From 1e09942283570a6d4e448d6b2c708fed296f85d7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:33 +0200 Subject: [PATCH 01/40] ASoC: SOF: ipc4-topology: change chain_dma handling in dai_config mainline inclusion from mainline-v6.9-rc1 The chain_dma mode is currently only handled for HDaudio, but can be used for orther DAIs starting with LunarLake. Move the chain_dma handling earlier. Error detection for the chain_dma case for older platforms is handled at a different level. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 3858464de57b77db51f83e3831950cf18a6aff28) --- sound/soc/sof/ipc4-topology.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index f269cd89909b..de55ed0834c8 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -2777,13 +2777,14 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * if (!data) return 0; + if (pipeline->use_chain_dma) { + pipeline->msg.primary &= ~SOF_IPC4_GLB_CHAIN_DMA_LINK_ID_MASK; + pipeline->msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_LINK_ID(data->dai_data); + return 0; + } + switch (ipc4_copier->dai_type) { case SOF_DAI_INTEL_HDA: - if (pipeline->use_chain_dma) { - pipeline->msg.primary &= ~SOF_IPC4_GLB_CHAIN_DMA_LINK_ID_MASK; - pipeline->msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_LINK_ID(data->dai_data); - break; - } gtw_attr = ipc4_copier->gtw_attr; gtw_attr->lp_buffer_alloc = pipeline->lp_mode; fallthrough; From 79b61425fb6df5cf61ce9c3ae8851f465b6a9ed0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:34 +0200 Subject: [PATCH 02/40] ASoC: SOF: ops: add new 'is_chain_dma_supported' callback mainline inclusion from mainline-v6.9-rc1 IPC4 introduced a 'chain-dma' mode when host and link DMA are connected by firmware without using a regular pipeline or the ability to add intermediate connections. This mode is not available on all platforms and all links, so add a platform-specific callback to help the SOF ipc4-topology core handle different hardware+firmware configurations. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit ba91d0919a78d344d19b02a3899d0921b2f903d1) --- sound/soc/sof/ops.h | 9 +++++++++ sound/soc/sof/sof-priv.h | 2 ++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 46bf8de1a3c4..455905b7dba8 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -569,6 +569,15 @@ snd_sof_set_mach_params(struct snd_soc_acpi_mach *mach, sof_ops(sdev)->set_mach_params(mach, sdev); } +static inline bool +snd_sof_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type) +{ + if (sof_ops(sdev) && sof_ops(sdev)->is_chain_dma_supported) + return sof_ops(sdev)->is_chain_dma_supported(sdev, dai_type); + + return false; +} + /** * snd_sof_dsp_register_poll_timeout - Periodically poll an address * until a condition is met or a timeout occurs diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 619931ca06ca..a4149c203a70 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -338,6 +338,8 @@ struct snd_sof_dsp_ops { struct snd_soc_dai_driver *drv; int num_drv; + bool (*is_chain_dma_supported)(struct snd_sof_dev *sdev, u32 dai_type); /* optional */ + /* ALSA HW info flags, will be stored in snd_pcm_runtime.hw.info */ u32 hw_info; From e006f2bf147aed9c52d4c2e21a703832118626ff Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:35 +0200 Subject: [PATCH 03/40] ASoC: SOF: Intel: hda: add 'is_chain_dma_supported' callback mainline inclusion from mainline-v6.9-rc1 Reuse existing function to get the interface mask and expose it to the SOF core with a callback - the main user is the IPC4 topology so only HDaudio platforms provide this callback. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit d69f9ecbe1ecfa97b9c8ab7b6332bd73ba2ff4d8) --- sound/soc/sof/intel/hda-common-ops.c | 1 + sound/soc/sof/intel/hda.c | 63 ++++++++++++++++++++++------ sound/soc/sof/intel/hda.h | 5 +++ sound/soc/sof/sof-priv.h | 7 ++++ 4 files changed, 64 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 1cc18fb2b75b..3f9d3db68a3c 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -82,6 +82,7 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { /* DAI drivers */ .drv = skl_dai, .num_drv = SOF_SKL_NUM_DAIS, + .is_chain_dma_supported = hda_is_chain_dma_supported, /* PM */ .suspend = hda_dsp_suspend, diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index cfe99dc0afa0..8401b9e4dba6 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -46,44 +46,83 @@ #define EXCEPT_MAX_HDR_SIZE 0x400 #define HDA_EXT_ROM_STATUS_SIZE 8 -static u32 hda_get_interface_mask(struct snd_sof_dev *sdev) +static void hda_get_interfaces(struct snd_sof_dev *sdev, u32 *interface_mask) { const struct sof_intel_dsp_desc *chip; - u32 interface_mask[2] = { 0 }; chip = get_chip_info(sdev->pdata); switch (chip->hw_ip_version) { case SOF_INTEL_TANGIER: case SOF_INTEL_BAYTRAIL: case SOF_INTEL_BROADWELL: - interface_mask[0] = BIT(SOF_DAI_INTEL_SSP); + interface_mask[SOF_DAI_DSP_ACCESS] = BIT(SOF_DAI_INTEL_SSP); break; case SOF_INTEL_CAVS_1_5: case SOF_INTEL_CAVS_1_5_PLUS: - interface_mask[0] = BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | - BIT(SOF_DAI_INTEL_HDA); - interface_mask[1] = BIT(SOF_DAI_INTEL_HDA); + interface_mask[SOF_DAI_DSP_ACCESS] = + BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | BIT(SOF_DAI_INTEL_HDA); + interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA); break; case SOF_INTEL_CAVS_1_8: case SOF_INTEL_CAVS_2_0: case SOF_INTEL_CAVS_2_5: case SOF_INTEL_ACE_1_0: - interface_mask[0] = BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | - BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); - interface_mask[1] = BIT(SOF_DAI_INTEL_HDA); + interface_mask[SOF_DAI_DSP_ACCESS] = + BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | + BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); + interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA); break; case SOF_INTEL_ACE_2_0: - interface_mask[0] = BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | - BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); - interface_mask[1] = interface_mask[0]; /* all interfaces accessible without DSP */ + interface_mask[SOF_DAI_DSP_ACCESS] = + BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | + BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); + /* all interfaces accessible without DSP */ + interface_mask[SOF_DAI_HOST_ACCESS] = + interface_mask[SOF_DAI_DSP_ACCESS]; break; default: break; } +} + +static u32 hda_get_interface_mask(struct snd_sof_dev *sdev) +{ + u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 }; + + hda_get_interfaces(sdev, interface_mask); return interface_mask[sdev->dspless_mode_selected]; } +bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type) +{ + u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 }; + const struct sof_intel_dsp_desc *chip; + + if (sdev->dspless_mode_selected) + return false; + + hda_get_interfaces(sdev, interface_mask); + + if (!(interface_mask[SOF_DAI_DSP_ACCESS] & BIT(dai_type))) + return false; + + if (dai_type == SOF_DAI_INTEL_HDA) + return true; + + switch (dai_type) { + case SOF_DAI_INTEL_SSP: + case SOF_DAI_INTEL_DMIC: + case SOF_DAI_INTEL_ALH: + chip = get_chip_info(sdev->pdata); + if (chip->hw_ip_version < SOF_INTEL_ACE_2_0) + return false; + return true; + default: + return false; + } +} + #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) /* diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index dc0791d931c1..326cce6b8cac 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -570,6 +570,11 @@ struct sof_intel_hda_stream { #define SOF_STREAM_SD_OFFSET_CRST 0x1 +/* + * DAI support + */ +bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type); + /* * DSP Core services. */ diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index a4149c203a70..f5f7386e37bd 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -157,6 +157,13 @@ struct sof_firmware { u32 payload_offset; }; +enum sof_dai_access { + SOF_DAI_DSP_ACCESS, /* access from DSP only */ + SOF_DAI_HOST_ACCESS, /* access from host only */ + + SOF_DAI_ACCESS_NUM +}; + /* * SOF DSP HW abstraction operations. * Used to abstract DSP HW architecture and any IO busses between host CPU From 5fbc095ab9f3662ef6b9495612ccdd37adfe7723 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:36 +0200 Subject: [PATCH 04/40] ASoC: SOF: Intel: hda-dai-ops: enable chain_dma for ALH mainline inclusion from mainline-v6.9-rc1 Use the existing callbacks and mix/match of HDaudio and SoundWire support. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit a5b7767723e739c700f5c56841790a85bd7f13ae) --- sound/soc/sof/intel/hda-dai-ops.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 87935554b1e4..9f67f6546e75 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -521,6 +521,17 @@ static const struct hda_dai_widget_dma_ops hda_ipc4_chain_dma_ops = { .get_hlink = hda_get_hlink, }; +static const struct hda_dai_widget_dma_ops sdw_ipc4_chain_dma_ops = { + .get_hext_stream = hda_get_hext_stream, + .assign_hext_stream = hda_assign_hext_stream, + .release_hext_stream = hda_release_hext_stream, + .setup_hext_stream = hda_setup_hext_stream, + .reset_hext_stream = hda_reset_hext_stream, + .trigger = hda_trigger, + .calc_stream_format = generic_calc_stream_format, + .get_hlink = sdw_get_hlink, +}; + static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, struct snd_pcm_substream *substream, int cmd) { @@ -619,6 +630,8 @@ hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidg } case SOF_IPC_TYPE_4: { + struct snd_sof_widget *pipe_widget = swidget->spipe->pipe_widget; + struct sof_ipc4_pipeline *pipeline = pipe_widget->private; struct sof_ipc4_copier *ipc4_copier = sdai->private; const struct sof_intel_dsp_desc *chip; @@ -626,15 +639,10 @@ hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidg switch (ipc4_copier->dai_type) { case SOF_DAI_INTEL_HDA: - { - struct snd_sof_widget *pipe_widget = swidget->spipe->pipe_widget; - struct sof_ipc4_pipeline *pipeline = pipe_widget->private; - if (pipeline->use_chain_dma) return &hda_ipc4_chain_dma_ops; return &hda_ipc4_dma_ops; - } case SOF_DAI_INTEL_SSP: if (chip->hw_ip_version < SOF_INTEL_ACE_2_0) return NULL; @@ -646,6 +654,8 @@ hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidg case SOF_DAI_INTEL_ALH: if (chip->hw_ip_version < SOF_INTEL_ACE_2_0) return NULL; + if (pipeline->use_chain_dma) + return &sdw_ipc4_chain_dma_ops; return &sdw_ipc4_dma_ops; default: From 1bd1a04d930590dfca3ae65d2f7ce68f1d5ac5d0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:37 +0200 Subject: [PATCH 05/40] ASoC: SOF: ipc4: store number of playback/capture streams MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit mainline inclusion from mainline-v6.9-rc1 The CHAIN_DMA IPC needs the number of playback streams as a start offset for the dma_id of a capture stream. This offset can be retrieved on Intel platforms from the GCAP information, and stored in the sof_ipc4_fw_data structure. One could argue that the fields added are not really dependent on any firmware definitions but rather on hardware capabilities, but they are required for the IPC CHAIN_DMA definitions so adding them in ipc4_fw_data isn't completely silly. The CHAIN_DMA IPC is currently only functional on Intel HDaudio DMAs, and gated by the snd_sof_is_chain_dma_supported() helper. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 426476344f01096c7dae6c5413cc8a8d9bbdea29) --- sound/soc/sof/intel/hda-stream.c | 9 +++++++++ sound/soc/sof/ipc4-priv.h | 4 ++++ 2 files changed, 13 insertions(+) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index d0b70ac39ae8..e02f13a36aad 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -21,6 +21,7 @@ #include #include "../ops.h" #include "../sof-audio.h" +#include "../ipc4-priv.h" #include "hda.h" #define HDA_LTRP_GB_VALUE_US 95 @@ -937,6 +938,14 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev) /* store total stream count (playback + capture) from GCAP */ sof_hda->stream_max = num_total; + /* store stream count from GCAP required for CHAIN_DMA */ + if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) { + struct sof_ipc4_fw_data *ipc4_data = sdev->private; + + ipc4_data->num_playback_streams = num_playback; + ipc4_data->num_capture_streams = num_capture; + } + return 0; } diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index 9e69b7c29117..14020c1ebb45 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -66,6 +66,8 @@ struct sof_ipc4_fw_library { * @nhlt: NHLT table either from the BIOS or the topology manifest * @mtrace_type: mtrace type supported on the booted platform * @mtrace_log_bytes: log bytes as reported by the firmware via fw_config reply + * @num_playback_streams: max number of playback DMAs, needed for CHAIN_DMA offset + * @num_capture_streams: max number of capture DMAs * @max_num_pipelines: max number of pipelines * @max_libs_count: Maximum number of libraries support by the FW including the * base firmware @@ -79,6 +81,8 @@ struct sof_ipc4_fw_data { void *nhlt; enum sof_ipc4_mtrace_type mtrace_type; u32 mtrace_log_bytes; + int num_playback_streams; + int num_capture_streams; int max_num_pipelines; u32 max_libs_count; From d801249c80a7f48bc75aefc20b9c2e9dac215e19 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:38 +0200 Subject: [PATCH 06/40] ASoC: SOF: ipc4-pcm: fix dma_id for CHAIN_DMA capture MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit mainline inclusion from mainline-v6.9-rc1 The existing code uses (stream_tag - 1) for the host and link dma id. This is correct for playback, but for capture this results in an invalid dma_type being used. The firmware assumes that the dma_id for capture is always larger than DAI_NUM_HDA_OUT This patch adds the offset for num_playback_streams, filled on Intel platforms with the value extracted from the hardware capabilities. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-7-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 8722d245a73ff32491bff390136367ec223e2906) --- sound/soc/sof/ipc4-pcm.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 07eb5c6d4adf..0f332c8cdbe6 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -231,9 +231,11 @@ sof_ipc4_update_pipeline_state(struct snd_sof_dev *sdev, int state, int cmd, */ static int sof_ipc4_chain_dma_trigger(struct snd_sof_dev *sdev, + int direction, struct snd_sof_pcm_stream_pipeline_list *pipeline_list, int state, int cmd) { + struct sof_ipc4_fw_data *ipc4_data = sdev->private; bool allocate, enable, set_fifo_size; struct sof_ipc4_msg msg = {{ 0 }}; int i; @@ -294,6 +296,20 @@ static int sof_ipc4_chain_dma_trigger(struct snd_sof_dev *sdev, msg.extension |= pipeline->msg.extension; } + if (direction == SNDRV_PCM_STREAM_CAPTURE) { + /* + * For ChainDMA the DMA ids are unique with the following mapping: + * playback: 0 - (num_playback_streams - 1) + * capture: num_playback_streams - (num_playback_streams + + * num_capture_streams - 1) + * + * Add the num_playback_streams offset to the DMA ids stored in + * msg.primary in case capture + */ + msg.primary += SOF_IPC4_GLB_CHAIN_DMA_HOST_ID(ipc4_data->num_playback_streams); + msg.primary += SOF_IPC4_GLB_CHAIN_DMA_LINK_ID(ipc4_data->num_playback_streams); + } + if (allocate) msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_ALLOCATE_MASK; @@ -340,7 +356,8 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, * trigger function that handles the rest for the substream. */ if (pipeline->use_chain_dma) - return sof_ipc4_chain_dma_trigger(sdev, pipeline_list, state, cmd); + return sof_ipc4_chain_dma_trigger(sdev, substream->stream, + pipeline_list, state, cmd); /* allocate memory for the pipeline data */ trigger_list = kzalloc(struct_size(trigger_list, pipeline_instance_ids, From 182ab6f2f612eece44ab3bb57b6b5a6e8e70fde8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:39 +0200 Subject: [PATCH 07/40] ASoC: SOF: ipc4-topology: allow chain_dma for all supported DAIs mainline inclusion from mainline-v6.9-rc1 Now that we have a 'is_chain_dma_supported' callback we can use it to double-check possible disconnects between a topology file enabling chain-dma for a DAI and the hardware/firmware capabilities. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-8-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit df82dbb5fb28a762113fc6d98985d36ef7785e32) --- sound/soc/sof/ipc4-topology.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index de55ed0834c8..2b357c1413fe 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -497,6 +497,7 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) { struct sof_ipc4_available_audio_format *available_fmt; struct snd_soc_component *scomp = swidget->scomp; + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct snd_sof_dai *dai = swidget->private; struct sof_ipc4_copier *ipc4_copier; struct snd_sof_widget *pipe_widget; @@ -540,10 +541,11 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) pipe_widget = swidget->spipe->pipe_widget; pipeline = pipe_widget->private; - if (pipeline->use_chain_dma && ipc4_copier->dai_type != SOF_DAI_INTEL_HDA) { - dev_err(scomp->dev, - "Bad DAI type '%d', Chained DMA is only supported by HDA DAIs (%d).\n", - ipc4_copier->dai_type, SOF_DAI_INTEL_HDA); + + if (pipeline->use_chain_dma && + !snd_sof_is_chain_dma_supported(sdev, ipc4_copier->dai_type)) { + dev_err(scomp->dev, "Bad DAI type '%d', Chain DMA is not supported\n", + ipc4_copier->dai_type); ret = -ENODEV; goto free_available_fmt; } From 9caee0688989f496f5570b232fb99466d267d331 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:40 +0200 Subject: [PATCH 08/40] ASoC: SOF: Intel: hda-dai: remove dspless special case MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit mainline inclusion from mainline-v6.9-rc1 The existing code forces a parameter to be NULL but that parameter is not used yet. Remove the special case in preparation for additional changes. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-9-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit daa09d0615ce9c781777802874cffa4380f883c3) --- sound/soc/sof/intel/hda-dai.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 63058b0104ca..52a49200d444 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -83,12 +83,8 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai sdev = widget_to_sdev(w); - /* - * The swidget parameter of hda_select_dai_widget_ops() is ignored in - * case of DSPless mode - */ if (sdev->dspless_mode_selected) - return hda_select_dai_widget_ops(sdev, NULL); + return hda_select_dai_widget_ops(sdev, swidget); sdai = swidget->private; From 5d008dd12d5f12cfab7248c28aadf3015ef9cb03 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:41 +0200 Subject: [PATCH 09/40] ASoC: SOF: topology: dynamically allocate and store DAI widget->private MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit mainline inclusion from mainline-v6.9-rc1 For dspless mode, we need to allocate and store an 'sdai' structure. The existing code allocate the data on the stack and does not set the widget->private pointer. This minor change should not have any impact on existing DAIs, even when the DSP is used. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-10-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 743eb6c68d3534e01e73d316ddcaa7334c0e29d3) --- sound/soc/sof/sof-audio.c | 8 +++----- sound/soc/sof/topology.c | 13 ++++++++++--- 2 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index c0625e79e368..d1a7d867f6a3 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -46,7 +46,6 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, { const struct sof_ipc_tplg_ops *tplg_ops = sof_ipc_get_ops(sdev, tplg); struct snd_sof_pipeline *spipe = swidget->spipe; - struct snd_sof_widget *pipe_widget; int err = 0; int ret; @@ -59,8 +58,6 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, if (--swidget->use_count) return 0; - pipe_widget = swidget->spipe->pipe_widget; - /* reset route setup status for all routes that contain this widget */ sof_reset_route_setup_status(sdev, swidget); @@ -109,8 +106,9 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, * free the scheduler widget (same as pipe_widget) associated with the current swidget. * skip for static pipelines */ - if (swidget->dynamic_pipeline_widget && swidget->id != snd_soc_dapm_scheduler) { - ret = sof_widget_free_unlocked(sdev, pipe_widget); + if (swidget->spipe && swidget->dynamic_pipeline_widget && + swidget->id != snd_soc_dapm_scheduler) { + ret = sof_widget_free_unlocked(sdev, swidget->spipe->pipe_widget); if (ret < 0 && !err) err = ret; } diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index cf1e63daad86..1165fd2560bc 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2347,23 +2347,29 @@ static int sof_dspless_widget_ready(struct snd_soc_component *scomp, int index, if (WIDGET_IS_DAI(w->id)) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct snd_sof_widget *swidget; - struct snd_sof_dai dai; + struct snd_sof_dai *sdai; int ret; swidget = kzalloc(sizeof(*swidget), GFP_KERNEL); if (!swidget) return -ENOMEM; - memset(&dai, 0, sizeof(dai)); + sdai = kzalloc(sizeof(*sdai), GFP_KERNEL); + if (!sdai) { + kfree(swidget); + return -ENOMEM; + } - ret = sof_connect_dai_widget(scomp, w, tw, &dai); + ret = sof_connect_dai_widget(scomp, w, tw, sdai); if (ret) { kfree(swidget); + kfree(sdai); return ret; } swidget->scomp = scomp; swidget->widget = w; + swidget->private = sdai; mutex_init(&swidget->setup_mutex); w->dobj.private = swidget; list_add(&swidget->list, &sdev->widget_list); @@ -2387,6 +2393,7 @@ static int sof_dspless_widget_unload(struct snd_soc_component *scomp, /* remove and free swidget object */ list_del(&swidget->list); + kfree(swidget->private); kfree(swidget); } From f1b296ab44be8021891f4a9365a29f88d474eb59 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 14:06:56 +0200 Subject: [PATCH 10/40] ALSA: hda: intel-nhlt: add intel_nhlt_ssp_device_type() function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit mainline inclusion from mainline-v6.9-rc2 commit 02545bc57512b7660625e454e60e3fb0d07f660d upstream. Add a helper function intel_nhlt_ssp_device_type() to detect the type of specific SSP port. The result is nhlt_device_type enum type which could be NHLT_DEVICE_BT or NHLT_DEVICE_I2S. Signed-off-by: Brent Lu Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Signed-off-by: Takashi Iwai Message-ID: <20231127120657.19764-2-peter.ujfalusi@linux.intel.com> (cherry picked from commit 02545bc57512b7660625e454e60e3fb0d07f660d) --- include/sound/intel-nhlt.h | 10 ++++++++++ sound/hda/intel-nhlt.c | 26 ++++++++++++++++++++++++++ 2 files changed, 36 insertions(+) diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h index 53470d6a28d6..24dbe16684ae 100644 --- a/include/sound/intel-nhlt.h +++ b/include/sound/intel-nhlt.h @@ -143,6 +143,9 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, u32 bus_id, u8 link_type, u8 vbps, u8 bps, u8 num_ch, u32 rate, u8 dir, u8 dev_type); +int intel_nhlt_ssp_device_type(struct device *dev, struct nhlt_acpi_table *nhlt, + u8 virtual_bus_id); + #else static inline struct nhlt_acpi_table *intel_nhlt_init(struct device *dev) @@ -184,6 +187,13 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, return NULL; } +static inline int intel_nhlt_ssp_device_type(struct device *dev, + struct nhlt_acpi_table *nhlt, + u8 virtual_bus_id) +{ + return -EINVAL; +} + #endif #endif diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 696a958d93e9..088cff799e0b 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -343,3 +343,29 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, return NULL; } EXPORT_SYMBOL(intel_nhlt_get_endpoint_blob); + +int intel_nhlt_ssp_device_type(struct device *dev, struct nhlt_acpi_table *nhlt, + u8 virtual_bus_id) +{ + struct nhlt_endpoint *epnt; + int i; + + if (!nhlt) + return -EINVAL; + + epnt = (struct nhlt_endpoint *)nhlt->desc; + for (i = 0; i < nhlt->endpoint_count; i++) { + /* for SSP link the virtual bus id is the SSP port number */ + if (epnt->linktype == NHLT_LINK_SSP && + epnt->virtual_bus_id == virtual_bus_id) { + dev_dbg(dev, "SSP%d: dev_type=%d\n", virtual_bus_id, + epnt->device_type); + return epnt->device_type; + } + + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); + } + + return -EINVAL; +} +EXPORT_SYMBOL(intel_nhlt_ssp_device_type); From f143944c15ca0c699a1ce96a0b9aad53660fe571 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 14:06:57 +0200 Subject: [PATCH 11/40] ASoC: SOF: ipc4-topology: support NHLT device type MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit mainline inclusion from mainline-v6.9-rc2 commit 188ab4bfd29d7c91e35873a360a31e95a6ff0816 upstream. The endpoint in NHLT table for a SSP port could have the device type NHLT_DEVICE_BT or NHLT_DEVICE_I2S. Use intel_nhlt_ssp_device_type() function to retrieve the device type before querying the endpoint blob to make sure we are always using correct device type parameter. Signed-off-by: Brent Lu Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Takashi Iwai Message-ID: <20231127120657.19764-3-peter.ujfalusi@linux.intel.com> (cherry picked from commit 188ab4bfd29d7c91e35873a360a31e95a6ff0816) --- sound/soc/sof/ipc4-topology.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 2b357c1413fe..974ddceecdad 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1345,6 +1345,7 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s int sample_rate, channel_count; int bit_depth, ret; u32 nhlt_type; + int dev_type = 0; /* convert to NHLT type */ switch (linktype) { @@ -1360,18 +1361,30 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s &bit_depth); if (ret < 0) return ret; + + /* + * We need to know the type of the external device attached to a SSP + * port to retrieve the blob from NHLT. However, device type is not + * specified in topology. + * Query the type for the port and then pass that information back + * to the blob lookup function. + */ + dev_type = intel_nhlt_ssp_device_type(sdev->dev, ipc4_data->nhlt, + dai_index); + if (dev_type < 0) + return dev_type; break; default: return 0; } - dev_dbg(sdev->dev, "dai index %d nhlt type %d direction %d\n", - dai_index, nhlt_type, dir); + dev_dbg(sdev->dev, "dai index %d nhlt type %d direction %d dev type %d\n", + dai_index, nhlt_type, dir, dev_type); /* find NHLT blob with matching params */ cfg = intel_nhlt_get_endpoint_blob(sdev->dev, ipc4_data->nhlt, dai_index, nhlt_type, bit_depth, bit_depth, channel_count, sample_rate, - dir, 0); + dir, dev_type); if (!cfg) { dev_err(sdev->dev, From d0bc0992ada8475047437e554d8c298f0f989d34 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:07:58 +0200 Subject: [PATCH 12/40] ASoC: SOF: Add dsp_max_burst_size_in_ms member to snd_sof_pcm_stream mainline inclusion from mainline-v6.9-rc3 The dsp_max_burst_size_in_ms can be used to save the length of the maximum burst size in ms the host DMA will use. Platform code can place constraint using this to avoid user space requesting too small ALSA buffer which will result xruns. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit fb9f8125ed9d9b8e11f309a7dbfbe7b40de48fba) --- sound/soc/sof/sof-audio.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 3606595a7500..d302ccc811f4 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -322,6 +322,7 @@ struct snd_sof_pcm_stream { struct work_struct period_elapsed_work; struct snd_soc_dapm_widget_list *list; /* list of connected DAPM widgets */ bool d0i3_compatible; /* DSP can be in D0I3 when this pcm is opened */ + unsigned int dsp_max_burst_size_in_ms; /* The maximum size of the host DMA burst in ms */ /* * flag to indicate that the DSP pipelines should be kept * active or not while suspending the stream From 6090cc711d0f85817e6cfa1d9d8636f907c873c2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:07:59 +0200 Subject: [PATCH 13/40] ASoC: SOF: ipc4-topology: Save the DMA maximum burst size for PCMs mainline inclusion from mainline-v6.9-rc3 When setting up the pcm widget, save the DSP buffer size (in ms) for platform code to place a constraint on playback. On playback the DMA will fill the buffer on start and if the period size is smaller it will immediately overrun. On capture the DMA will move data in 1ms bursts. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 842bb8b62cc6f3546d61eb63115b32ebc6dd4a87) --- sound/soc/sof/ipc4-topology.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 974ddceecdad..1229c4ec6110 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -400,8 +400,9 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) struct sof_ipc4_available_audio_format *available_fmt; struct snd_soc_component *scomp = swidget->scomp; struct sof_ipc4_copier *ipc4_copier; + struct snd_sof_pcm *spcm; int node_type = 0; - int ret; + int ret, dir; ipc4_copier = kzalloc(sizeof(*ipc4_copier), GFP_KERNEL); if (!ipc4_copier) @@ -435,6 +436,25 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) } dev_dbg(scomp->dev, "host copier '%s' node_type %u\n", swidget->widget->name, node_type); + spcm = snd_sof_find_spcm_comp(scomp, swidget->comp_id, &dir); + if (!spcm) + goto skip_gtw_cfg; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + struct snd_sof_pcm_stream *sps = &spcm->stream[dir]; + + sof_update_ipc_object(scomp, &sps->dsp_max_burst_size_in_ms, + SOF_COPIER_DEEP_BUFFER_TOKENS, + swidget->tuples, + swidget->num_tuples, sizeof(u32), 1); + /* Set default DMA buffer size if it is not specified in topology */ + if (!sps->dsp_max_burst_size_in_ms) + sps->dsp_max_burst_size_in_ms = SOF_IPC4_MIN_DMA_BUFFER_SIZE; + } else { + /* Capture data is copied from DSP to host in 1ms bursts */ + spcm->stream[dir].dsp_max_burst_size_in_ms = 1; + } + skip_gtw_cfg: ipc4_copier->gtw_attr = kzalloc(sizeof(*ipc4_copier->gtw_attr), GFP_KERNEL); if (!ipc4_copier->gtw_attr) { From 7260bfa5ed8344eeb9c323ec7d774177e365b7fd Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:00 +0200 Subject: [PATCH 14/40] ASoC: SOF: Intel: hda-pcm: Use dsp_max_burst_size_in_ms to place constraint mainline inclusion from mainline-v6.9-rc3 If the PCM have the dsp_max_burst_size_in_ms set then place a constraint to limit the minimum buffer time to avoid xruns caused by DMA bursts spinning on the ALSA buffer. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit fe76d2e75a6da97edd2b4ec5cfb9efd541be087a) --- sound/soc/sof/intel/hda-pcm.