Audio Introduction Generic commands and instructions

Most of the boards have simple audio setup which means we have one sound card with one playback and one capture PCM. To list the available sound cards and PCMs for playback:

aplay -l

To list the available sound cards and PCMs for capture:

arecord -l

In most cases -Dplughw:0,0 is the device we want to use for audio but in case we have several audio devices (onboard + USB for example) one need to specify which device to use for audio:

To play audio on card0’s PCM0 and let ALSA to decide if resampling is needed:

aplay -Dplughw:0,0 <path to wav file>

To record audio to a file:

arecord -Dplughw:0,0 -t wav <path to wav file>

To test full duplex audio (play back the recorded audio w/o intermediate file):

arecord -Dplughw:0,0 | aplay -Dplughw:0,0

To request specific audio format to be used for playback/capture take a look at the help of aplay/arecord. For example, one can specify the format with -f, the sampling rate with -r, or the number of channels with -c. In this case, one should open the hw device (not the plughw) via -Dhw:0,0. For example, record 48KHz, stereo 16bit audio:

arecord -Dhw:0,0 -fdat -t wav record_48K_stereo_16bit.wav

Or to record record 96KHz, stereo 24bit audio:

arecord -Dhw:0,0 -fS24_LE -c2 -r96000 -t wav record_96K_stereo_24bit.wav

It is a good practice to save the mixer settings found to be good and reload them after every boot (if your distribution is not doing this already)

Set the mixers for the board with amixer, alsamixer
alsactl -f board.aconf store

After booting up the board it can be restored with a single command:

alsactl -f board.aconf restore Board-specific instructions Potential issues

In case of XRUN (under or overrun)

The underrun can happen when an application does not feed new samples in time to alsa-lib (due CPU usage). The overrun can happen when an application does not take new captured samples in time from alsa-lib.
There could be several reasons for XRUN to happen, but it usually points to system latency issues connected to CPU utilization or latency caused by the storage device.
Things to try:
  • Increase the buffer size (ALSA buffer and period size)
  • Try to cache the file to be played in memory
  • Try to use application which uses threads for interacting with ALSA and with the filesystem

ALSA period size must be aligned with the FIFO depth (tx/rx numevt)

No longer relevant as the kernel side takes care of the AFIFO depth vs period size issue.
To decrease audio-caused stress on the system, the AFIFO is enabled and the depth is set to 32 for McASP.
If the ALSA period size is not aligned with this FIFO setting, a constant ‘trrrrr’ can be heard on the output. This is caused by the eDMA not being able to handle a fragment size that is not aligned with burst size (AFIFO depth).
Application needs to make sure that period_size / FIFO depth is even number. Additional Information

  1. ALSA SoC Project Homepage
  2. ALSA Project Homepage
  3. ALSA User Space Library
  4. Using ALSA Audio API Author: Paul Davis

Software Help

Audio hardware codecs