>, + <&topckgen CLK_TOP_AUD_K4_SRC_SEL>, + <&topckgen CLK_TOP_AUD_K1_SRC_DIV>, + <&topckgen CLK_TOP_AUD_K2_SRC_DIV>, + <&topckgen CLK_TOP_AUD_K3_SRC_DIV>, + <&topckgen CLK_TOP_AUD_K4_SRC_DIV>, + <&topckgen CLK_TOP_AUD_I2S1_MCLK>, + <&topckgen CLK_TOP_AUD_I2S2_MCLK>, + <&topckgen CLK_TOP_AUD_I2S3_MCLK>, + <&topckgen CLK_TOP_AUD_I2S4_MCLK>, + <&audsys CLK_AUD_I2SO1>, + <&audsys CLK_AUD_I2SO2>, + <&audsys CLK_AUD_I2SO3>, + <&audsys CLK_AUD_I2SO4>, + <&audsys CLK_AUD_I2SIN1>, + <&audsys CLK_AUD_I2SIN2>, + <&audsys CLK_AUD_I2SIN3>, + <&audsys CLK_AUD_I2SIN4>, + <&audsys CLK_AUD_ASRCO1>, + <&audsys CLK_AUD_ASRCO2>, + <&audsys CLK_AUD_ASRCO3>, + <&audsys CLK_AUD_ASRCO4>, + <&audsys CLK_AUD_AFE>, + <&audsys CLK_AUD_AFE_CONN>, + <&audsys CLK_AUD_A1SYS>, + <&audsys CLK_AUD_A2SYS>, + <&audsys CLK_AUD_AFE_MRGIF>; + + clock-names = "infra_sys_audio_clk", + "top_audio_mux1_sel", + "top_audio_mux2_sel", + "top_audio_a1sys_hp", + "top_audio_a2sys_hp", + "i2s0_src_sel", + "i2s1_src_sel", + "i2s2_src_sel", + "i2s3_src_sel", + "i2s0_src_div", + "i2s1_src_div", + "i2s2_src_div", + "i2s3_src_div", + "i2s0_mclk_en", + "i2s1_mclk_en", + "i2s2_mclk_en", + "i2s3_mclk_en", + "i2so0_hop_ck", + "i2so1_hop_ck", + "i2so2_hop_ck", + "i2so3_hop_ck", + "i2si0_hop_ck", + "i2si1_hop_ck", + "i2si2_hop_ck", + "i2si3_hop_ck", + "asrc0_out_ck", + "asrc1_out_ck", + "asrc2_out_ck", + "asrc3_out_ck", + "audio_afe_pd", + "audio_afe_conn_pd", + "audio_a1sys_pd", + "audio_a2sys_pd", + "audio_mrgif_pd"; + + assigned-clocks = <&topckgen CLK_TOP_AUD_MUX1_SEL>, + <&topckgen CLK_TOP_AUD_MUX2_SEL>, + <&topckgen CLK_TOP_AUD_MUX1_DIV>, + <&topckgen CLK_TOP_AUD_MUX2_DIV>; + assigned-clock-parents = <&topckgen CLK_TOP_AUD1PLL_98M>, + <&topckgen CLK_TOP_AUD2PLL_90M>; + assigned-clock-rates = <0>, <0>, <49152000>, <45158400>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt2701-audio.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt2701-audio.yaml new file mode 100644 index 000000000000..83fd7bb9cadb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt2701-audio.yaml @@ -0,0 +1,117 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt2701-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek Audio Front End (AFE) PCM controller for mt2701 + +description: + The AFE PCM node must be a subnode of the MediaTek audsys device tree node. + +maintainers: + - Eugen Hristev <eugen.hristev(a)collabora.com> + +properties: + compatible: + enum: + - mediatek,mt2701-audio + - mediatek,mt7622-audio + + interrupts: + items: + - description: AFE interrupt + - description: ASYS interrupt + + interrupt-names: + items: + - const: afe + - const: asys + + power-domains: + maxItems: 1 + + clocks: + items: + - description: audio infra sys clock + - description: top audio mux 1 + - description: top audio mux 2 + - description: top audio sys a1 clock + - description: top audio sys a2 clock + - description: i2s0 source selection + - description: i2s1 source selection + - description: i2s2 source selection + - description: i2s3 source selection + - description: i2s0 source divider + - description: i2s1 source divider + - description: i2s2 source divider + - description: i2s3 source divider + - description: i2s0 master clock + - description: i2s1 master clock + - description: i2s2 master clock + - description: i2s3 master clock + - description: i2so0 hopping clock + - description: i2so1 hopping clock + - description: i2so2 hopping clock + - description: i2so3 hopping clock + - description: i2si0 hopping clock + - description: i2si1 hopping clock + - description: i2si2 hopping clock + - description: i2si3 hopping clock + - description: asrc0 output clock + - description: asrc1 output clock + - description: asrc2 output clock + - description: asrc3 output clock + - description: audio front end pd clock + - description: audio front end conn pd clock + - description: top audio a1 sys pd + - description: top audio a2 sys pd + - description: audio merge interface pd + + clock-names: + items: + - const: infra_sys_audio_clk + - const: top_audio_mux1_sel + - const: top_audio_mux2_sel + - const: top_audio_a1sys_hp + - const: top_audio_a2sys_hp + - const: i2s0_src_sel + - const: i2s1_src_sel + - const: i2s2_src_sel + - const: i2s3_src_sel + - const: i2s0_src_div + - const: i2s1_src_div + - const: i2s2_src_div + - const: i2s3_src_div + - const: i2s0_mclk_en + - const: i2s1_mclk_en + - const: i2s2_mclk_en + - const: i2s3_mclk_en + - const: i2so0_hop_ck + - const: i2so1_hop_ck + - const: i2so2_hop_ck + - const: i2so3_hop_ck + - const: i2si0_hop_ck + - const: i2si1_hop_ck + - const: i2si2_hop_ck + - const: i2si3_hop_ck + - const: