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eed5391f67
The SOF driver removed the support for INFO_RESUME in the commit "ASoC: SOF: pcm: do not add SNDRV_PCM_INFO_RESUME to runtime hw info". And resuming is handled by the ALSA core with the .prepare and .trigger_start stages. So, remove handling of RESUME trigger in the component driver trigger op. Reviewed-by: Kai Vehmanen <kai.vehmanen@intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20211216231628.344687-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
940 lines
27 KiB
C
940 lines
27 KiB
C
// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
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//
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// This file is provided under a dual BSD/GPLv2 license. When using or
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// redistributing this file, you may do so under either license.
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//
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// Copyright(c) 2018 Intel Corporation. All rights reserved.
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//
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// Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
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//
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// PCM Layer, interface between ALSA and IPC.
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//
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#include <linux/pm_runtime.h>
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#include <sound/pcm_params.h>
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#include <sound/sof.h>
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#include "sof-priv.h"
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#include "sof-audio.h"
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#include "ops.h"
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#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
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#include "sof-probes.h"
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#endif
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/* Create DMA buffer page table for DSP */
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static int create_page_table(struct snd_soc_component *component,
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struct snd_pcm_substream *substream,
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unsigned char *dma_area, size_t size)
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{
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struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
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struct snd_sof_pcm *spcm;
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struct snd_dma_buffer *dmab = snd_pcm_get_dma_buf(substream);
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int stream = substream->stream;
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spcm = snd_sof_find_spcm_dai(component, rtd);
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if (!spcm)
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return -EINVAL;
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return snd_sof_create_page_table(component->dev, dmab,
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spcm->stream[stream].page_table.area, size);
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}
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static int sof_pcm_dsp_params(struct snd_sof_pcm *spcm, struct snd_pcm_substream *substream,
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const struct sof_ipc_pcm_params_reply *reply)
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{
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struct snd_soc_component *scomp = spcm->scomp;
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struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp);
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/* validate offset */
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int ret = snd_sof_ipc_pcm_params(sdev, substream, reply);
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if (ret < 0)
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dev_err(scomp->dev, "error: got wrong reply for PCM %d\n",
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spcm->pcm.pcm_id);
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return ret;
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}
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/*
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* sof pcm period elapse work
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*/
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static void snd_sof_pcm_period_elapsed_work(struct work_struct *work)
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{
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struct snd_sof_pcm_stream *sps =
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container_of(work, struct snd_sof_pcm_stream,
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period_elapsed_work);
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snd_pcm_period_elapsed(sps->substream);
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}
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void snd_sof_pcm_init_elapsed_work(struct work_struct *work)
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{
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INIT_WORK(work, snd_sof_pcm_period_elapsed_work);
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}
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/*
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* sof pcm period elapse, this could be called at irq thread context.
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*/
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void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream)
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{
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struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
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struct snd_soc_component *component =
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snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME);
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struct snd_sof_pcm *spcm;
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spcm = snd_sof_find_spcm_dai(component, rtd);
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if (!spcm) {
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dev_err(component->dev,
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"error: period elapsed for unknown stream!\n");
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return;
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}
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/*
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* snd_pcm_period_elapsed() can be called in interrupt context
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* before IRQ_HANDLED is returned. Inside snd_pcm_period_elapsed(),
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* when the PCM is done draining or xrun happened, a STOP IPC will
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* then be sent and this IPC will hit IPC timeout.
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* To avoid sending IPC before the previous IPC is handled, we
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* schedule delayed work here to call the snd_pcm_period_elapsed().
