3.2.4.2. Audio

3.2.4.2.1. Introduction

3.2.4.2.2. 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:

-Dplughw:omap5uevm,0 will use the onboard audio on OMAP5-uEVM board.

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

3.2.4.2.3. Board-specific instructions

AM43x-EPOS-EVM

The board uses tlv320aic3111 codec connected through McASP1 [AXR0 for playback, AXR1 for Capture] for audio. The board features internal stereo speakers and two 3.5mm jack for Headphone and Mic In

Kernel config

Device Drivers  --->
  Sound card support  --->
    Advanced Linux Sound Architecture  --->
      ALSA for SoC audio support  --->
        Audio support for Texas Instruments SoCs  --->
          <*> Multichannel Audio Serial Port (McASP) support
        CODEC drivers  --->
          <*> Texas Instruments TLV320AIC31xx CODECs
        <*>   ASoC Simple sound card support

User space

Note

Before audio playback ALSA mixers must be configured for either Headphone or Speaker output. The audio will not work with non correct mixer configuration!

To play audio through headphone jack run:

amixer sset 'DAC' 127
amixer sset 'HP Analog' 66
amixer sset 'HP Driver' 0 on
amixer sset 'HP Left' on
amixer sset 'HP Right' on
amixer sset 'Output Left From Left DAC' on
amixer sset 'Output Right From Right DAC' on

To play audio through internal speakers run:

amixer sset 'DAC' 127
amixer sset 'Speaker Analog' 127
amixer sset 'Speaker Driver' 0 on
amixer sset 'Speaker Left' on
amixer sset 'Speaker Right' on
amixer sset 'Output Left From Left DAC' on
amixer sset 'Output Right From Right DAC' on

To capture audio from both microphone channels run:

amixer sset 'MIC1RP P-Terminal' 'FFR 10 Ohm'
amixer sset 'MIC1LP P-Terminal' 'FFR 10 Ohm'
amixer sset 'ADC' 40
amixer cset name='ADC Capture Switch' on

If the captured audio has low volume you can try higer values for ‘Mic PGA’ mixer, for instance:

amixer sset 'Mic PGA' 50

Note: The codec on has only one channel ADC so the captured audio is dual channel mono signal.

AM437x-GP-EVM

The board uses tlv320aic3106 codec connected through McASP1 [AXR2 for playback, AXR3 for Capture] for audio. The board features two 3.5mm jack for Headphone and Line In.

Kernel config

Device Drivers  --->
  Sound card support  --->
    Advanced Linux Sound Architecture  --->
      ALSA for SoC audio support  --->
        Audio support for Texas Instruments SoCs  --->
          <*> Multichannel Audio Serial Port (McASP) support
        CODEC drivers  --->
          <*> Texas Instruments TLV320AIC3x CODECs
        <*>   ASoC Simple sound card support

User space

The hardware defaults are correct for audio playback, the routing is OK and the volume is ‘adequate’ but in case the volume is not correct:

amixer -c AM437xGPEVM sset PCM 90                            # Master Playback volume

Playback to Headphone only:

amixer -c AM437xGPEVM sset 'Left HP Mixer DACL1' on               # HP Left route enable
amixer -c AM437xGPEVM sset 'Right HP Mixer DACR1' on              # HP Right route enable
amixer -c AM437xGPEVM sset 'Left Line Mixer DACL1' off            # Line out Left disable
amixer -c AM437xGPEVM sset 'Right Line Mixer DACR1' off           # Line out Right disable
amixer -c AM437xGPEVM sset 'HP DAC' 90                            # Adjust HP volume

Record from Line In:

amixer -c AM437xGPEVM sset 'Left PGA Mixer Line1L' on             # Line in Left enable
amixer -c AM437xGPEVM sset 'Right PGA Mixer Line1R' on            # Line in Right enable
amixer -c AM437xGPEVM sset 'Left PGA Mixer Mic3L' off             # Analog mic Left disable
amixer -c AM437xGPEVM sset 'Right PGA Mixer Mic3R' off            # Analog mic Right disable
amixer -c AM437xGPEVM sset 'PGA' 40                               # Adjust Capture volume

3.2.4.2.4. 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.