asrc0_out_ck + - const: asrc1_out_ck + - const: asrc2_out_ck + - const: asrc3_out_ck + - const: audio_afe_pd + - const: audio_afe_conn_pd + - const: audio_a1sys_pd + - const: audio_a2sys_pd + - const: audio_mrgif_pd + +required: + - compatible + - interrupts + - interrupt-names + - power-domains + - clocks + - clock-names + +additionalProperties: false + diff --git a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt deleted file mode 100644 index f548e6a58240..000000000000 --- a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt +++ /dev/null @@ -1,146 +0,0 @@ -Mediatek AFE PCM controller for mt2701 - -Required properties: -- compatible: should be one of the following. - - "mediatek,mt2701-audio" - - "mediatek,mt7622-audio" -- interrupts: should contain AFE and ASYS interrupts -- interrupt-names: should be "afe" and "asys" -- power-domains: should define the power domain -- clocks: Must contain an entry for each entry in clock-names - See ../clocks/clock-bindings.txt for details -- clock-names: should have these clock names: - "infra_sys_audio_clk", - "top_audio_mux1_sel", - "top_audio_mux2_sel", - "top_audio_a1sys_hp", - "top_audio_a2sys_hp", - "i2s0_src_sel", - "i2s1_src_sel", - "i2s2_src_sel", - "i2s3_src_sel", - "i2s0_src_div", - "i2s1_src_div", - "i2s2_src_div", - "i2s3_src_div", - "i2s0_mclk_en", - "i2s1_mclk_en", - "i2s2_mclk_en", - "i2s3_mclk_en", - "i2so0_hop_ck", - "i2so1_hop_ck", - "i2so2_hop_ck", - "i2so3_hop_ck", - "i2si0_hop_ck", - "i2si1_hop_ck", - "i2si2_hop_ck", - "i2si3_hop_ck", - "asrc0_out_ck", - "asrc1_out_ck", - "asrc2_out_ck", - "asrc3_out_ck", - "audio_afe_pd", - "audio_afe_conn_pd", - "audio_a1sys_pd", - "audio_a2sys_pd", - "audio_mrgif_pd"; -- assigned-clocks: list of input clocks and dividers for the audio system. - See ../clocks/clock-bindings.txt for details. -- assigned-clocks-parents: parent of input clocks of assigned clocks. -- assigned-clock-rates: list of clock frequencies of assigned clocks. - -Must be a subnode of MediaTek audsys device tree node. -See ../arm/mediatek/mediatek,audsys.txt for details about the parent node. - -Example: - - audsys: audio-subsystem@11220000 { - compatible = "mediatek,mt2701-audsys", "syscon"; - ... - - afe: audio-controller { - compatible = "mediatek,mt2701-audio"; - interrupts = <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>, - <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>; - interrupt-names = "afe", "asys"; - power-domains = <&scpsys MT2701_POWER_DOMAIN_IFR_MSC>; - - clocks = <&infracfg CLK_INFRA_AUDIO>, - <&topckgen CLK_TOP_AUD_MUX1_SEL>, - <&topckgen CLK_TOP_AUD_MUX2_SEL>, - <&topckgen CLK_TOP_AUD_48K_TIMING>, - <&topckgen CLK_TOP_AUD_44K_TIMING>, - <&topckgen CLK_TOP_AUD_K1_SRC_SEL>, - <&topckgen CLK_TOP_AUD_K2_SRC_SEL>, - <&topckgen CLK_TOP_AUD_K3_SRC_SEL>, - <&topckgen CLK_TOP_AUD_K4_SRC_SEL>, - <&topckgen CLK_TOP_AUD_K1_SRC_DIV>, - <&topckgen CLK_TOP_AUD_K2_SRC_DIV>, - <&topckgen CLK_TOP_AUD_K3_SRC_DIV>, - <&topckgen CLK_TOP_AUD_K4_SRC_DIV>, - <&topckgen CLK_TOP_AUD_I2S1_MCLK>, - <&topckgen CLK_TOP_AUD_I2S2_MCLK>, - <&topckgen CLK_TOP_AUD_I2S3_MCLK>, - <&topckgen CLK_TOP_AUD_I2S4_MCLK>, - <&audsys CLK_AUD_I2SO1>, - <&audsys CLK_AUD_I2SO2>, - <&audsys CLK_AUD_I2SO3>, - <&audsys CLK_AUD_I2SO4>, - <&audsys CLK_AUD_I2SIN1>, - <&audsys CLK_AUD_I2SIN2>, - <&audsys CLK_AUD_I2SIN3>, - <&audsys CLK_AUD_I2SIN4>, - <&audsys CLK_AUD_ASRCO1>, - <&audsys CLK_AUD_ASRCO2>, - <&audsys CLK_AUD_ASRCO3>, - <&audsys CLK_AUD_ASRCO4>, - <&audsys CLK_AUD_AFE>, - <&audsys CLK_AUD_AFE_CONN>, - <&audsys CLK_AUD_A1SYS>, - <&audsys CLK_AUD_A2SYS>, - <&audsys CLK_AUD_AFE_MRGIF>; - - clock-names = "infra_sys_audio_clk", - "top_audio_mux1_sel", - "top_audio_mux2_sel", - "top_audio_a1sys_hp", - "top_audio_a2sys_hp", - "i2s0_src_sel", - "i2s1_src_sel", - "i2s2_src_sel", - "i2s3_src_sel", - "i2s0_src_div", - "i2s1_src_div", - "i2s2_src_div", - "i2s3_src_div", - "i2s0_mclk_en", - "i2s1_mclk_en", - "i2s2_mclk_en", - "i2s3_mclk_en", - "i2so0_hop_ck", - "i2so1_hop_ck", - "i2so2_hop_ck", - "i2so3_hop_ck", - "i2si0_hop_ck", - "i2si1_hop_ck", - "i2si2_hop_ck", - "i2si3_hop_ck", - "asrc0_out_ck", - "asrc1_out_ck", - "asrc2_out_ck", - "asrc3_out_ck", - "audio_afe_pd", - "audio_afe_conn_pd", - "audio_a1sys_pd", - "audio_a2sys_pd", - "audio_mrgif_pd"; - - assigned-clocks = <&topckgen CLK_TOP_AUD_MUX1_SEL>, - <&topckgen CLK_TOP_AUD_MUX2_SEL>, - <&topckgen CLK_TOP_AUD_MUX1_DIV>, - <&topckgen CLK_TOP_AUD_MUX2_DIV>; - assigned-clock-parents = <&topckgen CLK_TOP_AUD1PLL_98M>, - <&topckgen CLK_TOP_AUD2PLL_90M>; - assigned-clock-rates = <0>, <0>, <49152000>, <45158400>; - }; - }; -- 2.34.1

