On Thu 14 Dec 09:33 PST 2017, srinivas.kandagatla@xxxxxxxxxx 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
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