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*/
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schedule_work(&spcm->stream[substream->stream].period_elapsed_work);
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}
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EXPORT_SYMBOL(snd_sof_pcm_period_elapsed);
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int sof_pcm_dsp_pcm_free(struct snd_pcm_substream *substream, struct snd_sof_dev *sdev,
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struct snd_sof_pcm *spcm)
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{
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struct sof_ipc_stream stream;
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struct sof_ipc_reply reply;
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int ret;
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if (!spcm->prepared[substream->stream])
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return 0;
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stream.hdr.size = sizeof(stream);
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stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE;
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stream.comp_id = spcm->stream[substream->stream].comp_id;
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/* send IPC to the DSP */
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ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
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sizeof(stream), &reply, sizeof(reply));
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if (!ret)
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spcm->prepared[substream->stream] = false;
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return ret;
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}
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static int sof_pcm_setup_connected_widgets(struct snd_sof_dev *sdev,
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struct snd_soc_pcm_runtime *rtd,
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struct snd_sof_pcm *spcm, int dir)
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{
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struct snd_soc_dai *dai;
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int ret, j;
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/* query DAPM for list of connected widgets and set them up */
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for_each_rtd_cpu_dais(rtd, j, dai) {
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struct snd_soc_dapm_widget_list *list;
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ret = snd_soc_dapm_dai_get_connected_widgets(dai, dir, &list,
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dpcm_end_walk_at_be);
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if (ret < 0) {
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dev_err(sdev->dev, "error: dai %s has no valid %s path\n", dai->name,
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dir == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture");
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return ret;
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}
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spcm->stream[dir].list = list;
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ret = sof_widget_list_setup(sdev, spcm, dir);
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if (ret < 0) {
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dev_err(sdev->dev, "error: failed widget list set up for pcm %d dir %d\n",
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spcm->pcm.pcm_id, dir);
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spcm->stream[dir].list = NULL;
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snd_soc_dapm_dai_free_widgets(&list);
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return ret;
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}
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}
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return 0;
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}
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static int sof_pcm_hw_params(struct snd_soc_component *component,
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struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
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struct snd_sof_pcm *spcm;
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struct sof_ipc_pcm_params pcm;
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struct sof_ipc_pcm_params_reply ipc_params_reply;
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int ret;
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/* nothing to do for BE */
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if (rtd->dai_link->no_pcm)
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return 0;
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spcm = snd_sof_find_spcm_dai(component, rtd);
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if (!spcm)
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return -EINVAL;
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/*
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* Handle repeated calls to hw_params() without free_pcm() in
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* between. At least ALSA OSS emulation depends on this.
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*/
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ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
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if (ret < 0)
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return ret;
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dev_dbg(component->dev, "pcm: hw params stream %d dir %d\n",
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spcm->pcm.pcm_id, substream->stream);
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memset(&pcm, 0, sizeof(pcm));
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/* create compressed page table for audio firmware */
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if (runtime->buffer_changed) {
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ret = create_page_table(component, substream, runtime->dma_area,
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runtime->dma_bytes);
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if (ret < 0)
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return ret;
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}
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/* number of pages should be rounded up */
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pcm.params.buffer.pages = PFN_UP(runtime->dma_bytes);
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/* set IPC PCM parameters */
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pcm.hdr.size = sizeof(pcm);
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pcm.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_PARAMS;
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pcm.comp_id = spcm->stream[substream->stream].comp_id;
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pcm.params.hdr.size = sizeof(pcm.params);
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pcm.params.buffer.phy_addr =
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spcm->stream[substream->stream].page_table.addr;
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pcm.params.buffer.size = runtime->dma_bytes;
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pcm.params.direction = substream->stream;
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pcm.params.sample_valid_bytes = params_width(params) >> 3;
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pcm.params.buffer_fmt = SOF_IPC_BUFFER_INTERLEAVED;
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pcm.params.rate = params_rate(params);
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pcm.params.channels = params_channels(params);
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pcm.params.host_period_bytes = params_period_bytes(params);
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/* container size */
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ret = snd_pcm_format_physical_width(params_format(params));
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if (ret < 0)
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return ret;
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pcm.params.sample_container_bytes = ret >> 3;
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/* format */
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switch (params_format(params)) {
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case SNDRV_PCM_FORMAT_S16:
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pcm.params.frame_fmt = SOF_IPC_FRAME_S16_LE;
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break;
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case SNDRV_PCM_FORMAT_S24:
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pcm.params.frame_fmt = SOF_IPC_FRAME_S24_4LE;
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break;
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case SNDRV_PCM_FORMAT_S32:
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pcm.params.frame_fmt = SOF_IPC_FRAME_S32_LE;
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break;
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case SNDRV_PCM_FORMAT_FLOAT:
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pcm.params.frame_fmt = SOF_IPC_FRAME_FLOAT;
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break;
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default:
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return -EINVAL;
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}
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/* firmware already configured host stream */
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ret = snd_sof_pcm_platform_hw_params(sdev,
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substream,
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params,
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&pcm.params);
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if (ret < 0) {
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dev_err(component->dev, "error: platform hw params failed\n");
