diff options
author | Takashi Iwai <tiwai@suse.de> | 2024-04-05 09:48:12 +0300 |
---|---|---|
committer | Takashi Iwai <tiwai@suse.de> | 2024-04-05 09:48:12 +0300 |
commit | 100c85421b52e41269ada88f7d71a6b8a06c7a11 (patch) | |
tree | 1e291a2d1b8e68720a15dfd48964357502d03fc9 /sound/soc/sof | |
parent | c4e51e424e2c772ce1836912a8b0b87cd61bc9d5 (diff) | |
parent | 90f8917e7a15f6dd508779048bdf00ce119b6ca0 (diff) | |
download | linux-100c85421b52e41269ada88f7d71a6b8a06c7a11.tar.xz |
Merge tag 'asoc-fix-v6.9-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.9
A relatively large set of fixes here, the biggest piece of it is a
series correcting some problems with the delay reporting for Intel SOF
cards but there's a bunch of other things. Everything here is driver
specific except for a fix in the core for an issue with sign extension
handling volume controls.
Diffstat (limited to 'sound/soc/sof')
-rw-r--r-- | sound/soc/sof/amd/acp.c | 8 | ||||
-rw-r--r-- | sound/soc/sof/core.c | 14 | ||||
-rw-r--r-- | sound/soc/sof/intel/hda-common-ops.c | 3 | ||||
-rw-r--r-- | sound/soc/sof/intel/hda-dai-ops.c | 11 | ||||
-rw-r--r-- | sound/soc/sof/intel/hda-dsp.c | 20 | ||||
-rw-r--r-- | sound/soc/sof/intel/hda-pcm.c | 29 | ||||
-rw-r--r-- | sound/soc/sof/intel/hda-stream.c | 70 | ||||
-rw-r--r-- | sound/soc/sof/intel/hda.h | 6 | ||||
-rw-r--r-- | sound/soc/sof/intel/lnl.c | 34 | ||||
-rw-r--r-- | sound/soc/sof/intel/mtl.c | 14 | ||||
-rw-r--r-- | sound/soc/sof/intel/mtl.h | 10 | ||||
-rw-r--r-- | sound/soc/sof/ipc4-mtrace.c | 11 | ||||
-rw-r--r-- | sound/soc/sof/ipc4-pcm.c | 191 | ||||
-rw-r--r-- | sound/soc/sof/ipc4-priv.h | 14 | ||||
-rw-r--r-- | sound/soc/sof/ipc4-topology.c | 22 | ||||
-rw-r--r-- | sound/soc/sof/ops.h | 24 | ||||
-rw-r--r-- | sound/soc/sof/pcm.c | 8 | ||||
-rw-r--r-- | sound/soc/sof/sof-audio.h | 9 | ||||
-rw-r--r-- | sound/soc/sof/sof-priv.h | 24 |
19 files changed, 406 insertions, 116 deletions
diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index be7dc1e02284..c12c7f820529 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -704,6 +704,10 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) goto unregister_dev; } + ret = acp_init(sdev); + if (ret < 0) + goto free_smn_dev; + sdev->ipc_irq = pci->irq; ret = request_threaded_irq(sdev->ipc_irq, acp_irq_handler, acp_irq_thread, IRQF_SHARED, "AudioDSP", sdev); @@ -713,10 +717,6 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) goto free_smn_dev; } - ret = acp_init(sdev); - if (ret < 0) - goto free_ipc_irq; - /* scan SoundWire capabilities exposed by DSDT */ ret = acp_sof_scan_sdw_devices(sdev, chip->sdw_acpi_dev_addr); if (ret < 0) { diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 9b00ede2a486..cc84d4c81be9 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -339,8 +339,7 @@ static int sof_init_environment(struct snd_sof_dev *sdev) ret = snd_sof_probe(sdev); if (ret < 0) { dev_err(sdev->dev, "failed to probe DSP %d\n", ret); - sof_ops_free(sdev); - return ret; + goto err_sof_probe; } /* check machine info */ @@ -358,15 +357,18 @@ static int sof_init_environment(struct snd_sof_dev *sdev) ret = validate_sof_ops(sdev); if (ret < 0) { snd_sof_remove(sdev); + snd_sof_remove_late(sdev); return ret; } } + return 0; + err_machine_check: - if (ret) { - snd_sof_remove(sdev); - sof_ops_free(sdev); - } + snd_sof_remove(sdev); +err_sof_probe: + snd_sof_remove_late(sdev); + sof_ops_free(sdev); return ret; } diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 2b385cddc385..d71bb66b9991 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -57,6 +57,9 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { .pcm_pointer = hda_dsp_pcm_pointer, .pcm_ack = hda_dsp_pcm_ack, + .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-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index c50ca9e72d37..b073720b4cf4 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -7,6 +7,7 @@ #include <sound/pcm_params.h> #include <sound/hdaudio_ext.h> +#include <sound/hda_register.h> #include <sound/hda-mlink.h> #include <sound/sof/ipc4/header.h> #include <uapi/sound/sof/header.h> @@ -362,6 +363,16 @@ 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-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 31ffa1a8f2ac..ef5c915db8ff 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -681,17 +681,27 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; struct hdac_bus *bus = sof_to_bus(sdev); + bool imr_lost = false; int ret, j; /* - * The memory used for IMR boot loses its content