2 1

[PATCH] ASoC: rt5650: add mutex to avoid the jack detection failure
by shumingf＠realtek.com 23 Nov '23

23 Nov '23

From: Shuming Fan <shumingf(a)realtek.com> This patch adds the jd_mutex to protect the jack detection control flow. And only the headset type could check the button status. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5645.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7938b52d741d..a0d01d71d8b5 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -448,6 +448,7 @@ struct rt5645_priv { struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; struct rt5645_eq_param_s *eq_param; struct timer_list btn_check_timer; + struct mutex jd_mutex; int codec_type; int sysclk; @@ -3193,6 +3194,8 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse rt5645_enable_push_button_irq(component, true); } } else { + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(component, false); snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; @@ -3295,6 +3298,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (!rt5645->component) return; + mutex_lock(&rt5645->jd_mutex); + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { @@ -3321,7 +3326,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (!val && (rt5645->jack_type == 0)) { /* jack in */ report = rt5645_jack_detect(rt5645->component, 1); - } else if (!val && rt5645->jack_type != 0) { + } else if (!val && rt5645->jack_type == SND_JACK_HEADSET) { /* for push button and jack out */ btn_type = 0; if (snd_soc_component_read(rt5645->component, RT5645_INT_IRQ_ST) & 0x4) { @@ -3377,6 +3382,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) rt5645_jack_detect(rt5645->component, 0); } + mutex_unlock(&rt5645->jd_mutex); + snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE); snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); if (rt5645->en_button_func) @@ -4150,6 +4157,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c) } timer_setup(&rt5645->btn_check_timer, rt5645_btn_check_callback, 0); + mutex_init(&rt5645->jd_mutex); INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); -- 2.34.1

3 3

[PATCH 1/2] ASoC: fsl: mpc8610_hpcd: Remove unused driver
by Michael Ellerman 23 Nov '23