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return ret;
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}
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dev_dbg(component->dev, "stream_tag %d", pcm.params.stream_tag);
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/* if this is a repeated hw_params without hw_free, skip setting up widgets */
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if (!spcm->stream[substream->stream].list) {
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ret = sof_pcm_setup_connected_widgets(sdev, rtd, spcm, substream->stream);
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if (ret < 0)
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return ret;
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}
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/* send hw_params IPC to the DSP */
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ret = sof_ipc_tx_message(sdev->ipc, pcm.hdr.cmd, &pcm, sizeof(pcm),
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&ipc_params_reply, sizeof(ipc_params_reply));
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if (ret < 0) {
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dev_err(component->dev, "error: hw params ipc failed for stream %d\n",
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pcm.params.stream_tag);
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return ret;
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}
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ret = sof_pcm_dsp_params(spcm, substream, &ipc_params_reply);
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if (ret < 0)
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return ret;
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spcm->prepared[substream->stream] = true;
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/* save pcm hw_params */
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memcpy(&spcm->params[substream->stream], params, sizeof(*params));
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return ret;
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}
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static int sof_pcm_hw_free(struct snd_soc_component *component,
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struct snd_pcm_substream *substream)
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{
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struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
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struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
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struct snd_sof_pcm *spcm;
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int ret, err = 0;
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/* nothing to do for BE */
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if (rtd->dai_link->no_pcm)
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return 0;
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spcm = snd_sof_find_spcm_dai(component, rtd);
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if (!spcm)
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return -EINVAL;
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dev_dbg(component->dev, "pcm: free stream %d dir %d\n",
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spcm->pcm.pcm_id, substream->stream);
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/* free PCM in the DSP */
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ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
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if (ret < 0)
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err = ret;
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/* stop DMA */
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ret = snd_sof_pcm_platform_hw_free(sdev, substream);
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if (ret < 0) {
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dev_err(component->dev, "error: platform hw free failed\n");
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err = ret;
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}
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/* free the DAPM widget list */
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ret = sof_widget_list_free(sdev, spcm, substream->stream);
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if (ret < 0)
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err = ret;
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cancel_work_sync(&spcm->stream[substream->stream].period_elapsed_work);
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return err;
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}
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static int sof_pcm_prepare(struct snd_soc_component *component,
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struct snd_pcm_substream *substream)
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{
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struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
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struct snd_sof_pcm *spcm;
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int ret;
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/* nothing to do for BE */
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if (rtd->dai_link->no_pcm)
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return 0;
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spcm = snd_sof_find_spcm_dai(component, rtd);
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if (!spcm)
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return -EINVAL;
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if (spcm->prepared[substream->stream])
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return 0;
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dev_dbg(component->dev, "pcm: prepare stream %d dir %d\n",
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spcm->pcm.pcm_id, substream->stream);
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/* set hw_params */
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ret = sof_pcm_hw_params(component,
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substream, &spcm->params[substream->stream]);
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if (ret < 0) {
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dev_err(component->dev,
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"error: set pcm hw_params after resume\n");
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return ret;
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}
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return 0;
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}
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/*
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* FE dai link trigger actions are always executed in non-atomic context because
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* they involve IPC's.
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*/
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static int sof_pcm_trigger(struct snd_soc_component *component,
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struct snd_pcm_substream *substream, int cmd)
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{
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struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
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struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
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struct snd_sof_pcm *spcm;
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struct sof_ipc_stream stream;
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struct sof_ipc_reply reply;
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bool reset_hw_params = false;
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bool free_widget_list = false;
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bool ipc_first = false;
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int ret;
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/* nothing to do for BE */
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if (rtd->dai_link->no_pcm)
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return 0;
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spcm = snd_sof_find_spcm_dai(component, rtd);
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if (!spcm)
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return -EINVAL;
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dev_dbg(component->dev, "pcm: trigger stream %d dir %d cmd %d\n",
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spcm->pcm.pcm_id, substream->stream, cmd);
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stream.hdr.size = sizeof(stream);
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stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG;
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stream.comp_id = spcm->stream[substream->stream].comp_id;
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switch (cmd) {
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case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
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stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_PAUSE;
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ipc_first = true;
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break;
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case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
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stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_RELEASE;
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break;
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case SNDRV_PCM_TRIGGER_START:
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if (spcm->stream[substream->stream].suspend_ignored) {
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/*
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* This case will be triggered when INFO_RESUME is
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* not supported, no need to re-start streams that
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* remained enabled in D0ix.
