On Thu 14 Dec 09:33 PST 2017, srinivas.kandagatla@linaro.org wrote:
[..]
+enum stream_state {
- IDLE = 0,
- STOPPED,
- RUNNING,
These are too generic.
+};
+struct q6asm_dai_rtd {
- struct snd_pcm_substream *substream;
- dma_addr_t phys;
- unsigned int pcm_size;
- unsigned int pcm_count;
- unsigned int pcm_irq_pos; /* IRQ position */
- unsigned int periods;
- uint16_t bits_per_sample;
- uint16_t source; /* Encoding source bit mask */
- struct audio_client *audio_client;
- uint16_t session_id;
- enum stream_state state;
- bool set_channel_map;
- char channel_map[8];
There's a define for this 8
+};
+struct q6asm_dai_data {
- u64 sid;
+};
+static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
- .info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_BLOCK_TRANSFER |SNDRV_PCM_INFO_MMAP_VALID |SNDRV_PCM_INFO_INTERLEAVED |SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),- .formats = (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE),- .rates = SNDRV_PCM_RATE_8000_192000,
- .rate_min = 8000,
- .rate_max = 192000,
- .channels_min = 1,
- .channels_max = 8,
- .buffer_bytes_max = (PLAYBACK_MAX_NUM_PERIODS *
PLAYBACK_MAX_PERIOD_SIZE),- .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE,
- .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE,
- .periods_min = PLAYBACK_MIN_NUM_PERIODS,
- .periods_max = PLAYBACK_MAX_NUM_PERIODS,
If you just put the numbers here, instead of using the PLAYBACK_ defines, it's possible to grok the values of this struct without having to jump to the defines for each one.
- .fifo_size = 0,
+};
+/* Conventional and unconventional sample rate supported */ +static unsigned int supported_sample_rates[] = {
- 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
- 88200, 96000, 176400, 192000
+};
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
This is unreferenced.
- .count = ARRAY_SIZE(supported_sample_rates),
- .list = supported_sample_rates,
- .mask = 0,
+};
+static void event_handler(uint32_t opcode, uint32_t token,
uint32_t *payload, void *priv)+{
- struct q6asm_dai_rtd *prtd = priv;
- struct snd_pcm_substream *substream = prtd->substream;
- switch (opcode) {
- case ASM_CLIENT_EVENT_CMD_RUN_DONE:
q6asm_write_nolock(prtd->audio_client,prtd->pcm_count, 0, 0, NO_TIMESTAMP);break;- case ASM_CLIENT_EVENT_CMD_EOS_DONE:
prtd->state = STOPPED;break;- case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
prtd->pcm_irq_pos += prtd->pcm_count;snd_pcm_period_elapsed(substream);if (prtd->state == RUNNING)q6asm_write_nolock(prtd->audio_client,prtd->pcm_count, 0, 0, NO_TIMESTAMP);break;}- default:
break;- }
+}
+static int q6asm_dai_prepare(struct snd_pcm_substream *substream) +{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
- struct q6asm_dai_rtd *prtd = runtime->private_data;
- struct q6asm_dai_data *pdata;
- int ret;
- pdata = dev_get_drvdata(soc_prtd->platform->dev);
- if (!pdata)
return -EINVAL;- if (!prtd || !prtd->audio_client) {
pr_err("%s: private data null or audio client freed\n",__func__);return -EINVAL;- }
- prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
- prtd->pcm_irq_pos = 0;
- /* rate and channels are sent to audio driver */
- if (prtd->state) {
/* clear the previous setup if any */q6asm_cmd(prtd->audio_client, CMD_CLOSE);q6asm_unmap_memory_regions(substream->stream,prtd->audio_client);q6routing_dereg_phy_stream(soc_prtd->dai_link->id,SNDRV_PCM_STREAM_PLAYBACK);- }
- ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client,
prtd->phys,(prtd->pcm_size / prtd->periods),prtd->periods);- if (ret < 0) {
pr_err("Audio Start: Buffer Allocation failed rc = %d\n",ret);return -ENOMEM;- }
- ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
prtd->bits_per_sample);- if (ret < 0) {
pr_err("%s: q6asm_open_write failed\n", __func__);q6asm_audio_client_free(prtd->audio_client);prtd->audio_client = NULL;
Do you need to roll back the q6asm_map_memory_regions?
return -ENOMEM;- }
- prtd->session_id = q6asm_get_session_id(prtd->audio_client);
- ret = q6routing_reg_phy_stream(soc_prtd->dai_link->id, LEGACY_PCM_MODE,
prtd->session_id, substream->stream);- if (ret) {
pr_err("%s: stream reg failed ret:%d\n", __func__, ret);return ret;- }
- ret = q6asm_media_format_block_multi_ch_pcm(
prtd->audio_client, runtime->rate,runtime->channels, !prtd->set_channel_map,prtd->channel_map, prtd->bits_per_sample);
set_channel_map and channel_map aren't referenced elsewhere. If this isn't used consider removing it for now.
- if (ret < 0)
pr_info("%s: CMD Format block failed\n", __func__);- prtd->state = RUNNING;
- return 0;
+}
[..]
+static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd) +{
- struct snd_pcm *pcm = rtd->pcm;
- struct snd_pcm_substream *substream;
- struct snd_card *card = rtd->card->snd_card;
- struct device *dev = card->dev;
- struct device_node *node = dev->of_node;
- struct q6asm_dai_data *pdata = dev_get_drvdata(rtd->platform->dev);
- struct of_phandle_args args;
- int size, ret = 0;
- ret = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
- if (ret < 0)
pdata->sid = -1;- else
pdata->sid = args.args[0];
Is this really how you're supposed to deal with the iommu?
- substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
- size = q6asm_dai_hardware_playback.buffer_bytes_max;
- ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
&substream->dma_buffer);- if (ret) {
dev_err(dev, "Cannot allocate buffer(s)\n");return ret;
Just fall through.
- }
- return ret;
+}
[..]
+static struct snd_soc_dai_driver q6asm_fe_dais[] = {
- {
.playback = {.stream_name = "MultiMedia1 Playback",.rates = (SNDRV_PCM_RATE_8000_192000|SNDRV_PCM_RATE_KNOT),.formats = (SNDRV_PCM_FMTBIT_S16_LE |SNDRV_PCM_FMTBIT_S24_LE),.channels_min = 1,.channels_max = 8,.rate_min = 8000,.rate_max = 192000,},.name = "MM_DL1",.probe = fe_dai_probe,.id = MSM_FRONTEND_DAI_MULTIMEDIA1,- },
- {
.playback = {.stream_name = "MultiMedia2 Playback",.rates = (SNDRV_PCM_RATE_8000_192000|SNDRV_PCM_RATE_KNOT),.formats = (SNDRV_PCM_FMTBIT_S16_LE |SNDRV_PCM_FMTBIT_S24_LE),.channels_min = 1,.channels_max = 8,.rate_min = 8000,.rate_max = 192000,
I presume the listed frontend DAIs needs to match the firmware of the DSP (and features of hardware)? Can we get away with a single list for all versions of the adsp?
In msm-4.4 the max rate for these where changed to 384000, see:
9c46f74b2724 ("ASoC: msm: add 384KHz playback support")
},.name = "MM_DL2",.probe = fe_dai_probe,.id = MSM_FRONTEND_DAI_MULTIMEDIA2,- },
+};
Regards, Bjorn