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index b59b5156b5ad..c84e9d031d80 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -265,6 +265,27 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev, snd_pcm_hw_constraint_mask64(substream->runtime, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32); + /* + * The dsp_max_burst_size_in_ms is the length of the maximum burst size + * of the host DMA in the ALSA buffer. + * + * On playback start the DMA will transfer dsp_max_burst_size_in_ms + * amount of data in one initial burst to fill up the host DMA buffer. + * Consequent DMA burst sizes are shorter and their length can vary. + * To make sure that userspace allocate large enough ALSA buffer we need + * to place a constraint on the buffer time. + * + * On capture the DMA will transfer 1ms chunks. + * + * Exact dsp_max_burst_size_in_ms constraint is racy, so set the + * constraint to a minimum of 2x dsp_max_burst_size_in_ms. + */ + if (spcm->stream[direction].dsp_max_burst_size_in_ms) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_TIME, + spcm->stream[direction].dsp_max_burst_size_in_ms * USEC_PER_MSEC * 2, + UINT_MAX); + /* binding pcm substream to hda stream */ substream->runtime->private_data = &dsp_stream->hstream; return 0; From 1a9d914041aa3cdf8db5c40d342007387433803e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:01 +0200 Subject: [PATCH 15/40] ASoC: SOF: Intel: hda: Implement get_stream_position (Linear Link Position) mainline inclusion from mainline-v6.9-rc3 When the Linear Link Position is not available in firmware SRAM window we use the host accessible position registers to read it. The address of the PPLCLLPL/U registers depend on the number of streams (playback+capture). At probe time the pplc_addr is calculated for each stream and we can use it to read the LLP without the need of address re-calculation. Set the get_stream_position callback in sof_hda_common_ops for all platforms: The callback is used for IPC4 delay calculations only but the register is a generic HDA register, not tied to any specific IPC version. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 67b182bea08a8d1092b91b57aefdfe420fce1634) --- sound/soc/sof/intel/hda-common-ops.c | 2 ++ sound/soc/sof/intel/hda-stream.c | 32 ++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 3 +++ 3 files changed, 37 insertions(+) diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 3f9d3db68a3c..92379a855901 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -56,6 +56,8 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { .pcm_pointer = hda_dsp_pcm_pointer, .pcm_ack = hda_dsp_pcm_ack, + .get_stream_position = hda_dsp_get_stream_llp, + /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index e02f13a36aad..dc5b27b25d44 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1063,3 +1063,35 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, return pos; } + +/** + * hda_dsp_get_stream_llp - Retrieve the LLP (Linear Link Position) of the stream + * @sdev: SOF device + * @component: ASoC component + * @substream: PCM substream + * + * Returns the raw Linear Link Position value + */ +u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + u32 llp_l, llp_u; + + /* + * The pplc_addr have been calculated during probe in + * hda_dsp_stream_init(): + * pplc_addr = sdev->bar[HDA_DSP_PP_BAR] + + * SOF_HDA_PPLC_BASE + + * SOF_HDA_PPLC_MULTI * total_stream + + * SOF_HDA_PPLC_INTERVAL * stream_index + * + * Use this pre-calculated address to avoid repeated re-calculation. + */ + llp_l = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); + llp_u = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); + + return ((u64)llp_u << 32) | llp_l; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 326cce6b8cac..75c92b1cf897 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -658,6 +658,9 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, int direction, bool can_sleep); +u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); From 342b20eaa614f65528ac0a94dedaa97d58560b99 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:02 +0200 Subject: [PATCH 16/40] ASoC: SOF: Intel: mtl/lnl: Use the generic get_stream_position callback mainline inclusion from mainline-v6.9-rc3 Drop the MTL mtl_dsp_get_stream_hda_link_position() function and related defines since it can only work on platforms which have 19 streams because of the use of 0x948 as base offset for the LLP registers. The generic hda_dsp_get_stream_hda_link_position() takes the number of streams into consideration when reading the LLP registers for the stream and can handle different HDA configurations. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 4374f698d7d9f849b66f3fa8f7a64f0bc1a53d7f) --- sound/soc/sof/intel/lnl.c | 2 -- sound/soc/sof/intel/mtl.c | 14 -------------- sound/soc/sof/intel/mtl.h | 10 ---------- 3 files changed, 26 deletions(-) diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index b7c5d56e6418..beea09e27c93 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -119,8 +119,6 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) sof_lnl_ops.resume = lnl_hda_dsp_resume; sof_lnl_ops.runtime_resume = lnl_hda_dsp_runtime_resume; - sof_lnl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; - /* dsp core get/put */ sof_lnl_ops.core_get = mtl_dsp_core_get; sof_lnl_ops.core_put = mtl_dsp_core_put; diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 90d217c137b7..322ff0338742 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -654,18 +654,6 @@ static int mtl_dsp_disable_interrupts(struct snd_sof_dev *sdev) return mtl_enable_interrupts(sdev, false); } -u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - struct hdac_stream *hstream = substream->runtime->private_data; - u32 llp_l, llp_u; - - llp_l = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPL(hstream->index)); - llp_u = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPU(hstream->index)); - return ((u64)llp_u << 32) | llp_l; -} - int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core) { const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm; @@ -735,8 +723,6 @@ int sof_mtl_ops_init(struct snd_sof_dev *sdev) sof_mtl_ops.core_get = mtl_dsp_core_get; sof_mtl_ops.core_put = mtl_dsp_core_put; - sof_mtl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; - sdev->private = kzalloc(sizeof(struct sof_ipc4_fw_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index 82dd6b8c4859..3c56427a966b 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -6,12 +6,6 @@ * Copyright(c) 2020-2022 Intel Corporation. All rights reserved. */ -/* HDA Registers */ -#define MTL_PPLCLLPL_BASE 0x948 -#define MTL_PPLCLLPU_STRIDE 0x10 -#define MTL_PPLCLLPL(x) (MTL_PPLCLLPL_BASE + (x) * MTL_PPLCLLPU_STRIDE) -#define MTL_PPLCLLPU(x) (MTL_PPLCLLPL_BASE + 0x4 + (x) * MTL_PPLCLLPU_STRIDE) - /* DSP Registers */ #define MTL_HFDSSCS 0x1000 #define MTL_HFDSSCS_SPA_MASK BIT(16) @@ -103,9 +97,5 @@ int mtl_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id); void mtl_ipc_dump(struct snd_sof_dev *sdev); -u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream); - int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core); int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core); From 6bd6357694f32dbebf21e14735db072ac6c0d293 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:03 +0200 Subject: [PATCH 17/40] ASoC: SOF: Introduce a new callback pair to be used for PCM delay reporting mainline inclusion from mainline-v6.9-rc3 For delay calculation we need two information: Number of bytes transferred between the DSP and host memory (ALSA buffer) Number of frames transferred between the DSP and external device (link/codec/DMIC/etc). The reason for the different units (bytes vs frames) on host and dai side is that the format on the dai side is decided by the firmware and might not be the same as on the host side, thus the expectation is that the counter reflects the number of frames. The kernel know the host side format and in there we have access to the DMA position which is in bytes. In a simplified way, the DSP caused delay is the difference between the two counters. The existing get_stream_position callback is defined to retrieve the frame counter on the DAI side but it's name is too generic to be intuitive and makes it hard to define a callback for the host side. This patch introduces a new set of callbacks to replace the get_stream_position and define the host side equivalent: get_dai_frame_counter get_host_byte_counter Subsequent patches will remove the old callback. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-7-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit ce2faa9a180c1984225689b6b1cb26045f8b7470) --- sound/soc/sof/ops.h | 24 ++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 21 +++++++++++++++++++++ 2 files changed, 45 insertions(+) diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 455905b7dba8..3c8adbbf11a4 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -535,6 +535,30 @@ static inline u64 snd_sof_pcm_get_stream_position(struct snd_sof_dev *sdev, return 0; } +static inline u64 +snd_sof_pcm_get_dai_frame_counter(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (sof_ops(sdev) && sof_ops(sdev)->get_dai_frame_counter) + return sof_ops(sdev)->get_dai_frame_counter(sdev, component, + substream); + + return 0; +} + +static inline u64 +snd_sof_pcm_get_host_byte_counter(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (sof_ops(sdev) && sof_ops(sdev)->get_host_byte_counter) + return sof_ops(sdev)->get_host_byte_counter(sdev, component, + substream); + + return 0; +} + /* machine driver */ static inline int snd_sof_machine_register(struct snd_sof_dev *sdev, void *pdata) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index f5f7386e37bd..4752985d57e2 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -270,6 +270,27 @@ struct snd_sof_dsp_ops { struct snd_soc_component *component, struct snd_pcm_substream *substream); /* optional */ + /* + * optional callback to retrieve the number of frames left/arrived from/to + * the DSP on the DAI side (link/codec/DMIC/etc). + * + * The callback is used when the firmware does not provide this information + * via the shared SRAM window and it can be retrieved by host. + */ + u64 (*get_dai_frame_counter)(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); /* optional */ + + /* + * Optional callback to retrieve the number of bytes left/arrived from/to + * the DSP on the host side (bytes between host ALSA buffer and DSP). + * + * The callback is needed for ALSA delay reporting. + */ + u64 (*get_host_byte_counter)(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); /* optional */ + /* host read DSP stream data */ int (*ipc_msg_data)(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sps, From 719d718eb45a11293b42ab25ed77c2213d6b7401 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:04 +0200 Subject: [PATCH 18/40] ASoC: SOF: Intel: Set the dai/host get frame/byte counter callbacks mainline inclusion from mainline-v6.9-rc3 Add implementation for reading the LDP (Linear DMA Position) to be used as get_host_byte_counter(). The LDP is counting the number of bytes moved between the DSP and host memory. Set the get_dai_frame_counter to hda_dsp_get_stream_llp, which is counting the frames on the link side of the DSP. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-8-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit fd6f6a0632bc891673490bf4a92304172251825c) --- sound/soc/sof/intel/hda-common-ops.c | 2 ++ sound/soc/sof/intel/hda-stream.c | 31 ++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 3 +++ 3 files changed, 36 insertions(+) diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 92379a855901..e2b37486d8c8 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -57,6 +57,8 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { .pcm_ack = hda_dsp_pcm_ack, .get_stream_position = hda_dsp_get_stream_llp, + .get_dai_frame_counter = hda_dsp_get_stream_llp, + .get_host_byte_counter = hda_dsp_get_stream_ldp, /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index dc5b27b25d44..4f0e365597be 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1095,3 +1095,34 @@ u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, return ((u64)llp_u << 32) | llp_l; } + +/** + * hda_dsp_get_stream_ldp - Retrieve the LDP (Linear DMA Position) of the stream + * @sdev: SOF device + * @component: ASoC component + * @substream: PCM substream + * + * Returns the raw Linear Link Position value + */ +u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + u32 ldp_l, ldp_u; + + /* + * The pphc_addr have been calculated during probe in + * hda_dsp_stream_init(): + * pphc_addr = sdev->bar[HDA_DSP_PP_BAR] + + * SOF_HDA_PPHC_BASE + + * SOF_HDA_PPHC_INTERVAL * stream_index + * + * Use this pre-calculated address to avoid repeated re-calculation. + */ + ldp_l = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPL); + ldp_u = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPU); + + return ((u64)ldp_u << 32) | ldp_l; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 75c92b1cf897..21c219eb7b83 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -661,6 +661,9 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, struct snd_soc_component *component, struct snd_pcm_substream *substream); +u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); From 8d10ab0cf3a27d90124b337730a4fa52b1768d55 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:05 +0200 Subject: [PATCH 19/40] ASoC: SOF: ipc4-pcm: Use the snd_sof_pcm_get_dai_frame_counter() for pcm_delay mainline inclusion from mainline-v6.9-rc3 Switch to the new callback to retrieve the DAI (link) frame counter. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-9-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 37679a1bd372c8308a3faccf3438c9df642565b3) --- sound/soc/sof/ipc4-pcm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 0f332c8cdbe6..d0795f77cc15 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -897,11 +897,12 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, } /* - * HDaudio links don't support the LLP counter reported by firmware - * the link position is read directly from hardware registers. + * If the LLP counter is not reported by firmware in the SRAM window + * then read the dai (link) position via host accessible means if + * available. */ if (!time_info->llp_offset) { - tmp_ptr = snd_sof_pcm_get_stream_position(sdev, component, substream); + tmp_ptr = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream); if (!tmp_ptr) return 0; } else { From 2036e73117ebb5de1305897eb0e54856a7e12013 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:06 +0200 Subject: [PATCH 20/40] ASoC: SOF: Intel: hda-common-ops: Do not set the get_stream_position callback mainline inclusion from mainline-v6.9-rc3 The get_stream_position has been replaced by get_dai_frame_counter, it should not be set to allow it to be dropped from core code. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-10-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 4ab6c38c664442c1fc9911eb3c5c6953d3dbcca5) --- sound/soc/sof/intel/hda-common-ops.