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*/
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spcm->stream[substream->stream].suspend_ignored = false;
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return 0;
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}
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stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_START;
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break;
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case SNDRV_PCM_TRIGGER_SUSPEND:
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if (sdev->system_suspend_target == SOF_SUSPEND_S0IX &&
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spcm->stream[substream->stream].d0i3_compatible) {
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/*
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* trap the event, not sending trigger stop to
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* prevent the FW pipelines from being stopped,
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* and mark the flag to ignore the upcoming DAPM
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* PM events.
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*/
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spcm->stream[substream->stream].suspend_ignored = true;
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return 0;
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}
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free_widget_list = true;
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fallthrough;
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case SNDRV_PCM_TRIGGER_STOP:
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stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP;
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ipc_first = true;
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reset_hw_params = true;
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break;
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default:
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dev_err(component->dev, "error: unhandled trigger cmd %d\n",
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cmd);
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return -EINVAL;
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}
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/*
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* DMA and IPC sequence is different for start and stop. Need to send
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* STOP IPC before stop DMA
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*/
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if (!ipc_first)
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snd_sof_pcm_platform_trigger(sdev, substream, cmd);
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/* send IPC to the DSP */
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ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
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sizeof(stream), &reply, sizeof(reply));
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/* need to STOP DMA even if STOP IPC failed */
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if (ipc_first)
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snd_sof_pcm_platform_trigger(sdev, substream, cmd);
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/* free PCM if reset_hw_params is set and the STOP IPC is successful */
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if (!ret && reset_hw_params) {
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ret = sof_pcm_stream_free(sdev, substream, spcm, substream->stream,
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free_widget_list);
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if (ret < 0)
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return ret;
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}
