Audio Introduction

The audio subsystem present on various TI SoCs consists of two major components:

  1. Multi-channel Audio Serial Port (McASP) - Provides a full-duplex serial interface between the host processor and external audio peripherals like codecs over industry-standard protocols like Inter-IC sound (I2S).

  2. System DMA engine - Provides McASP with direct access to system memory to read audio samples from (for playback) or store audio samples to (for capture).

Along with the above, most TI EVMs and SKs have line input/output jack(s) wired to an on-board codec that can convert between the analog signals and the digital protocol supported by McASP. Software Architecture

All the hardware components are exposed to userspace applications using the Linux ALSA (Advance Linux Sound Architecture) framework, which allows control and configuration of the hardware through common APIs. For more details check the links below.

Within the kernel there are separate drivers for each component. For each board a sound-card instance is created, usually using the sound {} device tree node, that links together various components like different McASP instances to a codec or an HDMI bridge.

../../../../_images/audio-asoc-arch.png 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

SK-AM62x, SK-AM62Ax

The board uses tlv320aic3106 codec connected through McASP1 [AXR0 for playback, AXR2 for Capture] for audio. The board features one TRRS 3.5mm jack, that can be used for simultaneous stereo playback and mono recording. Same McASP1 lines are also muxed to the sii9022 HDMI bridge.

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 sset PCM 90%

For recording using the mic pin on the 3.5mm jack, you will need to unmute MIC3R on the codec, and increase the capture volume:

amixer sset 'Left PGA Mixer Mic3R' on
amixer sset 'Right PGA Mixer Mic3R' on
amixer sset PGA 90%

To switch to using HDMI for playback you can refer to the How to playback audio over HDMI guide. 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

In case of CPU stalls (when recording)

No longer relevant as DMA driver does a force teardown of the channel.
On some platforms, recording audio on high sample rates may work fine the first time, but due to issues with channel cleanup it may cause CPU stalls when recording the second time, requiring a reboot to fix.
In such scenarios, use smaller period sizes (64 to 256) while recording. For example:
arecord -Dplughw:0,0 -r 48000 -t wav --period-size=64 <path to wav file>

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.