23 Nov '23

The mpc8610_hpcd.c driver depends on CONFIG_MPC8610_HPCD which was removed in commit 248667f8bbde ("powerpc: drop HPCD/MPC8610 evaluation platform support"). That makes the driver unbuildable and unusable, so remove it. Depends-on: 248667f8bbde ("powerpc: drop HPCD/MPC8610 evaluation platform support") Signed-off-by: Michael Ellerman <mpe(a)ellerman.id.au> --- MAINTAINERS | 1 - sound/soc/fsl/Kconfig | 13 - sound/soc/fsl/Makefile | 4 - sound/soc/fsl/mpc8610_hpcd.c | 451 ----------------------------------- 4 files changed, 469 deletions(-) delete mode 100644 sound/soc/fsl/mpc8610_hpcd.c diff --git a/MAINTAINERS b/MAINTAINERS index ea790149af79..27b2f5bb2061 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -8581,7 +8581,6 @@ L: linuxppc-dev(a)lists.ozlabs.org S: Maintained F: sound/soc/fsl/fsl* F: sound/soc/fsl/imx* -F: sound/soc/fsl/mpc8610_hpcd.c FREESCALE SOC SOUND QMC DRIVER M: Herve Codina <herve.codina(a)bootlin.com> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 725c530a3636..bec2aa561930 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -183,19 +183,6 @@ config SND_SOC_POWERPC_QMC_AUDIO comment "SoC Audio support for Freescale PPC boards:" -config SND_SOC_MPC8610_HPCD - tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" - # I2C is necessary for the CS4270 driver - depends on MPC8610_HPCD && I2C - select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS - select SND_SOC_POWERPC_DMA - select SND_SOC_CS4270 - select SND_SOC_CS4270_VD33_ERRATA - default y if MPC8610_HPCD - help - Say Y if you want to enable audio on the Freescale MPC8610 HPCD. - config SND_SOC_P1022_DS tristate "ALSA SoC support for the Freescale P1022 DS board" # I2C is necessary for the WM8776 driver diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 8db7e97d0bd5..b45eda80c196 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -1,8 +1,4 @@ # SPDX-License-Identifier: GPL-2.0 -# MPC8610 HPCD Machine Support -snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o -obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o - # P1022 DS Machine Support snd-soc-p1022-ds-objs := p1022_ds.o obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c deleted file mode 100644 index 52fb9e7bcca4..000000000000 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ /dev/null @@ -1,451 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// Freescale MPC8610HPCD ALSA SoC Machine driver -// -// Author: Timur Tabi <timur(a)freescale.com> -// -// Copyright 2007-2010 Freescale Semiconductor, Inc. - -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/fsl/guts.h> -#include <linux/of.h> -#include <linux/of_address.h> -#include <linux/slab.h> -#include <sound/soc.h> - -#include "fsl_dma.h" -#include "fsl_ssi.h" -#include "fsl_utils.h" - -/* There's only one global utilities register */ -static phys_addr_t guts_phys; - -/** - * mpc8610_hpcd_data: machine-specific ASoC device data - * - * This structure contains data for a single sound platform device on an - * MPC8610 HPCD. Some of the data is taken from the device tree. - */ -struct mpc8610_hpcd_data { - struct snd_soc_dai_link dai[2]; - struct snd_soc_card card; - unsigned int dai_format; - unsigned int codec_clk_direction; - unsigned int cpu_clk_direction; - unsigned int clk_frequency; - unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ - unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ - unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ - char codec_dai_name[DAI_NAME_SIZE]; - char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ -}; - -/** - * mpc8610_hpcd_machine_probe: initialize the board - * - * This function is used to initialize the board-specific hardware. - * - * Here we program the DMACR and PMUXCR registers. - */ -static int mpc8610_hpcd_machine_probe(struct snd_soc_card *card) -{ - struct mpc8610_hpcd_data *machine_data = - container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; - - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); - if (!guts) { - dev_err(card->dev, "could not map global utilities\n"); - return -ENOMEM; - } - - /* Program the signal routing between the SSI and the DMA */ - guts_set_dmacr(guts, machine_data->dma_id[0], - machine_data->dma_channel_id[0], - CCSR_GUTS_DMACR_DEV_SSI); - guts_set_dmacr(guts, machine_data->dma_id[1], - machine_data->dma_channel_id[1], - CCSR_GUTS_DMACR_DEV_SSI); - - guts_set_pmuxcr_dma(guts, machine_data->dma_id[0], - machine_data->dma_channel_id[0], 0); - guts_set_pmuxcr_dma(guts, machine_data->dma_id[1], - machine_data->dma_channel_id[1], 0); - - switch (machine_data->ssi_id) { - case 0: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); - break; - case 1: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); - break; - } - - iounmap(guts); - - return 0; -} - -/** - * mpc8610_hpcd_startup: program the board with various hardware parameters - * - * This function takes board-specific