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return ret;
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}
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static snd_pcm_uframes_t sof_pcm_pointer(struct snd_soc_component *component,
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struct snd_pcm_substream *substream)
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{
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struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
|
|
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
|
|
struct snd_sof_pcm *spcm;
|
|
snd_pcm_uframes_t host, dai;
|
|
|
|
/* nothing to do for BE */
|
|
if (rtd->dai_link->no_pcm)
|
|
return 0;
|
|
|
|
/* use dsp ops pointer callback directly if set */
|
|
if (sof_ops(sdev)->pcm_pointer)
|
|
return sof_ops(sdev)->pcm_pointer(sdev, substream);
|
|
|
|
spcm = snd_sof_find_spcm_dai(component, rtd);
|
|
if (!spcm)
|
|
return -EINVAL;
|
|
|
|
/* read position from DSP */
|
|
host = bytes_to_frames(substream->runtime,
|
|
spcm->stream[substream->stream].posn.host_posn);
|
|
dai = bytes_to_frames(substream->runtime,
|
|
spcm->stream[substream->stream].posn.dai_posn);
|
|
|
|
dev_vdbg(component->dev,
|
|
"PCM: stream %d dir %d DMA position %lu DAI position %lu\n",
|
|
spcm->pcm.pcm_id, substream->stream, host, dai);
|
|
|
|
return host;
|
|
}
|
|
|
|
static int sof_pcm_open(struct snd_soc_component *component,
|
|
struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
|
|
const struct snd_sof_dsp_ops *ops = sof_ops(sdev);
|
|
struct snd_sof_pcm *spcm;
|
|
struct snd_soc_tplg_stream_caps *caps;
|
|
int ret;
|
|
|
|
/* nothing to do for BE */
|
|
if (rtd->dai_link->no_pcm)
|
|
return 0;
|
|
|
|
spcm = snd_sof_find_spcm_dai(component, rtd);
|
|
if (!spcm)
|
|
return -EINVAL;
|
|
|
|
dev_dbg(component->dev, "pcm: open stream %d dir %d\n",
|
|
spcm->pcm.pcm_id, substream->stream);
|
|
|
|
|
|
caps = &spcm->pcm.caps[substream->stream];
|
|
|
|
/* set runtime config */
|
|
runtime->hw.info = ops->hw_info; /* platform-specific */
|
|
|
|
/* set any runtime constraints based on topology */
|
|
runtime->hw.formats = le64_to_cpu(caps->formats);
|
|
runtime->hw.period_bytes_min = le32_to_cpu(caps->period_size_min);
|
|
runtime->hw.period_bytes_max = le32_to_cpu(caps->period_size_max);
|
|
runtime->hw.periods_min = le32_to_cpu(caps->periods_min);
|
|
runtime->hw.periods_max = le32_to_cpu(caps->periods_max);
|
|
|
|
/*
|
|
* caps->buffer_size_min is not used since the
|
|
* snd_pcm_hardware structure only defines buffer_bytes_max
|
|
*/
|
|
runtime->hw.buffer_bytes_max = le32_to_cpu(caps->buffer_size_max);
|
|
|
|
dev_dbg(component->dev, "period min %zd max %zd bytes\n",
|
|
runtime->hw.period_bytes_min,
|
|
runtime->hw.period_bytes_max);
|
|
dev_dbg(component->dev, "period count %d max %d\n",
|
|
runtime->hw.periods_min,
|
|
runtime->hw.periods_max);
|
|
dev_dbg(component->dev, "buffer max %zd bytes\n",
|
|
runtime->hw.buffer_bytes_max);
|
|
|
|
/* set wait time - TODO: come from topology */
|
|
substream->wait_time = 500;
|
|
|
|
spcm->stream[substream->stream].posn.host_posn = 0;
|
|
spcm->stream[substream->stream].posn.dai_posn = 0;
|
|
spcm->stream[substream->stream].substream = substream;
|
|
spcm->prepared[substream->stream] = false;
|
|
|
|
ret = snd_sof_pcm_platform_open(sdev, substream);
|
|
if (ret < 0)
|
|
dev_err(component->dev, "error: pcm open failed %d\n", ret);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int sof_pcm_close(struct snd_soc_component *component,
|
|
struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
|
|
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
|
|
struct snd_sof_pcm *spcm;
|
|
int err;
|
|
|
|
/* nothing to do for BE */
|
|
if (rtd->dai_link->no_pcm)
|
|
return 0;
|
|
|
|
spcm = snd_sof_find_spcm_dai(component, rtd);
|
|
if (!spcm)
|
|
return -EINVAL;
|
|
|
|
dev_dbg(component->dev, "pcm: close stream %d dir %d\n",
|
|
spcm->pcm.pcm_id, substream->stream);
|
|
|
|
err = snd_sof_pcm_platform_close(sdev, substream);
|
|
if (err < 0) {
|
|
