3.3.4.2. Audio

3.3.4.2.1. Introduction

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

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).
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.

3.3.4.2.2. 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.

3.3.4.2.3. 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

3.3.4.2.4. Board-specific instructions

OMAP5 uEVM

The board uses twl6040 codec connected through McPDM for onboard audio and features one Headset connector, one Stereo Line In and one Stereo Line Out 3.5mm jack connectors.

Kernel config

Device Drivers  --->
  Sound card support  --->
    Advanced Linux Sound Architecture  --->
      ALSA for SoC audio support  --->
        Audio support for Texas Instruments SoCs  --->
          <*> SoC Audio support for OMAP boards using ABE and twl6040 codec
          <*> OMAP4/5 HDMI audio support

User space

To set up the audio routing on the board (Headset playback/capture):

amixer -c omap5uevm sset 'Headset Left Playback' 'HS DAC'  # HS Left channel from DAC
amixer -c omap5uevm sset 'Headset Right Playback' 'HS DAC' # HS Right channel from DAC
amixer -c omap5uevm sset Headset 4                         # HS volume to -22dB
amixer -c omap5uevm sset 'Analog Left' 'Headset Mic'       # Analog Left capture source from HS mic
amixer -c omap5uevm sset 'Analog Right' 'Headset Mic'      # Analog Right capture source from HS mic
amixer -c omap5uevm sset Capture 1                         # Analog Capture gain to 12dB

To play audio to the HS:

aplay -Dplughw:omap5uevm,0 <path to wav file (stereo)>

On kernels where the AESS (ABE) support is not available the Line Out can be used only when playing 4 channel audio. In this case the first two channel will be routed to HS and the second two will be the Line Out.

amixer -c omap5uevm sset 'Handsfree Left Playback' 'HF DAC'  # HF Left channel from DAC
amixer -c omap5uevm sset 'Handsfree Right Playback' 'HF DAC' # HF Right channel from DAC
amixer -c omap5uevm sset AUXL on                             # Enable route to AUXL from the HF path
amixer -c omap5uevm sset AUXR on                             # Enable route to AUXR from the HF path
amixer -c omap5uevm sset Handsfree 11                        # HS volume to -30dB

To play audio to the Line Out one should have 4 channel sample crafted and channel 3,4 should have the audio destined to Line Out:

aplay -Dplughw:omap5uevm,0 <path to wav file (4 channel)>

DRA7 and DRA72 EVM

The board uses tlv320aic3106 codec connected through McASP3 [AXR0 for playback, AXR1 for Capture] for audio. The board features four 3.5mm jack for Headphone, Line In, Line Out and one for Microphone.

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
          <*> OMAP4/5 HDMI audio 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 DRA7xxEVM sset PCM 90                            # Master Playback volume

Playback to Headphone only:

amixer -c DRA7xxEVM sset 'Left HP Mixer DACL1' on               # HP Left route enable
amixer -c DRA7xxEVM sset 'Right HP Mixer DACR1' on              # HP Right route enable
amixer -c DRA7xxEVM sset 'Left Line Mixer DACL1' off            # Line out Left disable
amixer -c DRA7xxEVM sset 'Right Line Mixer DACR1' off           # Line out Right disable
amixer -c DRA7xxEVM sset 'HP DAC' 90                            # Adjust HP volume

Playback to Line Out only:

amixer -c DRA7xxEVM sset 'Left HP Mixer DACL1' off              # HP Left route disable
amixer -c DRA7xxEVM sset 'Right HP Mixer DACR1' off             # HP Right route disable
amixer -c DRA7xxEVM sset 'Left Line Mixer DACL1' on             # Line out Left enable
amixer -c DRA7xxEVM sset 'Right Line Mixer DACR1' on            # Line out Right enable
amixer -c DRA7xxEVM sset 'Line DAC' 90                          # Adjust Line out volume

Record from Line In:

amixer -c DRA7xxEVM sset 'Left PGA Mixer Line1L' on             # Line in Left enable
amixer -c DRA7xxEVM sset 'Right PGA Mixer Line1R' on            # Line in Right enable
amixer -c DRA7xxEVM sset 'Left PGA Mixer Mic3L' off             # Analog mic Left disable
amixer -c DRA7xxEVM sset 'Right PGA Mixer Mic3R' off            # Analog mic Right disable
amixer -c DRA7xxEVM sset 'PGA' 40                               # Adjust Capture volume

Record from Analog Mic IN:

amixer -c DRA7xxEVM sset 'Left PGA Mixer Line1L' off            # Line in Left disable
amixer -c DRA7xxEVM sset 'Right PGA Mixer Line1R' off           # Line in Right disable
amixer -c DRA7xxEVM sset 'Left PGA Mixer Mic3L' on              # Analog mic Left enable
amixer -c DRA7xxEVM sset 'Right PGA Mixer Mic3R' on             # Analog mic Right enable
amixer -c DRA7xxEVM sset 'PGA' 40                               # Adjust Capture volume

BeagleBoard-X15 and AM572x-GP-EVM

The board uses tlv320aic3104 codec connected through McASP3 [AXR0 for playback, AXR1 for Capture] for audio. The board features two 3.5mm jack for Line Out 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
          <*> OMAP4/5 HDMI audio 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 BeagleBoardX15 sset PCM 90                            # Master Playback volume

Playback (line out):

amixer -c BeagleBoardX15 sset 'Left Line Mixer DACL1' on             # Line out Left enable
amixer -c BeagleBoardX15 sset 'Right Line Mixer DACR1' on            # Line out Right enable
amixer -c BeagleBoardX15 sset 'Line DAC' 90                          # Adjust Line out volume

Record (line in):

amixer -c BeagleBoardX15 sset 'Left PGA Mixer Mic2L' on         # Line in Left enable (MIC2/LINE2)
amixer -c BeagleBoardX15 sset 'Right PGA Mixer Mic2R' on        # Line in Right enable (MIC2/LINE2)
amixer -c BeagleBoardX15 sset 'PGA' 40                          # Adjust Capture volume

K2G EVM

The board uses tlv320aic3106 codec connected through McASP2
[AXR2 for playback, AXR3 for Capture] for audio. The board features
two 3.5mm jack for Headphone and Line In
NOTE 1: The Headphone jack is labeld as LINE OUT on the board
NOTE 2: Both analog and HDMI audio is served by McASP2, this means that they must not be used at the same time!
NOTE 3: Sampling rate is restricted to 44.1KHz family due to the reference clock for McASP2 (22.5792MHz)

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 K2GEVM sset PCM 110                             # Master Playback volume

For audio capture from Line-in:

amixer -c K2GEVM sset 'Right PGA Mixer Line1R' on
amixer -c K2GEVM sset 'Left PGA Mixer Line1L' on

3.3.4.2.5. 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.

3.3.4.2.6. Additional Information

ALSA links

Software Help

Audio hardware codecs