Re: [alsa-devel] [PATCH 6/8] add PCM interface

o-takashi@sakamocchi.jp wrote:

• • Additionally, according to AudioFire Owner's Manual Version 2.2,
• • the number of PCM channels for digital input has more restriction
• • depending on which digital interface is selected.
• • • S/PDIF coaxial and optical : use input 1-2
• • • ADAT optical with 32.0-48.0 kHz : use input 1-8
• • • ADAT optical with 88.2-96.0 kHz : use input 1-4 (S/MUX format)
• • If these restriction is applied, the number of channels in stream is decided
• • according to above modes.
• • Currently this module doesn't have rules for the latter.

Does the number of channels in the AMDTP stream change, or are those channel still there, but filled with zeros?

+pcm_init_hw_params(struct snd_efw *efw,

• 	struct snd_pcm_substream *substream)
  

• .info = SNDRV_PCM_INFO_MMAP |
  

• 	SNDRV_PCM_INFO_BATCH |
  

• 	SNDRV_PCM_INFO_INTERLEAVED |
  

• 	SNDRV_PCM_INFO_SYNC_START |
  

• 	SNDRV_PCM_INFO_FIFO_IN_FRAMES |
  

• 	/* for Open Sound System compatibility */
  

• 	SNDRV_PCM_INFO_MMAP_VALID |
  

• 	SNDRV_PCM_INFO_BLOCK_TRANSFER,
  

Set SNDRV_PCM_INFO_JOINT_DUPLEX to specify that the playback/capture streams are somehow related (must use the same rate).

• /* format of PCM samples is 16bit or 24bit inner 32bit */
• err = snd_pcm_hw_constraint_step(substream->runtime, 0,

• 		SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
  

• err = snd_pcm_hw_constraint_step(substream->runtime, 0,

• 		SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
  

This has nothing to do with the sample format.

The AMDTP code requires(?) buffers and periods to be aligned with packets, so you need constraints for 8/16/32 *frames*.

• /*

• * The same sampling rate must be used for transmit and receive stream
  

• * as long as the streams include PCM samples
  

• */
  

• if ((amdtp_stream_running(&efw->receive_stream) &&

•     amdtp_stream_pcm_running(&efw->receive_stream)) ||
  

•    (amdtp_stream_running(&efw->transmit_stream) &&
  

•     amdtp_stream_pcm_running(&efw->transmit_stream))) {
  

• err = snd_efw_command_get_sampling_rate(efw, &sampling_rate);
  

• if (err < 0)
  

• 	goto end;
  

• substream->runtime->hw.rate_min = sampling_rate;
  

• substream->runtime->hw.rate_max = sampling_rate;
  

• }

What is the purpose of the "Sampling Rate" mixer control?

+pcm_pointer(struct snd_pcm_substream *substream) +{

• struct snd_efw *efw = substream->private_data;
• if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)

• return amdtp_stream_pcm_pointer(&efw->receive_stream);
  

• else

• return amdtp_stream_pcm_pointer(&efw->transmit_stream);
  

+}

If the playback/capture callbacks have nothing in common, just use two functions.

Regards, Clemens