dev_err(component->dev, "error: pcm close failed %d\n",
|
|
err);
|
|
/*
|
|
* keep going, no point in preventing the close
|
|
* from happening
|
|
*/
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* Pre-allocate playback/capture audio buffer pages.
|
|
* no need to explicitly release memory preallocated by sof_pcm_new in pcm_free
|
|
* snd_pcm_lib_preallocate_free_for_all() is called by the core.
|
|
*/
|
|
static int sof_pcm_new(struct snd_soc_component *component,
|
|
struct snd_soc_pcm_runtime *rtd)
|
|
{
|
|
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
|
|
struct snd_sof_pcm *spcm;
|
|
struct snd_pcm *pcm = rtd->pcm;
|
|
struct snd_soc_tplg_stream_caps *caps;
|
|
int stream = SNDRV_PCM_STREAM_PLAYBACK;
|
|
|
|
/* find SOF PCM for this RTD */
|
|
spcm = snd_sof_find_spcm_dai(component, rtd);
|
|
if (!spcm) {
|
|
dev_warn(component->dev, "warn: can't find PCM with DAI ID %d\n",
|
|
rtd->dai_link->id);
|
|
return 0;
|
|
}
|
|
|
|
dev_dbg(component->dev, "creating new PCM %s\n", spcm->pcm.pcm_name);
|
|
|
|
/* do we need to pre-allocate playback audio buffer pages */
|
|
if (!spcm->pcm.playback)
|
|
goto capture;
|
|
|
|
caps = &spcm->pcm.caps[stream];
|
|
|
|
/* pre-allocate playback audio buffer pages */
|
|
dev_dbg(component->dev,
|
|
"spcm: allocate %s playback DMA buffer size 0x%x max 0x%x\n",
|
|
caps->name, caps->buffer_size_min, caps->buffer_size_max);
|
|
|
|
if (!pcm->streams[stream].substream) {
|
|
dev_err(component->dev, "error: NULL playback substream!\n");
|
|
return -EINVAL;
|
|
}
|
|
|
|
snd_pcm_set_managed_buffer(pcm->streams[stream].substream,
|
|
SNDRV_DMA_TYPE_DEV_SG, sdev->dev,
|
|
0, le32_to_cpu(caps->buffer_size_max));
|
|
capture:
|
|
stream = SNDRV_PCM_STREAM_CAPTURE;
|
|
|
|
/* do we need to pre-allocate capture audio buffer pages */
|
|
if (!spcm->pcm.capture)
|
|
return 0;
|
|
|
|
caps = &spcm->pcm.caps[stream];
|
|
|
|
/* pre-allocate capture audio buffer pages */
|
|
dev_dbg(component->dev,
|
|
"spcm: allocate %s capture DMA buffer size 0x%x max 0x%x\n",
|
|
caps->name, caps->buffer_size_min, caps->buffer_size_max);
|
|
|
|
if (!pcm->streams[stream].substream) {
|
|
dev_err(component->dev, "error: NULL capture substream!\n");
|
|
return -EINVAL;
|
|
}
|
|
|
|
snd_pcm_set_managed_buffer(pcm->streams[stream].substream,
|
|
SNDRV_DMA_TYPE_DEV_SG, sdev->dev,
|
|
0, le32_to_cpu(caps->buffer_size_max));
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void ssp_dai_config_pcm_params_match(struct snd_sof_dev *sdev, const char *link_name,
|
|
struct snd_pcm_hw_params *params)
|
|
{
|
|
struct sof_ipc_dai_config *config;
|
|
struct snd_sof_dai *dai;
|
|
int i;
|
|
|
|
/*
|
|
* Search for all matching DAIs as we can have both playback and capture DAI
|
|
* associated with the same link.
|
|
*/
|
|
list_for_each_entry(dai, &sdev->dai_list, list) {
|
|
if (!dai->name || strcmp(link_name, dai->name))
|
|
continue;
|
|
for (i = 0; i < dai->number_configs; i++) {
|
|
config = &dai->dai_config[i];
|
|
if (config->ssp.fsync_rate == params_rate(params)) {
|
|
dev_dbg(sdev->dev, "DAI config %d matches pcm hw params\n", i);
|
|
dai->current_config = i;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/* fixup the BE DAI link to match any values from topology */
|
|
int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params)
|
|
{
|
|
struct snd_interval *rate = hw_param_interval(params,
|
|
SNDRV_PCM_HW_PARAM_RATE);
|
|
struct snd_interval *channels = hw_param_interval(params,
|
|
SNDRV_PCM_HW_PARAM_CHANNELS);
|
|
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
|
|
struct snd_soc_component *component =
|
|
snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME);
|
|
struct snd_sof_dai *dai =
|
|
snd_sof_find_dai(component, (char *)rtd->dai_link->name);
|
|
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
|
|
struct snd_soc_dpcm *dpcm;
|
|
|
|
/* no topology exists for this BE, try a common configuration */
|
|
if (!dai) {
|
|
dev_warn(component->dev,
|
|
"warning: no topology found for BE DAI %s config\n",