in deeper than S3 state - * We must not try IMR boot on next power up (as it will fail). - * + * The memory used for IMR boot loses its content in deeper than S3 + * state on CAVS platforms. + * On ACE platforms due to the system architecture the IMR content is + * lost at S3 state already, they are tailored for s2idle use. + * We must not try IMR boot on next power up in these cases as it will + * fail. + */ + if (sdev->system_suspend_target > SOF_SUSPEND_S3 || + (chip->hw_ip_version >= SOF_INTEL_ACE_1_0 && + sdev->system_suspend_target == SOF_SUSPEND_S3)) + imr_lost = true; + + /* * In case of firmware crash or boot failure set the skip_imr_boot to true * as well in order to try to re-load the firmware to do a 'cold' boot. */ - if (sdev->system_suspend_target > SOF_SUSPEND_S3 || - sdev->fw_state == SOF_FW_CRASHED || + if (imr_lost || sdev->fw_state == SOF_FW_CRASHED || sdev->fw_state == SOF_FW_BOOT_FAILED) hda->skip_imr_boot = true; diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 18f07364d219..d7b446f3f973 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -259,8 +259,37 @@ 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; + + /* + * 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 b387b1a69d7e..0c189d3b19c1 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1063,3 +1063,73 @@ 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 + * @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); + + /* 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); +} + +/** + * 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 b36eb7c78913..81a1d4606d3c 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -662,6 +662,12 @@ 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); +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); diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 7ae017a00184..aeb4350cce6b 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -29,15 +29,17 @@ static const struct snd_sof_debugfs_map lnl_dsp_debugfs[] = { }; /* this helps allows the DSP to setup DMIC/SSP */ -static int hdac_bus_offload_dmic_ssp(struct hdac_bus *bus) +static int hdac_bus_offload_dmic_ssp(struct hdac_bus *bus, bool enable) { int ret; - ret = hdac_bus_eml_enable_offload(bus, true, AZX_REG_ML_LEPTR_ID_INTEL_SSP, true); + ret = hdac_bus_eml_enable_offload(bus, true, + AZX_REG_ML_LEPTR_ID_INTEL_SSP, enable); if (ret < 0) return ret; - ret = hdac_bus_eml_enable_offload(bus, true, AZX_REG_ML_LEPTR_ID_INTEL_DMIC, true); + ret = hdac_bus_eml_enable_offload(bus, true, + AZX_REG_ML_LEPTR_ID_INTEL_DMIC, enable); if (ret < 0) return ret; @@ -52,7 +54,19 @@ static int lnl_hda_dsp_probe(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); +} + +static void lnl_hda_dsp_remove(struct snd_sof_dev *sdev) +{ + int ret; + + ret = hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), false); + if (ret < 0) + dev_warn(sdev->dev, + "Failed to disable offload for DMIC/SSP: %d\n", ret); + + hda_dsp_remove(sdev); } static int lnl_hda_dsp_resume(struct snd_sof_dev *sdev) @@ -63,7 +77,7 @@ static int lnl_hda_dsp_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); } static int lnl_hda_dsp_runtime_resume(struct snd_sof_dev *sdev) @@ -74,7 +88,7 @@ static int lnl_hda_dsp_runtime_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); } static int lnl_dsp_post_fw_run(struct snd_sof_dev *sdev) @@ -97,9 +111,11 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) /* common defaults */ memcpy(&sof_lnl_ops, &sof_hda_common_ops, sizeof(struct snd_sof_dsp_ops)); - /* probe */ - if (!sdev->dspless_mode_selected) + /* probe/remove */ + if (!sdev->dspless_mode_selected) { sof_lnl_ops.probe = lnl_hda_dsp_probe; + sof_lnl_ops.remove = lnl_hda_dsp_remove; + } /* shutdown */ sof_lnl_ops.shutdown = hda_dsp_shutdown; @@ -134,8 +150,6 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) 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 df05dc77b8d5..060c34988e90 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -626,18 +626,6 @@ 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; @@ -707,8 +695,6 @@ 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 cc5a1f46fd09..ea8c1b83f712 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); diff --git a/sound/soc/sof/ipc4-mtrace.c b/sound/soc/sof/ipc4-mtrace.c index 9f1e33ee8826..0e04bea9432d 100644 --- a/sound/soc/sof/ipc4-mtrace.c +++ b/sound/soc/sof/ipc4-mtrace.c @@ -4,6 +4,7 @@ #include <linux/debugfs.h> #include <linux/sched/signal.h> +#include <linux/sched/clock.h> #include <sound/sof/ipc4/header.h> #include "sof-priv.h" #include "ipc4-priv.h" @@ -412,7 +413,6 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev) const struct sof_ipc_ops *iops = sdev->ipc->ops; struct sof_ipc4_msg msg; u64 system_time; - ktime_t kt; int ret; if (priv->mtrace_state != SOF_MTRACE_DISABLED) @@ -424,9 +424,12 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev) msg.primary |= SOF_IPC4_MOD_INSTANCE(SOF_IPC4_MOD_INIT_BASEFW_INSTANCE_ID); msg.extension = SOF_IPC4_MOD_EXT_MSG_PARAM_ID(SOF_IPC4_FW_PARAM_SYSTEM_TIME); - /* The system time is in usec, UTC, epoch is 1601-01-01 00:00:00 */ - kt = ktime_add_us(ktime_get_real(), FW_EPOCH_DELTA * USEC_PER_SEC); - system_time = ktime_to_us(kt); + /* + * local_clock() is used to align with dmesg, so both kernel and firmware logs have + * the same base and a minor delta due to the IPC. system time is in us format but + * local_clock() returns the time in ns, so convert to ns. + */ + system_time = div64_u64(local_clock(), NSEC_PER_USEC); msg.data_size = sizeof(system_time); msg.data_ptr = &system_time; ret = iops->set_get_data(sdev, &msg, msg.data_size, true); diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 0f332c8cdbe6..e915f9f87a6c 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -15,6 +15,28 @@ #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 (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, struct ipc4_pipeline_set_state_data *trigger_list) { @@ -423,8 +445,19 @@ 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); @@ -464,14 +497,12 @@ 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; @@ -703,6 +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; @@ -835,7 +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; @@ -852,38 +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 @@ -893,45 +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)); + /* - * 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) counter via host accessible means if + * available. */ if (!time_info->llp_offset) { - tmp_ptr = snd_sof_pcm_get_stream_position(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 position to get accurate one + * 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 */ - tmp_ptr -= time_info->stream_start_offset; - /* Calculate the delay taking into account that both pointer can wrap */ - div64_u64_rem(tmp_ptr, substream->runtime->boundary, &tmp_ptr); + /* + * 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. + * + * The delay is the difference between the counters on the two + * sides of the DSP. + */ + 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); + 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 = { @@ -941,6 +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, diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index f3b908b093f9..afed618a15f0 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -92,20 +92,6 @@ 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; diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index f28edd9830c1..5cca05842126 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -412,8 +412,9 @@ 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) @@ -447,6 +448,25 @@ 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) { diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 6cf21e829e07..3cd748e13460 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -523,12 +523,26 @@ 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) +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_stream_position) - return sof_ops(sdev)->get_stream_position(sdev, component, 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; } diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 33d576b17647..f03cee94bce6 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -388,13 +388,21 @@ 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 9ea2ac5adac7..86bbb531e142 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 @@ -124,6 +127,9 @@ 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; @@ -322,6 +328,7 @@ 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 diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index d453a4ce3b21..d3c436f82604 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -262,13 +262,25 @@ struct snd_sof_dsp_ops { 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. + * 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_stream_position)(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream); /* optional */ + 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, |