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index e2b37486d8c8..0620536235f4 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -56,7 +56,6 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { .pcm_pointer = hda_dsp_pcm_pointer, .pcm_ack = hda_dsp_pcm_ack, - .get_stream_position = hda_dsp_get_stream_llp, .get_dai_frame_counter = hda_dsp_get_stream_llp, .get_host_byte_counter = hda_dsp_get_stream_ldp, From 6f3d9f35dd962703fb5f5255daee428b2df4d010 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:07 +0200 Subject: [PATCH 21/40] ASoC: SOF: Remove the get_stream_position callback mainline inclusion from mainline-v6.9-rc3 The get_stream_position has been replaced by get_dai_frame_counter and all related code can be dropped form the core. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-11-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 07007b8ac42cffc23043d00e56b0f67a75dc4b22) --- sound/soc/sof/ops.h | 10 ---------- sound/soc/sof/sof-priv.h | 9 --------- 2 files changed, 19 deletions(-) diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 3c8adbbf11a4..5e96748cd33c 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -525,16 +525,6 @@ static inline int snd_sof_pcm_platform_ack(struct snd_sof_dev *sdev, return 0; } -static inline u64 snd_sof_pcm_get_stream_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - if (sof_ops(sdev) && sof_ops(sdev)->get_stream_position) - return sof_ops(sdev)->get_stream_position(sdev, component, substream); - - return 0; -} - static inline u64 snd_sof_pcm_get_dai_frame_counter(struct snd_sof_dev *sdev, struct snd_soc_component *component, diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 4752985d57e2..980671daa0f7 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -261,15 +261,6 @@ struct snd_sof_dsp_ops { /* pcm ack */ int (*pcm_ack)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); /* optional */ - /* - * optional callback to retrieve the link DMA position for the substream - * when the position is not reported in the shared SRAM windows but - * instead from a host-accessible hardware counter. - */ - u64 (*get_stream_position)(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream); /* optional */ - /* * optional callback to retrieve the number of frames left/arrived from/to * the DSP on the DAI side (link/codec/DMIC/etc). From 65d2d0ec7a66a3904b08e416dcf84d1ba8c8e596 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:08 +0200 Subject: [PATCH 22/40] ASoC: SOF: ipc4-pcm: Move struct sof_ipc4_timestamp_info definition locally mainline inclusion from mainline-v6.9-rc3 The sof_ipc4_timestamp_info is only used by ipc4-pcm.c internally, it should not be in a generic header implying that it might be used elsewhere. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-12-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 31d2874d083ba6cc2a4f4b26dab73c3be1c92658) --- sound/soc/sof/ipc4-pcm.c | 14 ++++++++++++++ sound/soc/sof/ipc4-priv.h | 14 -------------- 2 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index d0795f77cc15..2d7295221884 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -15,6 +15,20 @@ #include "ipc4-topology.h" #include "ipc4-fw-reg.h" +/** + * struct sof_ipc4_timestamp_info - IPC4 timestamp info + * @host_copier: the host copier of the pcm stream + * @dai_copier: the dai copier of the pcm stream + * @stream_start_offset: reported by fw in memory window + * @llp_offset: llp offset in memory window + */ +struct sof_ipc4_timestamp_info { + struct sof_ipc4_copier *host_copier; + struct sof_ipc4_copier *dai_copier; + u64 stream_start_offset; + u32 llp_offset; +}; + static int sof_ipc4_set_multi_pipeline_state(struct snd_sof_dev *sdev, u32 state, struct ipc4_pipeline_set_state_data *trigger_list) { diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index 14020c1ebb45..38745b5937ad 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -91,20 +91,6 @@ struct sof_ipc4_fw_data { struct mutex pipeline_state_mutex; /* protect pipeline triggers, ref counts and states */ }; -/** - * struct sof_ipc4_timestamp_info - IPC4 timestamp info - * @host_copier: the host copier of the pcm stream - * @dai_copier: the dai copier of the pcm stream - * @stream_start_offset: reported by fw in memory window - * @llp_offset: llp offset in memory window - */ -struct sof_ipc4_timestamp_info { - struct sof_ipc4_copier *host_copier; - struct sof_ipc4_copier *dai_copier; - u64 stream_start_offset; - u32 llp_offset; -}; - extern const struct sof_ipc_fw_loader_ops ipc4_loader_ops; extern const struct sof_ipc_tplg_ops ipc4_tplg_ops; extern const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops; From 12dfe87716b52d3372868f974ae747bc076e4bae Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:09 +0200 Subject: [PATCH 23/40] ASoC: SOF: ipc4-pcm: Combine the SOF_IPC4_PIPE_PAUSED cases in pcm_trigger mainline inclusion from mainline-v6.9-rc3 The SNDRV_PCM_TRIGGER_PAUSE_PUSH does not need to be a separate case, it can be handled along with STOP and SUSPEND Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-13-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 55ca6ca227bfc5a8d0a0c2c5d6e239777226a604) --- sound/soc/sof/ipc4-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 2d7295221884..4e41b16d3205 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -478,14 +478,12 @@ static int sof_ipc4_pcm_trigger(struct snd_soc_component *component, /* determine the pipeline state */ switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - state = SOF_IPC4_PIPE_PAUSED; - break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: state = SOF_IPC4_PIPE_RUNNING; break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: state = SOF_IPC4_PIPE_PAUSED; From 0055b86700e659dbb36e1fbbe7bf49ac8143af43 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:10 +0200 Subject: [PATCH 24/40] ASoC: SOF: ipc4-pcm: Invalidate the stream_start_offset in PAUSED state mainline inclusion from mainline-v6.9-rc3 When the final state is SOF_IPC4_PIPE_PAUSED, it is possible that the stream will be restarted (resume or start) in which case we need to update the offset from the firmware. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-14-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 3ce3bc36d91510389955b47e36ea4c4e94fcbdd3) --- sound/soc/sof/ipc4-pcm.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 4e41b16d3205..905dbc4852b1 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -437,8 +437,19 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, } /* return if this is the final state */ - if (state == SOF_IPC4_PIPE_PAUSED) + if (state == SOF_IPC4_PIPE_PAUSED) { + struct sof_ipc4_timestamp_info *time_info; + + /* + * Invalidate the stream_start_offset to make sure that it is + * going to be updated if the stream resumes + */ + time_info = spcm->stream[substream->stream].private; + if (time_info) + time_info->stream_start_offset = SOF_IPC4_INVALID_STREAM_POSITION; + goto free; + } skip_pause_transition: /* else set the RUNNING/RESET state in the DSP */ ret = sof_ipc4_set_multi_pipeline_state(sdev, state, trigger_list); From 72ec7027c743f2ee6a879b333d3d727bd7c2c9d2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:11 +0200 Subject: [PATCH 25/40] ASoC: SOF: sof-pcm: Add pointer callback to sof_ipc_pcm_ops mainline inclusion from mainline-v6.9-rc3 The IPC specific pointer callback can be used when additional or custom handling is needed during the pointer calculation, like executing a delay calculation at the same time to minimize drift between the reported pointer and the calculated delay. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-15-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 77165bd955d55114028b06787a530b8f9220e4b0) --- sound/soc/sof/pcm.c | 8 ++++++++ sound/soc/sof/sof-audio.h | 8 +++++++- 2 files changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 2f86aef4bc9c..8804e00e7251 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -387,13 +387,21 @@ static snd_pcm_uframes_t sof_pcm_pointer(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + const struct sof_ipc_pcm_ops *pcm_ops = sof_ipc_get_ops(sdev, pcm); struct snd_sof_pcm *spcm; snd_pcm_uframes_t host, dai; + int ret = -EOPNOTSUPP; /* nothing to do for BE */ if (rtd->dai_link->no_pcm) return 0; + if (pcm_ops && pcm_ops->pointer) + ret = pcm_ops->pointer(component, substream, &host); + + if (ret != -EOPNOTSUPP) + return ret ? ret : host; + /* use dsp ops pointer callback directly if set */ if (sof_ops(sdev)->pcm_pointer) return sof_ops(sdev)->pcm_pointer(sdev, substream); diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index d302ccc811f4..6f0e06bdb875 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -103,7 +103,10 @@ struct snd_sof_dai_config_data { * additional memory in the SOF PCM stream structure * @pcm_free: Function pointer for PCM free that can be used for freeing any * additional memory in the SOF PCM stream structure - * @delay: Function pointer for pcm delay calculation + * @pointer: Function pointer for pcm pointer + * Note: the @pointer callback may return -EOPNOTSUPP which should be + * handled in a same way as if the callback is not provided + * @delay: Function pointer for pcm delay reporting * @reset_hw_params_during_stop: Flag indicating whether the hw_params should be reset during the * STOP pcm trigger * @ipc_first_on_start: Send IPC before invoking platform trigger during @@ -125,6 +128,9 @@ struct sof_ipc_pcm_ops { int (*dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); int (*pcm_setup)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm); void (*pcm_free)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm); + int (*pointer)(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + snd_pcm_uframes_t *pointer); snd_pcm_sframes_t (*delay)(struct snd_soc_component *component, struct snd_pcm_substream *substream); bool reset_hw_params_during_stop; From 56c53686851c333fd96038456ecb3a30529db043 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:12 +0200 Subject: [PATCH 26/40] ASoC: SOF: ipc4-pcm: Correct the delay calculation mainline inclusion from mainline-v6.9-rc3 This patch improves the delay calculation by relying on the LLP (Linear Link Position) on the DAI side and the LDP (Linear Data Pointer) on the host side. The LDP provides the same DMA position as LPIB, but with a linear count instead of a position in the ALSA ring buffer. The LDP values are provided in bytes and must be converted to frames. The difference in units means that the host counter will wrap earlier than the LLP. We need to wrap the LLP at the same boundary as the host counter. The ASoC framework relies on separate pointer and delay callback. Measurement errors can be reduced by processing all the counter values in the pointer callback. The delay value is stored, and will be reported to higher levels in the delay callback. For playback, the firmware provides a stream_start offset to handle mixing/pause usages, where the DAI might have started earlier than the PCM device. The delay calculation must be special-cased when the link counter has not reached the start offset value, i.e. no valid audio has left the DSP. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-16-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 0ea06680dfcb4464ac6c05968433d060efb44345) --- sound/soc/sof/ipc4-pcm.c | 159 +++++++++++++++++++++++++++++++-------- 1 file changed, 127 insertions(+), 32 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 905dbc4852b1..e915f9f87a6c 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -19,14 +19,22 @@ * struct sof_ipc4_timestamp_info - IPC4 timestamp info * @host_copier: the host copier of the pcm stream * @dai_copier: the dai copier of the pcm stream - * @stream_start_offset: reported by fw in memory window + * @stream_start_offset: reported by fw in memory window (converted to frames) + * @stream_end_offset: reported by fw in memory window (converted to frames) * @llp_offset: llp offset in memory window + * @boundary: wrap boundary should be used for the LLP frame counter + * @delay: Calculated and stored in pointer callback. The stored value is + * returned in the delay callback. */ struct sof_ipc4_timestamp_info { struct sof_ipc4_copier *host_copier; struct sof_ipc4_copier *dai_copier; u64 stream_start_offset; + u64 stream_end_offset; u32 llp_offset; + + u64 boundary; + snd_pcm_sframes_t delay; }; static int sof_ipc4_set_multi_pipeline_state(struct snd_sof_dev *sdev, u32 state, @@ -726,6 +734,10 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm if (abi_version < SOF_IPC4_FW_REGS_ABI_VER) support_info = false; + /* For delay reporting the get_host_byte_counter callback is needed */ + if (!sof_ops(sdev) || !sof_ops(sdev)->get_host_byte_counter) + support_info = false; + for_each_pcm_streams(stream) { pipeline_list = &spcm->stream[stream].pipeline_list; @@ -858,7 +870,6 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev, struct sof_ipc4_copier *host_copier = time_info->host_copier; struct sof_ipc4_copier *dai_copier = time_info->dai_copier; struct sof_ipc4_pipeline_registers ppl_reg; - u64 stream_start_position; u32 dai_sample_size; u32 ch, node_index; u32 offset; @@ -875,38 +886,51 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev, if (ppl_reg.stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) return -EINVAL; - stream_start_position = ppl_reg.stream_start_offset; ch = dai_copier->data.out_format.fmt_cfg; ch = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(ch); dai_sample_size = (dai_copier->data.out_format.bit_depth >> 3) * ch; - /* convert offset to sample count */ - do_div(stream_start_position, dai_sample_size); - time_info->stream_start_offset = stream_start_position; + + /* convert offsets to frame count */ + time_info->stream_start_offset = ppl_reg.stream_start_offset; + do_div(time_info->stream_start_offset, dai_sample_size); + time_info->stream_end_offset = ppl_reg.stream_end_offset; + do_div(time_info->stream_end_offset, dai_sample_size); + + /* + * Calculate the wrap boundary need to be used for delay calculation + * The host counter is in bytes, it will wrap earlier than the frames + * based link counter. + */ + time_info->boundary = div64_u64(~((u64)0), + frames_to_bytes(substream->runtime, 1)); + /* Initialize the delay value to 0 (no delay) */ + time_info->delay = 0; return 0; } -static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, - struct snd_pcm_substream *substream) +static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + snd_pcm_uframes_t *pointer) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sof_ipc4_timestamp_info *time_info; struct sof_ipc4_llp_reading_slot llp; - snd_pcm_uframes_t head_ptr, tail_ptr; + snd_pcm_uframes_t head_cnt, tail_cnt; struct snd_sof_pcm_stream *stream; + u64 dai_cnt, host_cnt, host_ptr; struct snd_sof_pcm *spcm; - u64 tmp_ptr; int ret; spcm = snd_sof_find_spcm_dai(component, rtd); if (!spcm) - return 0; + return -EOPNOTSUPP; stream = &spcm->stream[substream->stream]; time_info = stream->private; if (!time_info) - return 0; + return -EOPNOTSUPP; /* * stream_start_offset is updated to memory window by FW based on @@ -916,46 +940,116 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, if (time_info->stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) { ret = sof_ipc4_get_stream_start_offset(sdev, substream, stream, time_info); if (ret < 0) - return 0; + return -EOPNOTSUPP; } + /* For delay calculation we need the host counter */ + host_cnt = snd_sof_pcm_get_host_byte_counter(sdev, component, substream); + host_ptr = host_cnt; + + /* convert the host_cnt to frames */ + host_cnt = div64_u64(host_cnt, frames_to_bytes(substream->runtime, 1)); + /* * If the LLP counter is not reported by firmware in the SRAM window - * then read the dai (link) position via host accessible means if + * then read the dai (link) counter via host accessible means if * available. */ if (!time_info->llp_offset) { - tmp_ptr = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream); - if (!tmp_ptr) - return 0; + dai_cnt = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream); + if (!dai_cnt) + return -EOPNOTSUPP; } else { sof_mailbox_read(sdev, time_info->llp_offset, &llp, sizeof(llp)); - tmp_ptr = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l; + dai_cnt = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l; } + dai_cnt += time_info->stream_end_offset; - /* In two cases dai dma position is not accurate + /* In two cases dai dma counter is not accurate * (1) dai pipeline is started before host pipeline - * (2) multiple streams mixed into one. Each stream has the same dai dma position + * (2) multiple streams mixed into one. Each stream has the same dai dma + * counter + * + * Firmware calculates correct stream_start_offset for all cases + * including above two. + * Driver subtracts stream_start_offset from dai dma counter to get + * accurate one + */ + + /* + * On stream start the dai counter might not yet have reached the + * stream_start_offset value which means that no frames have left the + * DSP yet from the audio stream (on playback, capture streams have + * offset of 0 as we start capturing right away). + * In this case we need to adjust the distance between the counters by + * increasing the host counter by (offset - dai_counter). + * Otherwise the dai_counter needs to be adjusted to reflect the number + * of valid frames passed on the DAI side. * - * Firmware calculates correct stream_start_offset for all cases including above two. - * Driver subtracts stream_start_offset from dai dma position to get accurate one + * The delay is the difference between the counters on the two + * sides of the DSP. */ - tmp_ptr -= time_info->stream_start_offset; + if (dai_cnt < time_info->stream_start_offset) { + host_cnt += time_info->stream_start_offset - dai_cnt; + dai_cnt = 0; + } else { + dai_cnt -= time_info->stream_start_offset; + } + + /* Wrap the dai counter at the boundary where the host counter wraps */ + div64_u64_rem(dai_cnt, time_info->boundary, &dai_cnt); - /* Calculate the delay taking into account that both pointer can wrap */ - div64_u64_rem(tmp_ptr, substream->runtime->boundary, &tmp_ptr); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - head_ptr = substream->runtime->status->hw_ptr; - tail_ptr = tmp_ptr; + head_cnt = host_cnt; + tail_cnt = dai_cnt; } else { - head_ptr = tmp_ptr; - tail_ptr = substream->runtime->status->hw_ptr; + head_cnt = dai_cnt; + tail_cnt = host_cnt; + } + + if (head_cnt < tail_cnt) { + time_info->delay = time_info->boundary - tail_cnt + head_cnt; + goto out; } - if (head_ptr < tail_ptr) - return substream->runtime->boundary - tail_ptr + head_ptr; + time_info->delay = head_cnt - tail_cnt; + +out: + /* + * Convert the host byte counter to PCM pointer which wraps in buffer + * and it is in frames + */ + div64_u64_rem(host_ptr, snd_pcm_lib_buffer_bytes(substream), &host_ptr); + *pointer = bytes_to_frames(substream->runtime, host_ptr); + + return 0; +} + +static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct sof_ipc4_timestamp_info *time_info; + struct snd_sof_pcm_stream *stream; + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return 0; + + stream = &spcm->stream[substream->stream]; + time_info = stream->private; + /* + * Report the stored delay value calculated in the pointer callback. + * In the unlikely event that the calculation was skipped/aborted, the + * default 0 delay returned. + */ + if (time_info) + return time_info->delay; + + /* No delay information available, report 0 as delay */ + return 0; - return head_ptr - tail_ptr; } const struct sof_ipc_pcm_ops ipc4_pcm_ops = { @@ -965,6 +1059,7 @@ const struct sof_ipc_pcm_ops ipc4_pcm_ops = { .dai_link_fixup = sof_ipc4_pcm_dai_link_fixup, .pcm_setup = sof_ipc4_pcm_setup, .pcm_free = sof_ipc4_pcm_free, + .pointer = sof_ipc4_pcm_pointer, .delay = sof_ipc4_pcm_delay, .ipc_first_on_start = true, .platform_stop_during_hw_free = true, From ef03151ee7b89adf796bc6452ccca80785137427 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:13 +0200 Subject: [PATCH 27/40] ALSA: hda: Add pplcllpl/u members to hdac_ext_stream mainline inclusion from mainline-v6.9-rc3 The pplcllpl/u can be used to save the Link Connection Linear Link Position register value to be used for compensation of the LLP register value in case the counter is not reset (after pause/resume or stop/start without closing the stream). The LLP can be used along with PPHCLDP to calculate delay caused by the DSP processing for HDA links. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-17-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit f9eeb6bb13fb5d7af1ea5b74a10b1f8ead962540) --- include/sound/hdaudio_ext.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 511211f4a2b6..e8b2fcc0df93 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -56,6 +56,9 @@ struct hdac_ext_stream { u32 pphcldpl; u32 pphcldpu; + u32 pplcllpl; + u32 pplcllpu; + bool decoupled:1; bool link_locked:1; bool link_prepared; From 2450da77490c6abfe6aecc0de59decefe151d270 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:14 +0200 Subject: [PATCH 28/40] ASoC: SOF: Intel: hda: Compensate LLP in case it is not reset mainline inclusion from mainline-v6.9-rc3 During pause/reset or stop/start the LLP counter is not reset, which will result broken delay reporting. Read the LLP value on STOP/PAUSE trigger and use it in LLP reading to normalize the LLP from the register. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-18-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 1abc2642588e06f6180b3fbb21968cf5d0ba9e5f) --- sound/soc/sof/intel/hda-dai-ops.c | 11 +++++++++++ sound/soc/sof/intel/hda-pcm.c | 8 ++++++++ sound/soc/sof/intel/hda-stream.c | 9 ++++++++- 3 files changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 9f67f6546e75..f7190ee88f43 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -7,6 +7,7 @@ #include #include +#include #include #include #include @@ -361,6 +362,16 @@ static int hda_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_hdac_ext_stream_clear(hext_stream); + + /* + * Save the LLP registers in case the stream is + * restarting due PAUSE_RELEASE, or START without a pcm + * close/open since in this case the LLP register is not reset + * to 0 and the delay calculation will return with invalid + * results. + */ + hext_stream->pplcllpl = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); + hext_stream->pplcllpu = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); break; default: dev_err(sdev->dev, "unknown trigger command %d\n", cmd); diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index c84e9d031d80..8b5fbbc777bd 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -288,6 +288,14 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev, /* binding pcm substream to hda stream */ substream->runtime->private_data = &dsp_stream->hstream; + + /* + * Reset the llp cache values (they are used for LLP compensation in + * case the counter is not reset) + */ + dsp_stream->pplcllpl = 0; + dsp_stream->pplcllpu = 0; + return 0; } diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 4f0e365597be..51d85485ed2e 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1064,6 +1064,8 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, return pos; } +#define merge_u64(u32_u, u32_l) (((u64)(u32_u) << 32) | (u32_l)) + /** * hda_dsp_get_stream_llp - Retrieve the LLP (Linear Link Position) of the stream * @sdev: SOF device @@ -1093,7 +1095,12 @@ u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, llp_l = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); llp_u = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); - return ((u64)llp_u << 32) | llp_l; + /* Compensate the LLP counter with the saved offset */ + if (hext_stream->pplcllpl || hext_stream->pplcllpu) + return merge_u64(llp_u, llp_l) - + merge_u64(hext_stream->pplcllpu, hext_stream->pplcllpl); + + return merge_u64(llp_u, llp_l); } /** From 2b3a607111737402e666277d8f4cae527ab1f1c7 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 14:00:33 +0300 Subject: [PATCH 29/40] ASoC: SOF: ipc4-pcm: Use consistent name for snd_sof_pcm_stream pointer mainline inclusion from mainline-v6.9-rc7 Throughout the file the pointer for snd_sof_pcm_stream is named either 'stream' or (wrongly) 'spcm' which confuses the reader. Use 'sps' for the pointer name as it is the most common name used in SOF codebase. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240409110036.9411-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 965e49cdf8c19f21b8308adeded3a8139cff5c84) --- sound/soc/sof/ipc4-pcm.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index e915f9f87a6c..f989cc2992bf 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -764,7 +764,7 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm return 0; } -static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *spcm) +static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sps) { struct sof_ipc4_copier *host_copier = NULL; struct sof_ipc4_copier *dai_copier = NULL; @@ -775,7 +775,7 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc int i; /* find host & dai to locate info in memory window */ - for_each_dapm_widgets(spcm->list, i, widget) { + for_each_dapm_widgets(sps->list, i, widget) { struct snd_sof_widget *swidget = widget->dobj.private; if (!swidget) @@ -795,7 +795,7 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc return; } - info = spcm->private; + info = sps->private; info->host_copier = host_copier; info->dai_copier = dai_copier; info->llp_offset = offsetof(struct sof_ipc4_fw_registers, llp_gpdma_reading_slots) + @@ -864,7 +864,7 @@ static int sof_ipc4_pcm_hw_params(struct snd_soc_component *component, static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, - struct snd_sof_pcm_stream *stream, + struct snd_sof_pcm_stream *sps, struct sof_ipc4_timestamp_info *time_info) { struct sof_ipc4_copier *host_copier = time_info->host_copier; @@ -918,7 +918,7 @@ static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, struct sof_ipc4_timestamp_info *time_info; struct sof_ipc4_llp_reading_slot llp; snd_pcm_uframes_t head_cnt, tail_cnt; - struct snd_sof_pcm_stream *stream; + struct snd_sof_pcm_stream *sps; u64 dai_cnt, host_cnt, host_ptr; struct snd_sof_pcm *spcm; int ret; @@ -927,8 +927,8 @@ static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, if (!spcm) return -EOPNOTSUPP; - stream = &spcm->stream[substream->stream]; - time_info = stream->private; + sps = &spcm->stream[substream->stream]; + time_info = sps->private; if (!time_info) return -EOPNOTSUPP; @@ -938,7 +938,7 @@ static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, * the statistics is complete. And it will not change after the first initiailization. */ if (time_info->stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) { - ret = sof_ipc4_get_stream_start_offset(sdev, substream, stream, time_info); + ret = sof_ipc4_get_stream_start_offset(sdev, substream, sps, time_info); if (ret < 0) return -EOPNOTSUPP; } @@ -1030,15 +1030,15 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sof_ipc4_timestamp_info *time_info; - struct snd_sof_pcm_stream *stream; + struct snd_sof_pcm_stream *sps; struct snd_sof_pcm *spcm; spcm = snd_sof_find_spcm_dai(component, rtd); if (!spcm) return 0; - stream = &spcm->stream[substream->stream]; - time_info = stream->private; + sps = &spcm->stream[substream->stream]; + time_info = sps->private; /* * Report the stored delay value calculated in the pointer callback. * In the unlikely event that the calculation was skipped/aborted, the From bf2227b9fd205af000e008f3a7f33732dcd21911 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 14:00:34 +0300 Subject: [PATCH 30/40] ASoC: SOF: ipc4-pcm: Use consistent name for sof_ipc4_timestamp_info pointer mainline inclusion from mainline-v6.9-rc7 The pointer to sof_ipc4_timestamp_info named most of the time as 'time_info' only to be named as 'stream_info' or 'info' in two function. Use the consistent name of 'time_info' throughout the file. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240409110036.9411-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 36e980050b0733829e4e0f97b97f7907ba9f00bb) --- sound/soc/sof/ipc4-pcm.c | 40 ++++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index f989cc2992bf..74b0d0d00270 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -721,7 +721,7 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm { struct snd_sof_pcm_stream_pipeline_list *pipeline_list; struct sof_ipc4_fw_data *ipc4_data = sdev->private; - struct sof_ipc4_timestamp_info *stream_info; + struct sof_ipc4_timestamp_info *time_info; bool support_info = true; u32 abi_version; u32 abi_offset; @@ -752,13 +752,13 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm if (!support_info) continue; - stream_info = kzalloc(sizeof(*stream_info), GFP_KERNEL); - if (!stream_info) { + time_info = kzalloc(sizeof(*time_info), GFP_KERNEL); + if (!time_info) { sof_ipc4_pcm_free(sdev, spcm); return -ENOMEM; } - spcm->stream[stream].private = stream_info; + spcm->stream[stream].private = time_info; } return 0; @@ -769,7 +769,7 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc struct sof_ipc4_copier *host_copier = NULL; struct sof_ipc4_copier *dai_copier = NULL; struct sof_ipc4_llp_reading_slot llp_slot; - struct sof_ipc4_timestamp_info *info; + struct sof_ipc4_timestamp_info *time_info; struct snd_soc_dapm_widget *widget; struct snd_sof_dai *dai; int i; @@ -795,44 +795,44 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc return; } - info = sps->private; - info->host_copier = host_copier; - info->dai_copier = dai_copier; - info->llp_offset = offsetof(struct sof_ipc4_fw_registers, llp_gpdma_reading_slots) + - sdev->fw_info_box.offset; + time_info = sps->private; + time_info->host_copier = host_copier; + time_info->dai_copier = dai_copier; + time_info->llp_offset = offsetof(struct sof_ipc4_fw_registers, + llp_gpdma_reading_slots) + sdev->fw_info_box.offset; /* find llp slot used by current dai */ for (i = 0; i < SOF_IPC4_MAX_LLP_GPDMA_READING_SLOTS; i++) { - sof_mailbox_read(sdev, info->llp_offset, &llp_slot, sizeof(llp_slot)); + sof_mailbox_read(sdev, time_info->llp_offset, &llp_slot, sizeof(llp_slot)); if (llp_slot.node_id == dai_copier->data.gtw_cfg.node_id) break; - info->llp_offset += sizeof(llp_slot); + time_info->llp_offset += sizeof(llp_slot); } if (i < SOF_IPC4_MAX_LLP_GPDMA_READING_SLOTS) return; /* if no llp gpdma slot is used, check aggregated sdw slot */ - info->llp_offset = offsetof(struct sof_ipc4_fw_registers, llp_sndw_reading_slots) + - sdev->fw_info_box.offset; + time_info->llp_offset = offsetof(struct sof_ipc4_fw_registers, + llp_sndw_reading_slots) + sdev->fw_info_box.offset; for (i = 0; i < SOF_IPC4_MAX_LLP_SNDW_READING_SLOTS; i++) { - sof_mailbox_read(sdev, info->llp_offset, &llp_slot, sizeof(llp_slot)); + sof_mailbox_read(sdev, time_info->llp_offset, &llp_slot, sizeof(llp_slot)); if (llp_slot.node_id == dai_copier->data.gtw_cfg.node_id) break; - info->llp_offset += sizeof(llp_slot); + time_info->llp_offset += sizeof(llp_slot); } if (i < SOF_IPC4_MAX_LLP_SNDW_READING_SLOTS) return; /* check EVAD slot */ - info->llp_offset = offsetof(struct sof_ipc4_fw_registers, llp_evad_reading_slot) + - sdev->fw_info_box.offset; - sof_mailbox_read(sdev, info->llp_offset, &llp_slot, sizeof(llp_slot)); + time_info->llp_offset = offsetof(struct sof_ipc4_fw_registers, + llp_evad_reading_slot) + sdev->fw_info_box.offset; + sof_mailbox_read(sdev, time_info->llp_offset, &llp_slot, sizeof(llp_slot)); if (llp_slot.node_id != dai_copier->data.gtw_cfg.node_id) - info->llp_offset = 0; + time_info->llp_offset = 0; } static int sof_ipc4_pcm_hw_params(struct snd_soc_component *component, From 464be2bfaeeb6a8d5783767f1b06b400a08d2542 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 14:00:35 +0300 Subject: [PATCH 31/40] ASoC: SOF: ipc4-pcm: Introduce generic sof_ipc4_pcm_stream_priv mainline inclusion from mainline-v6.9-rc7 Using the sof_ipc4_timestamp_info struct directly as sps->private data is too restrictive, add a new generic sof_ipc4_pcm_stream_priv struct containing the time_info to allow new information to be stored in a generic way. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240409110036.9411-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 551af3280c16166244425bbb1d73028f3a907e1f) --- sound/soc/sof/ipc4-pcm.c | 43 ++++++++++++++++++++++++++++++++-------- 1 file changed, 35 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 74b0d0d00270..34ce6bb7f37d 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -37,6 +37,22 @@ struct sof_ipc4_timestamp_info { snd_pcm_sframes_t delay; }; +/** + * struct sof_ipc4_pcm_stream_priv - IPC4 specific private data + * @time_info: pointer to time info struct if it is supported, otherwise NULL + */ +struct sof_ipc4_pcm_stream_priv { + struct sof_ipc4_timestamp_info *time_info; +}; + +static inline struct sof_ipc4_timestamp_info * +sof_ipc4_sps_to_time_info(struct snd_sof_pcm_stream *sps) +{ + struct sof_ipc4_pcm_stream_priv *stream_priv = sps->private; + + return stream_priv->time_info; +} + static int sof_ipc4_set_multi_pipeline_state(struct snd_sof_dev *sdev, u32 state, struct ipc4_pipeline_set_state_data *trigger_list) { @@ -452,7 +468,7 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, * Invalidate the stream_start_offset to make sure that it is * going to be updated if the stream resumes */ - time_info = spcm->stream[substream->stream].private; + time_info = sof_ipc4_sps_to_time_info(&spcm->stream[substream->stream]); if (time_info) time_info->stream_start_offset = SOF_IPC4_INVALID_STREAM_POSITION; @@ -706,12 +722,16 @@ static int sof_ipc4_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, static void sof_ipc4_pcm_free(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm) { struct snd_sof_pcm_stream_pipeline_list *pipeline_list; + struct sof_ipc4_pcm_stream_priv *stream_priv; int stream; for_each_pcm_streams(stream) { pipeline_list = &spcm->stream[stream].pipeline_list; kfree(pipeline_list->pipelines); pipeline_list->pipelines = NULL; + + stream_priv = spcm->stream[stream].private; + kfree(stream_priv->time_info); kfree(spcm->stream[stream].private); spcm->stream[stream].private = NULL; } @@ -721,6 +741,7 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm { struct snd_sof_pcm_stream_pipeline_list *pipeline_list; struct sof_ipc4_fw_data *ipc4_data = sdev->private; + struct sof_ipc4_pcm_stream_priv *stream_priv; struct sof_ipc4_timestamp_info *time_info; bool support_info = true; u32 abi_version; @@ -749,6 +770,14 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm return -ENOMEM; } + stream_priv = kzalloc(sizeof(*stream_priv), GFP_KERNEL); + if (!stream_priv) { + sof_ipc4_pcm_free(sdev, spcm); + return -ENOMEM; + } + + spcm->stream[stream].private = stream_priv; + if (!support_info) continue; @@ -758,7 +787,7 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm return -ENOMEM; } - spcm->stream[stream].private = time_info; + stream_priv->time_info = time_info; } return 0; @@ -795,7 +824,7 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc return; } - time_info = sps->private; + time_info = sof_ipc4_sps_to_time_info(sps); time_info->host_copier = host_copier; time_info->dai_copier = dai_copier; time_info->llp_offset = offsetof(struct sof_ipc4_fw_registers, @@ -849,7 +878,7 @@ static int sof_ipc4_pcm_hw_params(struct snd_soc_component *component, if (!spcm) return -EINVAL; - time_info = spcm->stream[substream->stream].private; + time_info = sof_ipc4_sps_to_time_info(&spcm->stream[substream->stream]); /* delay calculation is not supported by current fw_reg ABI */ if (!time_info) return 0; @@ -928,7 +957,7 @@ static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, return -EOPNOTSUPP; sps = &spcm->stream[substream->stream]; - time_info = sps->private; + time_info = sof_ipc4_sps_to_time_info(sps); if (!time_info) return -EOPNOTSUPP; @@ -1030,15 +1059,13 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sof_ipc4_timestamp_info *time_info; - struct snd_sof_pcm_stream *sps; struct snd_sof_pcm *spcm; spcm = snd_sof_find_spcm_dai(component, rtd); if (!spcm) return 0; - sps = &spcm->stream[substream->stream]; - time_info = sps->private; + time_info = sof_ipc4_sps_to_time_info(&spcm->stream[substream->stream]); /* * Report the stored delay value calculated in the pointer callback. * In the unlikely event that the calculation was skipped/aborted, the From 9b97560508aaea0eecd41b1fc6d324070ab61630 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 14:00:36 +0300 Subject: [PATCH 32/40] ASoC: SOF: ipc4-pcm: Do not reset the ChainDMA if it has not been allocated mainline inclusion from mainline-v6.9-rc7 The ChainDMA operation differs from normal pipelines that it is only created when the stream started, in fact a PCM using ChainDMA has no pipelines, modules. To reset a ChainDMA, it needs to be first allocated in firmware. When PulseAudio/PipeWire starts, they will probe the PCMs by opening them, check hw_params and then close the PCM without starting audio. Unconditionally resetting the ChainDMA can result the following error: ipc tx : 0xe040000|0x0: GLB_CHAIN_DMA ipc tx reply: 0x2e000007|0x0: GLB_CHAIN_DMA FW reported error: 7 - Unsupported operation requested ipc error for msg 0xe040000|0x0 sof_pcm_stream_free: pcm_ops hw_free failed -22 Add a new chain_dma_allocated flag to sof_ipc4_pcm_stream_priv to store the ChainDMA allocation state and use this flag to skip sending the reset if the ChainDMA is not allocated. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240409110036.9411-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 7211814f2adcf376b8db6321447a9725c33b6ae7) --- sound/soc/sof/ipc4-pcm.c | 24 ++++++++++++++++++++---- 1 file changed, 20 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 34ce6bb7f37d..4594470ed08b 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -40,9 +40,12 @@ struct sof_ipc4_timestamp_info { /** * struct sof_ipc4_pcm_stream_priv - IPC4 specific private data * @time_info: pointer to time info struct if it is supported, otherwise NULL + * @chain_dma_allocated: indicates the ChainDMA allocation state */ struct sof_ipc4_pcm_stream_priv { struct sof_ipc4_timestamp_info *time_info; + + bool chain_dma_allocated; }; static inline struct sof_ipc4_timestamp_info * @@ -269,14 +272,17 @@ sof_ipc4_update_pipeline_state(struct snd_sof_dev *sdev, int state, int cmd, */ static int sof_ipc4_chain_dma_trigger(struct snd_sof_dev *sdev, - int direction, + struct snd_sof_pcm *spcm, int direction, struct snd_sof_pcm_stream_pipeline_list *pipeline_list, int state, int cmd) { struct sof_ipc4_fw_data *ipc4_data = sdev->private; + struct sof_ipc4_pcm_stream_priv *stream_priv; bool allocate, enable, set_fifo_size; struct sof_ipc4_msg msg = {{ 0 }}; - int i; + int ret, i; + + stream_priv = spcm->stream[direction].private; switch (state) { case SOF_IPC4_PIPE_RUNNING: /* Allocate and start chained dma */ @@ -297,6 +303,11 @@ static int sof_ipc4_chain_dma_trigger(struct snd_sof_dev *sdev, set_fifo_size = false; break; case SOF_IPC4_PIPE_RESET: /* Disable and free chained DMA. */ + + /* ChainDMA can only be reset if it has been allocated */ + if (!stream_priv->chain_dma_allocated) + return 0; + allocate = false; enable = false; set_fifo_size = false; @@ -354,7 +365,12 @@ static int sof_ipc4_chain_dma_trigger(struct snd_sof_dev *sdev, if (enable) msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_ENABLE_MASK; - return sof_ipc_tx_message_no_reply(sdev->ipc, &msg, 0); + ret = sof_ipc_tx_message_no_reply(sdev->ipc, &msg, 0); + /* Update the ChainDMA allocation state */ + if (!ret) + stream_priv->chain_dma_allocated = allocate; + + return ret; } static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, @@ -394,7 +410,7 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, * trigger function that handles the rest for the substream. */ if (pipeline->use_chain_dma) - return sof_ipc4_chain_dma_trigger(sdev, substream->stream, + return sof_ipc4_chain_dma_trigger(sdev, spcm, substream->stream, pipeline_list, state, cmd); /* allocate memory for the pipeline data */ From 875e9e18bc78b80789a487c1083badfdcc3ecf49 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 08:32:51 -0500 Subject: [PATCH 33/40] ASoC: SOF: Intel: discard SoundWire configuration if HDaudio codec is reported MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit mainline inclusion from mainline-v6.10-rc1 The machine driver and topology selection starts with I2S, then SoundWire and last uses HDaudio as a fallback. That assumes that the ACPI information is correct but there are of course exceptions to the rule. On a Lenovo platform, an external HDaudio codec is detected, but the ACPI tables expose TWO RT711 jack codecs. This patch skips the SoundWire selection in case an external HDaudio codec is detected - which only works with the additional assumption that no one will mix HDaudio and SoundWire. Closes: https://github.com/thesofproject/linux/issues/4962 Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240503133253.108201-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 5a7543d0ca01d68d992f480d151efd693807e0ce) --- sound/soc/sof/intel/hda.c | 9 +++++++-- sound/soc/sof/intel/hda.h | 2 ++ 2 files changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 8401b9e4dba6..ed4180b08469 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1666,6 +1666,7 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) u32 interface_mask = hda_get_interface_mask(sdev); struct snd_sof_pdata *sof_pdata = sdev->pdata; const struct sof_dev_desc *desc = sof_pdata->desc; + struct hdac_bus *bus = sof_to_bus(sdev); struct snd_soc_acpi_mach *mach = NULL; const char *tplg_filename; @@ -1779,8 +1780,12 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) } } - /* If I2S fails, try SoundWire if it is supported */ - if (!mach && (interface_mask & BIT(SOF_DAI_INTEL_ALH))) + /* + * If I2S fails and no external HDaudio codec is detected, + * try SoundWire if it is supported + */ + if (!mach && !HDA_EXT_CODEC(bus->codec_mask) && + (interface_mask & BIT(SOF_DAI_INTEL_ALH))) mach = hda_sdw_machine_select(sdev); /* diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 21c219eb7b83..94bef8762965 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -450,6 +450,8 @@ #define SSP_SET_SFRM_CONSUMER BIT(24) #define SSP_SET_CBP_CFP (SSP_SET_SCLK_CONSUMER | SSP_SET_SFRM_CONSUMER) +#define HDA_EXT_ADDR 0 +#define HDA_EXT_CODEC(x) ((x) & BIT(HDA_EXT_ADDR)) #define HDA_IDISP_ADDR 2 #define HDA_IDISP_CODEC(x) ((x) & BIT(HDA_IDISP_ADDR)) From aba04aea4109261622a88684d6ba4b8352a64c4c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 May 2024 08:32:52 -0500 Subject: [PATCH 34/40] ASoC: SOF: ipc4-topology: Allow selective update in sof_ipc4_update_hw_params mainline inclusion from mainline-v6.10-rc1 Add a bitmask parameter to sof_ipc4_update_hw_params() to be able to select the param to be updated. This feature can be used when not all params should be updated, for example if caller only wants to update the format in the params, leaving the channels and rates untouched. Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240503133253.108201-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit b679302526d637700e998da85ed1e06d8f7d3674) --- sound/soc/sof/ipc4-topology.c | 83 +++++++++++++++++++++-------------- 1 file changed, 51 insertions(+), 32 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 1229c4ec6110..add26b92e713 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1053,42 +1053,50 @@ static int sof_ipc4_widget_assign_instance_id(struct snd_sof_dev *sdev, /* update hw_params based on the audio stream format */ static int sof_ipc4_update_hw_params(struct snd_sof_dev *sdev, struct snd_pcm_hw_params *params, - struct sof_ipc4_audio_format *fmt) + struct sof_ipc4_audio_format *fmt, u32 param_to_update) { - snd_pcm_format_t snd_fmt; struct snd_interval *i; - struct snd_mask *m; - int valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg); - unsigned int channels, rate; - switch (valid_bits) { - case 16: - snd_fmt = SNDRV_PCM_FORMAT_S16_LE; - break; - case 24: - snd_fmt = SNDRV_PCM_FORMAT_S24_LE; - break; - case 32: - snd_fmt = SNDRV_PCM_FORMAT_S32_LE; - break; - default: - dev_err(sdev->dev, "invalid PCM valid_bits %d\n", valid_bits); - return -EINVAL; + if (param_to_update & BIT(SNDRV_PCM_HW_PARAM_FORMAT)) { + int valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg); + snd_pcm_format_t snd_fmt; + struct snd_mask *m; + + switch (valid_bits) { + case 16: + snd_fmt = SNDRV_PCM_FORMAT_S16_LE; + break; + case 24: + snd_fmt = SNDRV_PCM_FORMAT_S24_LE; + break; + case 32: + snd_fmt = SNDRV_PCM_FORMAT_S32_LE; + break; + default: + dev_err(sdev->dev, "invalid PCM valid_bits %d\n", valid_bits); + return -EINVAL; + } + + m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(m); + snd_mask_set_format(m, snd_fmt); } - m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - snd_mask_none(m); - snd_mask_set_format(m, snd_fmt); + if (param_to_update & BIT(SNDRV_PCM_HW_PARAM_RATE)) { + unsigned int rate = fmt->sampling_frequency; - rate = fmt->sampling_frequency; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - i->min = rate; - i->max = rate; + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + i->min = rate; + i->max = rate; + } - channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg); - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - i->min = channels; - i->max = channels; + if (param_to_update & BIT(SNDRV_PCM_HW_PARAM_CHANNELS)) { + unsigned int channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + i->min = channels; + i->max = channels; + } return 0; } @@ -1770,7 +1778,11 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, } /* modify the input params for the next widget */ - ret = sof_ipc4_update_hw_params(sdev, pipeline_params, &copier_data->out_format); + ret = sof_ipc4_update_hw_params(sdev, pipeline_params, + &copier_data->out_format, + BIT(SNDRV_PCM_HW_PARAM_FORMAT) | + BIT(SNDRV_PCM_HW_PARAM_CHANNELS) | + BIT(SNDRV_PCM_HW_PARAM_RATE)); if (ret) return ret; @@ -1991,7 +2003,10 @@ static int sof_ipc4_prepare_src_module(struct snd_sof_widget *swidget, src->data.sink_rate = out_audio_fmt->sampling_frequency; /* update pipeline_params for sink widgets */ - return sof_ipc4_update_hw_params(sdev, pipeline_params, out_audio_fmt); + return sof_ipc4_update_hw_params(sdev, pipeline_params, out_audio_fmt, + BIT(SNDRV_PCM_HW_PARAM_FORMAT) | + BIT(SNDRV_PCM_HW_PARAM_CHANNELS) | + BIT(SNDRV_PCM_HW_PARAM_RATE)); } static int @@ -2115,7 +2130,11 @@ static int sof_ipc4_prepare_process_module(struct snd_sof_widget *swidget, sizeof(struct sof_ipc4_audio_format)); /* modify the pipeline params with the pin 0 output format */ - ret = sof_ipc4_update_hw_params(sdev, pipeline_params, &process->output_format); + ret = sof_ipc4_update_hw_params(sdev, pipeline_params, + &process->output_format, + BIT(SNDRV_PCM_HW_PARAM_FORMAT) | + BIT(SNDRV_PCM_HW_PARAM_CHANNELS) | + BIT(SNDRV_PCM_HW_PARAM_RATE)); if (ret) return ret; } From 34795df24564d5c03aad64f9845009b2d153b4f5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 May 2024 08:32:53 -0500 Subject: [PATCH 35/40] ASoC: SOF: ipc4-topology: Correct DAI copier config and NHLT blob request mainline inclusion from mainline-v6.10-rc1 In case of capture and when the DAI copier have single bit depth supported on it's input side we should use this format instead of the one in fe_params. Regardless of the stream direction for the NHLT blob lookup when the DAI copier only supports single bit depth on the DAI side we should only look for a blob which matches with this single configuration. For DMIC if the DAI copier supports multiple bit depths, try to request 32-bit blob first if the requested bit depth is 16-bit. If the 32-bit blob is available then look for marching (32-bit) copier format to make sure that both the blob and copier have correct parameters. Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240503133253.108201-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit f9209644ae7688e82f629e737417bc8916db7b57) --- sound/soc/sof/ipc4-topology.c | 162 ++++++++++++++++++++++++++++------ 1 file changed, 136 insertions(+), 26 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index add26b92e713..99c40ef113ee 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1364,13 +1364,16 @@ static int snd_sof_get_hw_config_params(struct snd_sof_dev *sdev, struct snd_sof return 0; } -static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, - struct snd_pcm_hw_params *params, u32 dai_index, - u32 linktype, u8 dir, u32 **dst, u32 *len) +static int +snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, + bool single_format, + struct snd_pcm_hw_params *params, u32 dai_index, + u32 linktype, u8 dir, u32 **dst, u32 *len) { struct sof_ipc4_fw_data *ipc4_data = sdev->private; struct nhlt_specific_cfg *cfg; int sample_rate, channel_count; + bool format_change = false; int bit_depth, ret; u32 nhlt_type; int dev_type = 0; @@ -1379,9 +1382,18 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s switch (linktype) { case SOF_DAI_INTEL_DMIC: nhlt_type = NHLT_LINK_DMIC; - bit_depth = params_width(params); channel_count = params_channels(params); sample_rate = params_rate(params); + bit_depth = params_width(params); + /* + * Look for 32-bit blob first instead of 16-bit if copier + * supports multiple formats + */ + if (bit_depth == 16 && !single_format) { + dev_dbg(sdev->dev, "Looking for 32-bit blob first for DMIC\n"); + format_change = true; + bit_depth = 32; + } break; case SOF_DAI_INTEL_SSP: nhlt_type = NHLT_LINK_SSP; @@ -1415,22 +1427,56 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s dir, dev_type); if (!cfg) { + if (format_change) { + /* + * The 32-bit blob was not found in NHLT table, try to + * look for one based on the params + */ + bit_depth = params_width(params); + format_change = false; + + cfg = intel_nhlt_get_endpoint_blob(sdev->dev, ipc4_data->nhlt, + dai_index, nhlt_type, + bit_depth, bit_depth, + channel_count, sample_rate, + dir, dev_type); + if (cfg) + goto out; + } + dev_err(sdev->dev, "no matching blob for sample rate: %d sample width: %d channels: %d\n", sample_rate, bit_depth, channel_count); return -EINVAL; } +out: /* config length should be in dwords */ *len = cfg->size >> 2; *dst = (u32 *)cfg->caps; + if (format_change) { + /* + * Update the params to reflect that we have loaded 32-bit blob + * instead of the 16-bit. + * This information is going to be used by the caller to find + * matching copier format on the dai side. + */ + struct snd_mask *m; + + m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(m); + snd_mask_set_format(m, SNDRV_PCM_FORMAT_S32_LE); + } + return 0; } #else -static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, - struct snd_pcm_hw_params *params, u32 dai_index, - u32 linktype, u8 dir, u32 **dst, u32 *len) +static int +snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, + bool single_format, + struct snd_pcm_hw_params *params, u32 dai_index, + u32 linktype, u8 dir, u32 **dst, u32 *len) { return 0; } @@ -1461,6 +1507,68 @@ bool sof_ipc4_copier_is_single_format(struct snd_sof_dev *sdev, return true; } +static int +sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, + struct snd_pcm_hw_params *params, int dir) +{ + struct sof_ipc4_available_audio_format *available_fmt; + struct snd_pcm_hw_params dai_params = *params; + struct sof_ipc4_copier_data *copier_data; + struct sof_ipc4_copier *ipc4_copier; + bool single_format; + int ret; + + ipc4_copier = dai->private; + copier_data = &ipc4_copier->data; + available_fmt = &ipc4_copier->available_fmt; + + /* + * If the copier on the DAI side supports only single bit depth then + * this depth (format) should be used to look for the NHLT blob (if + * needed) and in case of capture this should be used for the input + * format lookup + */ + if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + single_format = sof_ipc4_copier_is_single_format(sdev, + available_fmt->output_pin_fmts, + available_fmt->num_output_formats); + + /* Update the dai_params with the only supported format */ + if (single_format) { + ret = sof_ipc4_update_hw_params(sdev, &dai_params, + &available_fmt->output_pin_fmts[0].audio_fmt, + BIT(SNDRV_PCM_HW_PARAM_FORMAT)); + if (ret) + return ret; + } + } else { + single_format = sof_ipc4_copier_is_single_format(sdev, + available_fmt->input_pin_fmts, + available_fmt->num_input_formats); + + /* Update the dai_params with the only supported format */ + if (single_format) { + ret = sof_ipc4_update_hw_params(sdev, &dai_params, + &available_fmt->input_pin_fmts[0].audio_fmt, + BIT(SNDRV_PCM_HW_PARAM_FORMAT)); + if (ret) + return ret; + } + } + + ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, single_format, + &dai_params, + ipc4_copier->dai_index, + ipc4_copier->dai_type, dir, + &ipc4_copier->copier_config, + &copier_data->gtw_cfg.config_length); + /* Update the params to reflect the changes made in this function */ + if (!ret) + *params = dai_params; + + return ret; +} + static int sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, struct snd_pcm_hw_params *fe_params, @@ -1471,7 +1579,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, struct snd_soc_component *scomp = swidget->scomp; struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct sof_ipc4_copier_data *copier_data; - struct snd_pcm_hw_params *ref_params; + struct snd_pcm_hw_params ref_params; struct sof_ipc4_copier *ipc4_copier; struct snd_sof_dai *dai; u32 gtw_cfg_config_length; @@ -1548,9 +1656,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, * for capture. */ if (dir == SNDRV_PCM_STREAM_PLAYBACK) - ref_params = fe_params; + ref_params = *fe_params; else - ref_params = pipeline_params; + ref_params = *pipeline_params; copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; copier_data->gtw_cfg.node_id |= @@ -1576,23 +1684,25 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, available_fmt = &ipc4_copier->available_fmt; /* - * When there is format conversion within a pipeline, the number of supported - * output formats is typically limited to just 1 for the DAI copiers. But when there - * is no format conversion, the DAI copiers input format must match that of the - * FE hw_params for capture and the pipeline params for playback. + * Use the fe_params as a base for the copier configuration. + * The ref_params might get updated to reflect what format is + * supported by the copier on the DAI side. + * + * In case of capture the ref_params returned will be used to + * find the input configuration of the copier. */ - if (dir == SNDRV_PCM_STREAM_PLAYBACK) - ref_params = pipeline_params; - else - ref_params = fe_params; - - ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, fe_params, ipc4_copier->dai_index, - ipc4_copier->dai_type, dir, - &ipc4_copier->copier_config, - &copier_data->gtw_cfg.config_length); + ref_params = *fe_params; + ret = sof_ipc4_prepare_dai_copier(sdev, dai, &ref_params, dir); if (ret < 0) return ret; + /* + * For playback the pipeline_params needs to be used to find the + * input configuration of the copier. + */ + if (dir == SNDRV_PCM_STREAM_PLAYBACK) + ref_params = *pipeline_params; + break; } case snd_soc_dapm_buffer: @@ -1600,7 +1710,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, ipc4_copier = (struct sof_ipc4_copier *)swidget->private; copier_data = &ipc4_copier->data; available_fmt = &ipc4_copier->available_fmt; - ref_params = pipeline_params; + ref_params = *pipeline_params; break; } @@ -1611,8 +1721,8 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, } /* set input and output audio formats */ - ret = sof_ipc4_init_input_audio_fmt(sdev, swidget, &copier_data->base_config, ref_params, - available_fmt); + ret = sof_ipc4_init_input_audio_fmt(sdev, swidget, &copier_data->base_config, + &ref_params, available_fmt); if (ret < 0) return ret; From 6718dc73f60ee83e004e5ef534525ea8ff2e0979 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 May 2024 14:19:14 +0300 Subject: [PATCH 36/40] ASoC: SOF: ipc4-topology: Add support for NHLT with 16-bit only DMIC blob mainline inclusion from mainline-v6.10-rc2 The ACPI NHLT table always had 32-bit DMIC blob even if 16-bit was also present and taken as a 'rule' which obviously got broken and there is at least one device on the market which ships with only 16-bit DMIC configuration blob. This corner case has never been supported and it is going to need topology updates for DMIC copier to support multiple formats. As for the kernel side: if the copier supports multiple formats and the preferred 32-bit DMIC blob is not found then we will try to get a 16-bit DMIC configuration and look for a 16-bit copier config. Fixes: f9209644ae76 ("ASoC: SOF: ipc4-topology: Correct DAI copier config and NHLT blob request") Link: https://github.com/thesofproject/linux/issues/4973 Signed-off-by: Peter Ujfalusi Reviewed-by: Seppo Ingalsuo Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240530111918.21974-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 49cb894d567980235b6e64d5e69950ff77debd8c) --- sound/soc/sof/ipc4-topology.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 99c40ef113ee..e564e0442186 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1427,6 +1427,8 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai dir, dev_type); if (!cfg) { + bool get_new_blob = false; + if (format_change) { /* * The 32-bit blob was not found in NHLT table, try to @@ -1434,7 +1436,20 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai */ bit_depth = params_width(params); format_change = false; + get_new_blob = true; + } else if (linktype == SOF_DAI_INTEL_DMIC && !single_format) { + /* + * The requested 32-bit blob (no format change for the + * blob request) was not found in NHLT table, try to + * look for 16-bit blob if the copier supports multiple + * formats + */ + bit_depth = 16; + format_change = true; + get_new_blob = true; + } + if (get_new_blob) { cfg = intel_nhlt_get_endpoint_blob(sdev->dev, ipc4_data->nhlt, dai_index, nhlt_type, bit_depth, bit_depth, @@ -1457,8 +1472,8 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai if (format_change) { /* - * Update the params to reflect that we have loaded 32-bit blob - * instead of the 16-bit. + * Update the params to reflect that different blob was loaded + * instead of the requested bit depth (16 -> 32 or 32 -> 16). * This information is going to be used by the caller to find * matching copier format on the dai side. */ @@ -1466,7 +1481,11 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_none(m); - snd_mask_set_format(m, SNDRV_PCM_FORMAT_S32_LE); + if (bit_depth == 16) + snd_mask_set_format(m, SNDRV_PCM_FORMAT_S16_LE); + else + snd_mask_set_format(m, SNDRV_PCM_FORMAT_S32_LE); + } return 0; From 9ed039abaafca6809d0a3de7a8c63327c838b444 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 May 2024 14:19:15 +0300 Subject: [PATCH 37/40] ASoC: SOF: ipc4-topology: Print out the channel count in sof_ipc4_dbg_audio_format mainline inclusion from mainline-v6.10-rc2 Print out the number of channels for the format explicitly instead of having the reader to understand how to interpret the ch_map and ch_cfg values. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Link: https://msgid.link/r/20240530111918.21974-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 2a865c9c3fb0289a95f1cb51b42d248736ff45cb) --- sound/soc/sof/ipc4-topology.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index e564e0442186..273008c1eb57 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -175,9 +175,10 @@ static void sof_ipc4_dbg_audio_format(struct device *dev, struct sof_ipc4_pin_fo for (i = 0; i < num_formats; i++) { struct sof_ipc4_audio_format *fmt = &pin_fmt[i].audio_fmt; dev_dbg(dev, - "Pin index #%d: %uHz, %ubit (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x) buffer size %d\n", - pin_fmt[i].pin_index, fmt->sampling_frequency, fmt->bit_depth, fmt->ch_map, - fmt->ch_cfg, fmt->interleaving_style, fmt->fmt_cfg, + "Pin index #%d: %uHz, %ubit, %luch (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x) buffer size %d\n", + pin_fmt[i].pin_index, fmt->sampling_frequency, fmt->bit_depth, + SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg), + fmt->ch_map, fmt->ch_cfg, fmt->interleaving_style, fmt->fmt_cfg, pin_fmt[i].buffer_size); } } From b1c23a29597e392d123e4aadbdfd6a170f737d58 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 May 2024 14:19:16 +0300 Subject: [PATCH 38/40] ASoC: SOF: ipc4-topology/pcm: Rename sof_ipc4_copier_is_single_format() mainline inclusion from mainline-v6.10-rc1 Rename the sof_ipc4_copier_is_single_format() to sof_ipc4_copier_is_single_bitdepth() to clear the confusion of the use of 'format' when we are querying information on the bit depth. Format is used to describe a combination of parameters (rate, channels, sample format / bit depth). Rename the flags used to store the result at the same time. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240530111918.21974-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 3b64fd2f83f203f5a34faed3dadf6464313f827d) --- sound/soc/sof/ipc4-pcm.c | 12 +++++------ sound/soc/sof/ipc4-topology.c | 40 +++++++++++++++++------------------ sound/soc/sof/ipc4-topology.h | 6 +++--- 3 files changed, 29 insertions(+), 29 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 4594470ed08b..e0c2103a74fb 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -650,7 +650,7 @@ static int sof_ipc4_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sof_ipc4_audio_format *ipc4_fmt; struct sof_ipc4_copier *ipc4_copier; - bool single_fmt = false; + bool single_bitdepth = false; u32 valid_bits = 0; int dir, ret; @@ -682,18 +682,18 @@ static int sof_ipc4_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, return 0; if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - if (sof_ipc4_copier_is_single_format(sdev, + if (sof_ipc4_copier_is_single_bitdepth(sdev, available_fmt->output_pin_fmts, available_fmt->num_output_formats)) { ipc4_fmt = &available_fmt->output_pin_fmts->audio_fmt; - single_fmt = true; + single_bitdepth = true; } } else { - if (sof_ipc4_copier_is_single_format(sdev, + if (sof_ipc4_copier_is_single_bitdepth(sdev, available_fmt->input_pin_fmts, available_fmt->num_input_formats)) { ipc4_fmt = &available_fmt->input_pin_fmts->audio_fmt; - single_fmt = true; + single_bitdepth = true; } } } @@ -703,7 +703,7 @@ static int sof_ipc4_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, if (ret) return ret; - if (single_fmt) { + if (single_bitdepth) { snd_mask_none(fmt); valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(ipc4_fmt->fmt_cfg); dev_dbg(component->dev, "Set %s to %d bit format\n", dai->name, valid_bits); diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 273008c1eb57..c2f01a5b76fb 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1367,7 +1367,7 @@ static int snd_sof_get_hw_config_params(struct snd_sof_dev *sdev, struct snd_sof static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, - bool single_format, + bool single_bitdepth, struct snd_pcm_hw_params *params, u32 dai_index, u32 linktype, u8 dir, u32 **dst, u32 *len) { @@ -1390,7 +1390,7 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai * Look for 32-bit blob first instead of 16-bit if copier * supports multiple formats */ - if (bit_depth == 16 && !single_format) { + if (bit_depth == 16 && !single_bitdepth) { dev_dbg(sdev->dev, "Looking for 32-bit blob first for DMIC\n"); format_change = true; bit_depth = 32; @@ -1438,7 +1438,7 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai bit_depth = params_width(params); format_change = false; get_new_blob = true; - } else if (linktype == SOF_DAI_INTEL_DMIC && !single_format) { + } else if (linktype == SOF_DAI_INTEL_DMIC && !single_bitdepth) { /* * The requested 32-bit blob (no format change for the * blob request) was not found in NHLT table, try to @@ -1494,7 +1494,7 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai #else static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, - bool single_format, + bool single_bitdepth, struct snd_pcm_hw_params *params, u32 dai_index, u32 linktype, u8 dir, u32 **dst, u32 *len) { @@ -1502,9 +1502,9 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai } #endif -bool sof_ipc4_copier_is_single_format(struct snd_sof_dev *sdev, - struct sof_ipc4_pin_format *pin_fmts, - u32 pin_fmts_size) +bool sof_ipc4_copier_is_single_bitdepth(struct snd_sof_dev *sdev, + struct sof_ipc4_pin_format *pin_fmts, + u32 pin_fmts_size) { struct sof_ipc4_audio_format *fmt; u32 valid_bits; @@ -1535,7 +1535,7 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, struct snd_pcm_hw_params dai_params = *params; struct sof_ipc4_copier_data *copier_data; struct sof_ipc4_copier *ipc4_copier; - bool single_format; + bool single_bitdepth; int ret; ipc4_copier = dai->private; @@ -1549,12 +1549,12 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, * format lookup */ if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - single_format = sof_ipc4_copier_is_single_format(sdev, + single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, available_fmt->output_pin_fmts, available_fmt->num_output_formats); /* Update the dai_params with the only supported format */ - if (single_format) { + if (single_bitdepth) { ret = sof_ipc4_update_hw_params(sdev, &dai_params, &available_fmt->output_pin_fmts[0].audio_fmt, BIT(SNDRV_PCM_HW_PARAM_FORMAT)); @@ -1562,12 +1562,12 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, return ret; } } else { - single_format = sof_ipc4_copier_is_single_format(sdev, + single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, available_fmt->input_pin_fmts, available_fmt->num_input_formats); /* Update the dai_params with the only supported format */ - if (single_format) { + if (single_bitdepth) { ret = sof_ipc4_update_hw_params(sdev, &dai_params, &available_fmt->input_pin_fmts[0].audio_fmt, BIT(SNDRV_PCM_HW_PARAM_FORMAT)); @@ -1576,7 +1576,7 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, } } - ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, single_format, + ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, single_bitdepth, &dai_params, ipc4_copier->dai_index, ipc4_copier->dai_type, dir, @@ -1611,7 +1611,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, u32 out_ref_rate, out_ref_channels; u32 deep_buffer_dma_ms = 0; int output_fmt_index; - bool single_output_format; + bool single_output_bitdepth; dev_dbg(sdev->dev, "copier %s, type %d", swidget->widget->name, swidget->id); @@ -1747,9 +1747,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, return ret; /* set the reference params for output format selection */ - single_output_format = sof_ipc4_copier_is_single_format(sdev, - available_fmt->output_pin_fmts, - available_fmt->num_output_formats); + single_output_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, + available_fmt->output_pin_fmts, + available_fmt->num_output_formats); switch (swidget->id) { case snd_soc_dapm_aif_in: case snd_soc_dapm_dai_out: @@ -1761,7 +1761,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, out_ref_rate = in_fmt->sampling_frequency; out_ref_channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(in_fmt->fmt_cfg); - if (!single_output_format) + if (!single_output_bitdepth) out_ref_valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(in_fmt->fmt_cfg); break; @@ -1770,7 +1770,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, case snd_soc_dapm_dai_in: out_ref_rate = params_rate(fe_params); out_ref_channels = params_channels(fe_params); - if (!single_output_format) { + if (!single_output_bitdepth) { out_ref_valid_bits = sof_ipc4_get_valid_bits(sdev, fe_params); if (out_ref_valid_bits < 0) return out_ref_valid_bits; @@ -1788,7 +1788,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, * if the output format is the same across all available output formats, choose * that as the reference. */ - if (single_output_format) { + if (single_output_bitdepth) { struct sof_ipc4_audio_format *out_fmt; out_fmt = &available_fmt->output_pin_fmts[0].audio_fmt; diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index dce174a190dd..78a8ef74c37a 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -475,7 +475,7 @@ struct sof_ipc4_process { u32 init_config; }; -bool sof_ipc4_copier_is_single_format(struct snd_sof_dev *sdev, - struct sof_ipc4_pin_format *pin_fmts, - u32 pin_fmts_size); +bool sof_ipc4_copier_is_single_bitdepth(struct snd_sof_dev *sdev, + struct sof_ipc4_pin_format *pin_fmts, + u32 pin_fmts_size); #endif From ff29ed7e2181ee07ae7189dd594b9188176e0d3c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 May 2024 14:19:17 +0300 Subject: [PATCH 39/40] ASoC: SOF: ipc4-topology: Improve readability of sof_ipc4_prepare_dai_copier() mainline inclusion from mainline-v6.10-rc1 Remove the duplicated code paths to check for single bit depth and to update the params with storing the parameters needed by the function and have a single code section. No functional change but the code is easier to follow. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240530111918.21974-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit 2fcad03eaba1b86e6b829f73a9e75e681b7f3106) --- sound/soc/sof/ipc4-topology.c | 42 +++++++++++++++-------------------- 1 file changed, 18 insertions(+), 24 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index c2f01a5b76fb..8b5c8ee2b275 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1534,8 +1534,10 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, struct sof_ipc4_available_audio_format *available_fmt; struct snd_pcm_hw_params dai_params = *params; struct sof_ipc4_copier_data *copier_data; + struct sof_ipc4_pin_format *pin_fmts; struct sof_ipc4_copier *ipc4_copier; bool single_bitdepth; + u32 num_pin_fmts; int ret; ipc4_copier = dai->private; @@ -1549,31 +1551,23 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, * format lookup */ if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, - available_fmt->output_pin_fmts, - available_fmt->num_output_formats); - - /* Update the dai_params with the only supported format */ - if (single_bitdepth) { - ret = sof_ipc4_update_hw_params(sdev, &dai_params, - &available_fmt->output_pin_fmts[0].audio_fmt, - BIT(SNDRV_PCM_HW_PARAM_FORMAT)); - if (ret) - return ret; - } + pin_fmts = available_fmt->output_pin_fmts; + num_pin_fmts = available_fmt->num_output_formats; } else { - single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, - available_fmt->input_pin_fmts, - available_fmt->num_input_formats); - - /* Update the dai_params with the only supported format */ - if (single_bitdepth) { - ret = sof_ipc4_update_hw_params(sdev, &dai_params, - &available_fmt->input_pin_fmts[0].audio_fmt, - BIT(SNDRV_PCM_HW_PARAM_FORMAT)); - if (ret) - return ret; - } + pin_fmts = available_fmt->input_pin_fmts; + num_pin_fmts = available_fmt->num_input_formats; + } + + single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, pin_fmts, + num_pin_fmts); + + /* Update the dai_params with the only supported format */ + if (single_bitdepth) { + ret = sof_ipc4_update_hw_params(sdev, &dai_params, + &pin_fmts[0].audio_fmt, + BIT(SNDRV_PCM_HW_PARAM_FORMAT)); + if (ret) + return ret; } ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, single_bitdepth, From b79d5a880eb6f18fb6fdcc56ec84f64da4dff419 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 May 2024 14:19:18 +0300 Subject: [PATCH 40/40] ASoC: SOF: ipc4-topology: Adjust the params based on DAI formats mainline inclusion from mainline-v6.10-rc1 Currently we only check the bit depth value among to DAI formats, but other parameters might be constant, like number of channels and/or rate. In capture we use the fe params as a reference to find the format and blob which should be used, but in the path we can have components which can handle expanding/narrowing number of channels or do a resample. In these cases the topology is expected to have 'fixed' parameter for channels/rates/bit depth and the conversion to the fe format is going to be done within the path. In practice this patch fixes issues like: All DMIC formats are fixed four channels We have a component which converts the four channel to stereo FE is opened with 2 channel Even if we have the correct bit depth format and blob (for four channel) we will still be looking for stereo configurations, which will fail. Note: the adjustment of params have switched order with the checking of single bit depth (needed for the NHLT blob fallback support). This change is non function, just that if the sof_ipc4_narrow_params_to_format() would fail, there is no point of checking the single bit depth. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240530111918.21974-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown (cherry picked from commit b65456b7b379e20ab225a4e906dc4a0c98fddd7a) --- sound/soc/sof/ipc4-topology.c | 71 ++++++++++++++++++++++++++++------- 1 file changed, 57 insertions(+), 14 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 8b5c8ee2b275..27fad6821d68 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1527,6 +1527,55 @@ bool sof_ipc4_copier_is_single_bitdepth(struct snd_sof_dev *sdev, return true; } +static int +sof_ipc4_adjust_params_to_dai_format(struct snd_sof_dev *sdev, + struct snd_pcm_hw_params *params, + struct sof_ipc4_pin_format *pin_fmts, + u32 pin_fmts_size) +{ + u32 params_mask = BIT(SNDRV_PCM_HW_PARAM_RATE) | + BIT(SNDRV_PCM_HW_PARAM_CHANNELS) | + BIT(SNDRV_PCM_HW_PARAM_FORMAT); + struct sof_ipc4_audio_format *fmt; + u32 rate, channels, valid_bits; + int i; + + fmt = &pin_fmts[0].audio_fmt; + rate = fmt->sampling_frequency; + channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg); + valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg); + + /* check if parameters in topology defined formats are the same */ + for (i = 1; i < pin_fmts_size; i++) { + u32 val; + + fmt = &pin_fmts[i].audio_fmt; + + if (params_mask & BIT(SNDRV_PCM_HW_PARAM_RATE)) { + val = fmt->sampling_frequency; + if (val != rate) + params_mask &= ~BIT(SNDRV_PCM_HW_PARAM_RATE); + } + if (params_mask & BIT(SNDRV_PCM_HW_PARAM_CHANNELS)) { + val = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg); + if (val != channels) + params_mask &= ~BIT(SNDRV_PCM_HW_PARAM_CHANNELS); + } + if (params_mask & BIT(SNDRV_PCM_HW_PARAM_FORMAT)) { + val = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg); + if (val != valid_bits) + params_mask &= ~BIT(SNDRV_PCM_HW_PARAM_FORMAT); + } + } + + if (params_mask) + return sof_ipc4_update_hw_params(sdev, params, + &pin_fmts[0].audio_fmt, + params_mask); + + return 0; +} + static int sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, struct snd_pcm_hw_params *params, int dir) @@ -1545,10 +1594,9 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, available_fmt = &ipc4_copier->available_fmt; /* - * If the copier on the DAI side supports only single bit depth then - * this depth (format) should be used to look for the NHLT blob (if - * needed) and in case of capture this should be used for the input - * format lookup + * Fixup the params based on the format parameters of the DAI. If any + * of the RATE, CHANNELS, bit depth is static among the formats then + * narrow the params to only allow that specific parameter value. */ if (dir == SNDRV_PCM_STREAM_PLAYBACK) { pin_fmts = available_fmt->output_pin_fmts; @@ -1558,18 +1606,13 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, num_pin_fmts = available_fmt->num_input_formats; } + ret = sof_ipc4_adjust_params_to_dai_format(sdev, &dai_params, pin_fmts, + num_pin_fmts); + if (ret) + return ret; + single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, pin_fmts, num_pin_fmts); - - /* Update the dai_params with the only supported format */ - if (single_bitdepth) { - ret = sof_ipc4_update_hw_params(sdev, &dai_params, - &pin_fmts[0].audio_fmt, - BIT(SNDRV_PCM_HW_PARAM_FORMAT)); - if (ret) - return ret; - } - ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, single_bitdepth, &dai_params, ipc4_copier->dai_index,