|
|
rtd->dai_link->name);
|
|
|
|
/* set 48k, stereo, 16bits by default */
|
|
rate->min = 48000;
|
|
rate->max = 48000;
|
|
|
|
channels->min = 2;
|
|
channels->max = 2;
|
|
|
|
snd_mask_none(fmt);
|
|
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* read format from topology */
|
|
snd_mask_none(fmt);
|
|
|
|
switch (dai->comp_dai.config.frame_fmt) {
|
|
case SOF_IPC_FRAME_S16_LE:
|
|
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
|
|
break;
|
|
case SOF_IPC_FRAME_S24_4LE:
|
|
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
|
|
break;
|
|
case SOF_IPC_FRAME_S32_LE:
|
|
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S32_LE);
|
|
break;
|
|
default:
|
|
dev_err(component->dev, "error: No available DAI format!\n");
|
|
return -EINVAL;
|
|
}
|
|
|
|
/* read rate and channels from topology */
|
|
switch (dai->dai_config->type) {
|
|
case SOF_DAI_INTEL_SSP:
|
|
/* search for config to pcm params match, if not found use default */
|
|
ssp_dai_config_pcm_params_match(sdev, (char *)rtd->dai_link->name, params);
|
|
|
|
rate->min = dai->dai_config[dai->current_config].ssp.fsync_rate;
|
|
rate->max = dai->dai_config[dai->current_config].ssp.fsync_rate;
|
|
channels->min = dai->dai_config[dai->current_config].ssp.tdm_slots;
|
|
channels->max = dai->dai_config[dai->current_config].ssp.tdm_slots;
|
|
|
|
dev_dbg(component->dev,
|
|
"rate_min: %d rate_max: %d\n", rate->min, rate->max);
|
|
dev_dbg(component->dev,
|
|
"channels_min: %d channels_max: %d\n",
|
|
channels->min, channels->max);
|
|
|
|
break;
|
|
case SOF_DAI_INTEL_DMIC:
|
|
/* DMIC only supports 16 or 32 bit formats */
|
|
if (dai->comp_dai.config.frame_fmt == SOF_IPC_FRAME_S24_4LE) {
|
|
dev_err(component->dev,
|
|
"error: invalid fmt %d for DAI type %d\n",
|
|
dai->comp_dai.config.frame_fmt,
|
|
dai->dai_config->type);
|
|
}
|
|
break;
|
|
case SOF_DAI_INTEL_HDA:
|
|
/*
|
|
* HDAudio does not follow the default trigger
|
|
* sequence due to firmware implementation
|
|
*/
|
|
for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
|
|
struct snd_soc_pcm_runtime *fe = dpcm->fe;
|
|
|
|
fe->dai_link->trigger[SNDRV_PCM_STREAM_PLAYBACK] =
|
|
SND_SOC_DPCM_TRIGGER_POST;
|
|
}
|
|
break;
|
|
case SOF_DAI_INTEL_ALH:
|
|
/*
|
|
* Dai could run with different channel count compared with
|
|
* front end, so get dai channel count from topology
|
|
*/
|
|
channels->min = dai->dai_config->alh.channels;
|
|
channels->max = dai->dai_config->alh.channels;
|
|
break;
|
|
case SOF_DAI_IMX_ESAI:
|
|
rate->min = dai->dai_config->esai.fsync_rate;
|
|
rate->max = dai->dai_config->esai.fsync_rate;
|
|
channels->min = dai->dai_config->esai.tdm_slots;
|
|
channels->max = dai->dai_config->esai.tdm_slots;
|
|
|
|
dev_dbg(component->dev,
|
|
"rate_min: %d rate_max: %d\n", rate->min, rate->max);
|
|
dev_dbg(component->dev,
|
|
"channels_min: %d channels_max: %d\n",
|
|
channels->min, channels->max);
|
|
break;
|
|
case SOF_DAI_MEDIATEK_AFE:
|
|
rate->min = dai->dai_config->afe.rate;
|
|
rate->max = dai->dai_config->afe.rate;
|
|
channels->min = dai->dai_config->afe.channels;
|
|
channels->max = dai->dai_config->afe.channels;
|
|
|
|
dev_dbg(component->dev,
|
|
"rate_min: %d rate_max: %d\n", rate->min, rate->max);
|
|
dev_dbg(component->dev,
|
|
"channels_min: %d channels_max: %d\n",
|
|
channels->min, channels->max);
|
|
break;
|
|
case SOF_DAI_IMX_SAI:
|
|
rate->min = dai->dai_config->sai.fsync_rate;
|
|
rate->max = dai->dai_config->sai.fsync_rate;
|
|
channels->min = dai->dai_config->sai.tdm_slots;
|
|
channels->max = dai->dai_config->sai.tdm_slots;
|
|
|
|
dev_dbg(component->dev,
|
|
"rate_min: %d rate_max: %d\n", rate->min, rate->max);
|
|
dev_dbg(component->dev,
|
|
"channels_min: %d channels_max: %d\n",
|
|
channels->min, channels->max);
|
|
break;
|
|
case SOF_DAI_AMD_BT:
|
|
rate->min = dai->dai_config->acpbt.fsync_rate;
|
|
rate->max = dai->dai_config->acpbt.fsync_rate;
|
|
channels->min = dai->dai_config->acpbt.tdm_slots;
|
|
channels->max = dai->dai_config->acpbt.tdm_slots;
|
|
|
|
