I am using DMA to transfer data to SPORT1 on my Blackfin BF537. I am using a CS42448 CODEC that has 8 DACs. I am using TDM mode and have set the following:
DMA Buffer Size For Each SPORT Data Line (Rx or Tx) = 131072 bytes DMA Periods = 8
static struct snd_pcm_hardware snd_pcm_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_RESUME), .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, .rate_max = 48000, .buffer_bytes_max = 32768, .channels_min = 2, .channels_max = 2, .period_bytes_min = 8, .period_bytes_max = 4096, .periods_min = 8, .periods_max = 8, };
Everytime I receive a DMA interrupt from the SPORT driver I call snd_pcm_period_elapsed for all of the substreams that are currently active. In the PCM Copy callback I calculate the position within the DMA buffer to write to based on the TDM slot for the channel and the position passed into the function.
The audio is not being placed into the DMA buffer properly. The audio plays faster than it should and has a constant stutter. I have read the "writing an ALSA driver" guide multiple times but it does not explain the relationship between struct snd_pcm_hardware, snd_pcm_period_elapsed, and the copy callback clearly enough for me to understand what I am doing wrong.
Please explain how these interact in more detail to help me fix this problem. I can also send the full source code to the CS42448 driver I am working on if more details are needed.
Thank you, Adam