information, like clock frequencies - * and serial data formats, and passes that information to the codec and - * transport drivers. - */ -static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct mpc8610_hpcd_data *machine_data = - container_of(rtd->card, struct mpc8610_hpcd_data, card); - struct device *dev = rtd->card->dev; - int ret = 0; - - /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(snd_soc_rtd_to_codec(rtd, 0), machine_data->dai_format); - if (ret < 0) { - dev_err(dev, "could not set codec driver audio format\n"); - return ret; - } - - /* - * Tell the codec driver what the MCLK frequency is, and whether it's - * a slave or master. - */ - ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_codec(rtd, 0), 0, - machine_data->clk_frequency, - machine_data->codec_clk_direction); - if (ret < 0) { - dev_err(dev, "could not set codec driver clock params\n"); - return ret; - } - - return 0; -} - -/** - * mpc8610_hpcd_machine_remove: Remove the sound device - * - * This function is called to remove the sound device for one SSI. We - * de-program the DMACR and PMUXCR register. - */ -static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card) -{ - struct mpc8610_hpcd_data *machine_data = - container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; - - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); - if (!guts) { - dev_err(card->dev, "could not map global utilities\n"); - return -ENOMEM; - } - - /* Restore the signal routing */ - - guts_set_dmacr(guts, machine_data->dma_id[0], - machine_data->dma_channel_id[0], 0); - guts_set_dmacr(guts, machine_data->dma_id[1], - machine_data->dma_channel_id[1], 0); - - switch (machine_data->ssi_id) { - case 0: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); - break; - case 1: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA); - break; - } - - iounmap(guts); - - return 0; -} - -/** - * mpc8610_hpcd_ops: ASoC machine driver operations - */ -static const struct snd_soc_ops mpc8610_hpcd_ops = { - .startup = mpc8610_hpcd_startup, -}; - -/** - * mpc8610_hpcd_probe: platform probe function for the machine driver - * - * Although this is a machine driver, the SSI node is the "master" node with - * respect to audio hardware connections. Therefore, we create a new ASoC - * device for each new SSI node that has a codec attached. - */ -static int mpc8610_hpcd_probe(struct platform_device *pdev) -{ - struct device *dev = pdev->dev.parent; - /* ssi_pdev is the platform device for the SSI node that probed us */ - struct platform_device *ssi_pdev = to_platform_device(dev); - struct device_node *np = ssi_pdev->dev.of_node; - struct device_node *codec_np = NULL; - struct mpc8610_hpcd_data *machine_data; - struct snd_soc_dai_link_component *comp; - int ret; - const char *sprop; - const u32 *iprop; - - /* Find the codec node for this SSI. */ - codec_np = of_parse_phandle(np, "codec-handle", 0); - if (!codec_np) { - dev_err(dev, "invalid codec node\n"); - return -EINVAL; - } - - machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); - if (!machine_data) { - ret = -ENOMEM; - goto error_alloc; - } - - comp = devm_kzalloc(&pdev->dev, 6 * sizeof(*comp), GFP_KERNEL); - if (!comp) { - ret = -ENOMEM; - goto error_alloc; - } - - machine_data->dai[0].cpus = &comp[0]; - machine_data->dai[0].codecs = &comp[1]; - machine_data->dai[0].platforms = &comp[2]; - - machine_data->dai[0].num_cpus = 1; - machine_data->dai[0].num_codecs = 1; - machine_data->dai[0].num_platforms = 1; - - machine_data->dai[1].cpus = &comp[3]; - machine_data->dai[1].codecs = &comp[4]; - machine_data->dai[1].platforms = &comp[5]; - - machine_data->dai[1].num_cpus = 1; - machine_data->dai[1].num_codecs = 1; - machine_data->dai[1].num_platforms = 1; - - machine_data->dai[0].cpus->dai_name = dev_name(&ssi_pdev->dev); - machine_data->dai[0].ops = &mpc8610_hpcd_ops; - - /* ASoC core can match codec with device node */ - machine_data->dai[0].codecs->of_node = codec_np; - - /* The DAI name from the codec (snd_soc_dai_driver.name) */ - machine_data->dai[0].codecs->dai_name = "cs4270-hifi"; - - /* We register two DAIs per SSI, one for playback and the other for - * capture. Currently, we only support codecs that have one DAI for - * both playback and capture. - */ - memcpy(&machine_data->dai[1], &machine_data->dai[0], - sizeof(struct snd_soc_dai_link)); - - /* Get the device ID */ - iprop = of_get_property(np, "cell-index", NULL); - if (!iprop) { - dev_err(&pdev->dev, "cell-index property not found\n"); - ret = -EINVAL; - goto error; - } - machine_data->ssi_id = be32_to_cpup(iprop); - - /* Get the serial format and clock direction. */ - sprop = of_get_property(np, "fsl,mode", NULL); - if (!sprop) { - dev_err(&pdev->dev, "fsl,mode property not found\n"); - ret = -EINVAL; - goto error; - } - - if (strcasecmp(sprop, "i2s-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBP_CFP; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - - /* In i2s-slave mode, the codec has its own clock source, so we - * need to get the frequency from the device tree and pass it to - * the codec driver. - */ - iprop = of_get_property(codec_np, "clock-frequency", NULL); - if (!iprop || !