dev_dbg(component->dev,
|
|
"AMD_BT rate_min: %d rate_max: %d\n", rate->min, rate->max);
|
|
dev_dbg(component->dev,
|
|
"AMD_BT channels_min: %d channels_max: %d\n",
|
|
channels->min, channels->max);
|
|
break;
|
|
case SOF_DAI_AMD_SP:
|
|
rate->min = dai->dai_config->acpsp.fsync_rate;
|
|
rate->max = dai->dai_config->acpsp.fsync_rate;
|
|
channels->min = dai->dai_config->acpsp.tdm_slots;
|
|
channels->max = dai->dai_config->acpsp.tdm_slots;
|
|
|
|
dev_dbg(component->dev,
|
|
"AMD_SP rate_min: %d rate_max: %d\n", rate->min, rate->max);
|
|
dev_dbg(component->dev,
|
|
"AMD_SP channels_min: %d channels_max: %d\n",
|
|
channels->min, channels->max);
|
|
break;
|
|
case SOF_DAI_AMD_DMIC:
|
|
rate->min = dai->dai_config->acpdmic.fsync_rate;
|
|
rate->max = dai->dai_config->acpdmic.fsync_rate;
|
|
channels->min = dai->dai_config->acpdmic.tdm_slots;
|
|
channels->max = dai->dai_config->acpdmic.tdm_slots;
|
|
|
|
dev_dbg(component->dev,
|
|
"AMD_DMIC rate_min: %d rate_max: %d\n", rate->min, rate->max);
|
|
dev_dbg(component->dev,
|
|
"AMD_DMIC channels_min: %d channels_max: %d\n",
|
|
channels->min, channels->max);
|
|
break;
|
|
default:
|
|
dev_err(component->dev, "error: invalid DAI type %d\n",
|
|
dai->dai_config->type);
|
|
break;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL(sof_pcm_dai_link_fixup);
|
|
|
|
static int sof_pcm_probe(struct snd_soc_component *component)
|
|
{
|
|
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
|
|
struct snd_sof_pdata *plat_data = sdev->pdata;
|
|
const char *tplg_filename;
|
|
int ret;
|
|
|
|
/* load the default topology */
|
|
sdev->component = component;
|
|
|
|
tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL,
|
|
"%s/%s",
|
|
plat_data->tplg_filename_prefix,
|
|
plat_data->tplg_filename);
|
|
if (!tplg_filename)
|
|
return -ENOMEM;
|
|
|
|
ret = snd_sof_load_topology(component, tplg_filename);
|
|
if (ret < 0) {
|
|
dev_err(component->dev, "error: failed to load DSP topology %d\n",
|
|
ret);
|
|
return ret;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void sof_pcm_remove(struct snd_soc_component *component)
|
|
{
|
|
/* remove topology */
|
|
snd_soc_tplg_component_remove(component);
|
|
}
|
|
|
|
static int sof_pcm_ack(struct snd_soc_component *component,
|
|
struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
|
|
|
|
return snd_sof_pcm_platform_ack(sdev, substream);
|
|
}
|
|
|
|
void snd_sof_new_platform_drv(struct snd_sof_dev *sdev)
|
|
{
|
|
struct snd_soc_component_driver *pd = &sdev->plat_drv;
|
|
struct snd_sof_pdata *plat_data = sdev->pdata;
|
|
const char *drv_name;
|
|
|
|
drv_name = plat_data->machine->drv_name;
|
|
|
|
pd->name = "sof-audio-component";
|
|
pd->probe = sof_pcm_probe;
|
|
pd->remove = sof_pcm_remove;
|
|
pd->open = sof_pcm_open;
|
|
pd->close = sof_pcm_close;
|
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pd->hw_params = sof_pcm_hw_params;
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|
pd->prepare = sof_pcm_prepare;
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|
pd->hw_free = sof_pcm_hw_free;
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|
pd->trigger = sof_pcm_trigger;
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|
pd->pointer = sof_pcm_pointer;
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|
pd->ack = sof_pcm_ack;
|
|
|
|
#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
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pd->compress_ops = &sof_probe_compressed_ops;
|
|
#endif
|
|
pd->pcm_construct = sof_pcm_new;
|
|
pd->ignore_machine = drv_name;
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|
pd->be_hw_params_fixup = sof_pcm_dai_link_fixup;
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|
pd->be_pcm_base = SOF_BE_PCM_BASE;
|
|
pd->use_dai_pcm_id = true;
|
|
pd->topology_name_prefix = "sof";
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|
|
|
/* increment module refcount when a pcm is opened */
|
|
pd->module_get_upon_open = 1;
|
|
}
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