*iprop) { - dev_err(&pdev->dev, "codec bus-frequency " - "property is missing or invalid\n"); - ret = -EINVAL; - goto error; - } - machine_data->clk_frequency = be32_to_cpup(iprop); - } else if (strcasecmp(sprop, "i2s-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBC_CFC; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "lj-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBP_CFP; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "lj-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBC_CFC; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "rj-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBP_CFP; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "rj-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBC_CFC; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "ac97-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBP_CFP; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "ac97-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBC_CFC; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else { - dev_err(&pdev->dev, - "unrecognized fsl,mode property '%s'\n", sprop); - ret = -EINVAL; - goto error; - } - - if (!machine_data->clk_frequency) { - dev_err(&pdev->dev, "unknown clock frequency\n"); - ret = -EINVAL; - goto error; - } - - /* Find the playback DMA channel to use. */ - machine_data->dai[0].platforms->name = machine_data->platform_name[0]; - ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", - &machine_data->dai[0], - &machine_data->dma_channel_id[0], - &machine_data->dma_id[0]); - if (ret) { - dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); - goto error; - } - - /* Find the capture DMA channel to use. */ - machine_data->dai[1].platforms->name = machine_data->platform_name[1]; - ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", - &machine_data->dai[1], - &machine_data->dma_channel_id[1], - &machine_data->dma_id[1]); - if (ret) { - dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); - goto error; - } - - /* Initialize our DAI data structure. */ - machine_data->dai[0].stream_name = "playback"; - machine_data->dai[1].stream_name = "capture"; - machine_data->dai[0].name = machine_data->dai[0].stream_name; - machine_data->dai[1].name = machine_data->dai[1].stream_name; - - machine_data->card.probe = mpc8610_hpcd_machine_probe; - machine_data->card.remove = mpc8610_hpcd_machine_remove; - machine_data->card.name = pdev->name; /* The platform driver name */ - machine_data->card.owner = THIS_MODULE; - machine_data->card.dev = &pdev->dev; - machine_data->card.num_links = 2; - machine_data->card.dai_link = machine_data->dai; - - /* Register with ASoC */ - ret = snd_soc_register_card(&machine_data->card); - if (ret) { - dev_err(&pdev->dev, "could not register card\n"); - goto error; - } - - of_node_put(codec_np); - - return 0; - -error: - kfree(machine_data); -error_alloc: - of_node_put(codec_np); - return ret; -} - -/** - * mpc8610_hpcd_remove: remove the platform device - * - * This function is called when the platform device is removed. - */ -static void mpc8610_hpcd_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct mpc8610_hpcd_data *machine_data = - container_of(card, struct mpc8610_hpcd_data, card); - - snd_soc_unregister_card(card); - kfree(machine_data); -} - -static struct platform_driver mpc8610_hpcd_driver = { - .probe = mpc8610_hpcd_probe, - .remove_new = mpc8610_hpcd_remove, - .driver = { - /* The name must match 'compatible' property in the device tree, - * in lowercase letters. - */ - .name = "snd-soc-mpc8610hpcd", - }, -}; - -/** - * mpc8610_hpcd_init: machine driver initialization. - * - * This function is called when this module is loaded. - */ -static int __init mpc8610_hpcd_init(void) -{ - struct device_node *guts_np; - struct resource res; - - pr_info("Freescale MPC8610 HPCD ALSA SoC machine driver\n"); - - /* Get the physical address of the global utilities registers */ - guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); - if (of_address_to_resource(guts_np, 0, &res)) { - pr_err("mpc8610-hpcd: missing/invalid global utilities node\n"); - of_node_put(guts_np); - return -EINVAL; - } - guts_phys = res.start; - of_node_put(guts_np); - - return platform_driver_register(&mpc8610_hpcd_driver); -} - -/** - * mpc8610_hpcd_exit: machine driver exit - * - * This function is called when this driver is unloaded. - */ -static void __exit mpc8610_hpcd_exit(void) -{ - platform_driver_unregister(&mpc8610_hpcd_driver); -} - -module_init(mpc8610_hpcd_init); -module_exit(mpc8610_hpcd_exit); - -MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC machine driver"); -MODULE_LICENSE("GPL v2"); -- 2.41.0

2 4

[PATCH] soundwire: stream: constify sdw_port_config when adding devices
by Krzysztof Kozlowski 23 Nov '23

23 Nov '23

sdw_stream_add_master() and sdw_stream_add_slave() do not modify contents of passed sdw_port_config, so it can be made const for code safety and as documentation of expected usage. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/stream.c | 10 +++++----- include/linux/soundwire/sdw.h | 6 +++--- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 69719b335bcb..9dc6399f206a 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -897,7 +897,7 @@ static struct sdw_port_runtime *sdw_port_alloc(struct list_head *port_list) } static int sdw_port_config(struct sdw_port_runtime *p_rt, - struct sdw_port_config *port_config, + const struct sdw_port_config *port_config, int port_index) { p_rt->ch_mask = port_config[port_index].ch_mask; @@ -970,7 +970,7 @@ static int sdw_slave_port_is_valid_range(struct device *dev, int num) static int sdw_slave_port_config(struct sdw_slave *slave, struct sdw_slave_runtime *s_rt, - struct sdw_port_config *port_config) + const struct sdw_port_config *port_config) { struct sdw_port_runtime *p_rt; int ret; @@ -1026,7 +1026,7 @@ static int sdw_master_port_alloc(struct sdw_master_runtime *m_rt, } static int sdw_master_port_config(struct sdw_master_runtime *m_rt, - struct sdw_port_config *port_config) + const struct sdw_port_config *port_config) { struct sdw_port_runtime *p_rt; int ret; @@ -1861,7 +1861,7 @@ EXPORT_SYMBOL(sdw_release_stream); */ int sdw_stream_add_master(struct sdw_bus *bus, struct sdw_stream_config *stream_config, - struct sdw_port_config *port_config, + const struct sdw_port_config *port_config, unsigned int num_ports, struct sdw_stream_runtime *stream) { @@ -1981,7 +1981,7 @@ EXPORT_SYMBOL(sdw_stream_remove_master); */ int sdw_stream_add_slave(struct sdw_slave *slave, struct sdw_stream_config *stream_config, - struct sdw_port_config *port_config, + const struct sdw_port_config *port_config, unsigned int num_ports, struct sdw_stream_runtime *stream) { diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 4f3d14bb1538..904004d8b562 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -1040,7 +1040,7 @@ int sdw_compute_params(struct sdw_bus *bus); int sdw_stream_add_master(struct sdw_bus *bus, struct sdw_stream_config *stream_config, - struct sdw_port_config *port_config, + const struct sdw_port_config *port_config, unsigned int num_ports, struct sdw_stream_runtime *stream); int sdw_stream_remove_master(struct sdw_bus *bus, @@ -1062,7 +1062,7 @@ void sdw_extract_slave_id(struct sdw_bus *bus, u64 addr, struct sdw_slave_id *id int sdw_stream_add_slave(struct sdw_slave *slave, struct sdw_stream_config *stream_config, - struct sdw_port_config *port_config, + const struct sdw_port_config *port_config, unsigned int num_ports, struct sdw_stream_runtime *stream); int sdw_stream_remove_slave(struct sdw_slave *slave, @@ -1084,7 +1084,7 @@ int sdw_update_no_pm(struct sdw_slave *slave, u32 addr, u8 mask, u8 val); static inline int sdw_stream_add_slave(struct sdw_slave *slave, struct sdw_stream_config *stream_config, - struct sdw_port_config *port_config, + const struct sdw_port_config *port_config, unsigned int num_ports, struct sdw_stream_runtime *stream) { -- 2.34.1

3 2

[PATCH 1/2] soundwire: qcom: drop unneeded qcom_swrm_stream_alloc_ports() cleanup
by Krzysztof Kozlowski 23 Nov '23

23 Nov '23

The cleanup in "err" goto label clears bits from pconfig array which is a local variable. This does not have any effect outside of this function, so drop this useless code. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/qcom.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index a1e2d6c98186..754870a4a047 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1157,7 +1157,7 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, struct sdw_port_runtime *p_rt; struct sdw_slave *slave; unsigned long *port_mask; - int i, maxport, pn, nports = 0, ret = 0; + int maxport, pn, nports = 0, ret = 0; unsigned int m_port; mutex_lock(&ctrl->port_lock); @@ -1183,7 +1183,7 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, if (pn > maxport) { dev_err(ctrl->dev, "All ports busy\n"); ret = -EBUSY; - goto err; + goto out; } set_bit(pn, port_mask); pconfig[nports].num = pn; @@ -1205,12 +1205,7 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, sconfig.bps = 1; sdw_stream_add_master(&ctrl->bus, &sconfig, pconfig, nports, stream); -err: - if (ret) { - for (i = 0; i < nports; i++) - clear_bit(pconfig[i].num, port_mask); - } - +out: mutex_unlock(&ctrl->port_lock); return ret; -- 2.34.1

3 4

[PATCH] ASoC: fsl_xcvr: Enable 2 * TX bit clock for spdif only case
by Shengjiu Wang 22 Nov '23

22 Nov '23

The bit 10 in TX_DPTH_CTRL register controls the TX clock rate. If this bit is set, TX datapath clock should be = 2* TX bit rate. If this bit is not set, TX datapath clock should be 10* TX bit rate. As the spdif only case, we always use 2 * TX bit clock, so this bit need to be set. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index fa0a15263c66..77f8e2394bf9 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -414,6 +414,16 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, switch (xcvr->mode) { case FSL_XCVR_MODE_SPDIF: + if (xcvr->soc_data->spdif_only && tx) { + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_TX_DPTH_CTRL_SET, + FSL_XCVR_TX_DPTH_CTRL_BYPASS_FEM, + FSL_XCVR_TX_DPTH_CTRL_BYPASS_FEM); + if (ret < 0) { + dev_err(dai->dev, "Failed to set bypass fem: %d\n", ret); + return ret; + } + } + fallthrough; case FSL_XCVR_MODE_ARC: if (tx) { ret = fsl_xcvr_en_aud_pll(xcvr, fout); -- 2.34.1

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[PATCH v4] ASoC: cs43130: Allow driver to work without IRQ connection
by Maciej Strozek 22 Nov '23

22 Nov '23

Add a polling mechanism that will keep the driver operational even in absence of physical IRQ connection. If IRQ line is detected, the driver will continue working as usual, in case of missing IRQ line it will fallback to the polling mechanism introduced in this change. This will support users which choose not to connect an IRQ line as it is not critical to part's operation. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- V3 -> V4: - Removed unnecesary else and indentation - Changed cs43130_wait_for_completion return codes and updated callers accordingly - Changed approach for checking for presence of IRQ connection: see if client->irq equals 0 which seems to be a standard way of checking it used by other drivers (instead of inspecting devm_request_threaded_irq error code) V2 -> V3: Amended changelog message and subject line V1 -> V2: Add changelog message sound/soc/codecs/cs43130.c | 91 +++++++++++++++++++++++++++----------- sound/soc/codecs/cs43130.h | 1 + 2 files changed, 65 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 02e7f45b9208..bd51bb5b6f14 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -326,6 +326,40 @@ static int cs43130_set_pll(struct snd_soc_component *component, int pll_id, int return ret; } +static int cs43130_wait_for_completion(struct cs43130_private *cs43130, struct completion *to_poll, + int time) +{ + int stickies, offset, flag, ret; + + if (cs43130->has_irq_line) { + ret = wait_for_completion_timeout(to_poll, msecs_to_jiffies(time)); + if (ret == 0) + return -ETIMEDOUT; + else + return 0; // Discard number of jiffies left till timeout and return success + } + + if (to_poll == &cs43130->xtal_rdy) { + offset = 0; + flag = CS43130_XTAL_RDY_INT; + } else if (to_poll == &cs43130->pll_rdy) { + offset = 0; + flag = CS43130_PLL_RDY_INT; + } else if (to_poll == &cs43130->hpload_evt) { + offset = 3; + flag = CS43130_HPLOAD_NO_DC_INT | CS43130_HPLOAD_UNPLUG_INT | + CS43130_HPLOAD_OOR_INT | CS43130_HPLOAD_AC_INT | + CS43130_HPLOAD_DC_INT | CS43130_HPLOAD_ON_INT | + CS43130_HPLOAD_OFF_INT; + } else { + return -EINVAL; + } + + return regmap_read_poll_timeout(cs43130->regmap, CS43130_INT_STATUS_1 + offset, + stickies, (stickies & flag), + 1000, time * 1000); +} + static int cs43130_change_clksrc(struct snd_soc_component *component, enum cs43130_mclk_src_sel src) { @@ -364,14 +398,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for XTAL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for XTAL_READY interrupt: %d\n", ret); + return ret; } } @@ -400,14 +433,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for XTAL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for XTAL_READY interrupt: %d\n", ret); + return ret; } } @@ -416,14 +448,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_PLL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_PLL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->pll_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->pll_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_PLL_RDY_INT_MASK, 1 << CS43130_PLL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for PLL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for PLL_READY interrupt: %d\n", ret); + return ret; } regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, @@ -2048,12 +2079,12 @@ static int cs43130_hpload_proc(struct cs43130_private *cs43130, regmap_multi_reg_write(cs43130->regmap, seq, seq_size); - ret = wait_for_completion_timeout(&cs43130->hpload_evt, - msecs_to_jiffies(1000)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->hpload_evt, 1000); + regmap_read(cs43130->regmap, CS43130_INT_MASK_4, &msk); - if (!ret) { - dev_err(cs43130->dev, "Timeout waiting for HPLOAD interrupt\n"); - return -1; + if (ret) { + dev_err(cs43130->dev, "Timeout waiting for HPLOAD interrupt: %d\n", ret); + return ret; } dev_dbg(cs43130->dev, "HP load stat: %x, INT_MASK_4: %x\n", @@ -2548,13 +2579,19 @@ static int cs43130_i2c_probe(struct i2c_client *client) init_completion(&cs43130->pll_rdy); init_completion(&cs43130->hpload_evt); - ret = devm_request_threaded_irq(cs43130->dev, client->irq, - NULL, cs43130_irq_thread, - IRQF_ONESHOT | IRQF_TRIGGER_LOW, - "cs43130", cs43130); - if (ret != 0) { - dev_err(cs43130->dev, "Failed to request IRQ: %d\n", ret); - goto err; + if (!client->irq) { + dev_dbg(cs43130->dev, "IRQ not found, will poll instead\n"); + cs43130->has_irq_line = 0; + } else { + ret = devm_request_threaded_irq(cs43130->dev, client->irq, + NULL, cs43130_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs43130", cs43130); + if (ret != 0) { + dev_err(cs43130->dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + cs43130->has_irq_line = 1; } cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO; diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h index d3f595bbd3ba..dbdb5b262f1b 100644 --- a/sound/soc/codecs/cs43130.h +++ b/sound/soc/codecs/cs43130.h @@ -508,6 +508,7 @@ struct cs43130_private { struct gpio_desc *reset_gpio; unsigned int dev_id; /* codec device ID */ int xtal_ibias; + bool has_irq_line; /* shared by both DAIs */ struct mutex clk_mutex; -- 2.34.1

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