Our Embedded Linux BSPs use the regular Linux audio subsystem Advanced Linux Sound Architecture (ALSA). The ALSA subsystem can be configured using amixer or alsamixer whereas playing and recording is possible using the ALSA utilities aplay and arecord. Higher-level frameworks such as GStreamer allow to convert audio streams (e.g. sample rate) and play/record encoded audio streams such as MP3, FLAC, or Ogg/Vorbis file formats. This article provides some example commands showing how to control and use audio with a Toradex Embedded Linux BSP.
If you want to use audio interfaces on Torizon, there is a complementary article with a sample application and remarks: How to play audio on TorizonCore using Alsa and C/C++. The Torizon article does not replace the current one, it rather extends it.
You can use the speaker-test to test audio, or alternatively aplay if you have an audio file deployed to the board. See a simple example below.
List the available audio devices:
# cat /proc/asound/cards
0 [DWHDMI ]: dw-hdmi-ahb-aud - DW-HDMI
DW-HDMI rev 0x0a, irq 21
1 [imxspdif ]: imx-spdif - imx-spdif
imx-spdif
2 [sgtl5000audio ]: sgtl5000-audio - sgtl5000-audio
sgtl5000-audio
In the example above, sgtl5000audio, interface number 2, is the one connected to the headphone jack available on our carrier board. We'll use it. It may vary depending on your SoM; if you have questions, jump to the section specific to your SoM!
First, unmute the interface and adjust the volume:
# amixer -c 2 sset Headphone unmute
# amixer -c 2 set Headphone 50%
Then, play a test sound:
# speaker-test --device hw:2,0 --rate 48000 --channels 2 --format S16_LE --test wav
Or, if you have a wave file on the board, for instance, we'll use Gong.wav:
# aplay -Ddefault:sgtl5000audio Gong.wav
Bootlin has a great blog post on the topic, read Audio multi-channel routing and mixing using alsalib.
The Colibri T20 modules have a Wolfson WM9715L audio codec on board. To find pin assignment refer to the Hardware reference.
The Apalis T30, Apalis TK1 and Colibri T30 modules feature an on-module SGTL5000 audio codec:
root@apalis-t30:~# cat /proc/asound/cards
0 [apalist30sgtl50]: apalis_t30-sgtl - apalis_t30-sgtl5000
apalis_t30-sgtl5000
1 [Tegra ]: HDA-Intel - HDA NVIDIA Tegra
HDA NVIDIA Tegra at 0x70038000 irq 113
root@apalis-t30:~# aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: apalist30sgtl50 [apalis_t30-sgtl5000], device 0: SGTL5000 PCM sgtl5000-0 []
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: apalist30sgtl50 [apalis_t30-sgtl5000], device 1: SPDIF PCM dit-hifi-1 []
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Tegra [HDA NVIDIA Tegra], device 0: ALC898 Analog [ALC898 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Tegra [HDA NVIDIA Tegra], device 1: ALC898 Digital [ALC898 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Tegra [HDA NVIDIA Tegra], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
root@apalis-t30:~# aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
default:CARD=apalist30sgtl50
apalis_t30-sgtl5000,
Default Audio Device
sysdefault:CARD=apalist30sgtl50
apalis_t30-sgtl5000,
Default Audio Device
default:CARD=Tegra
HDA NVIDIA Tegra, ALC898 Analog
Default Audio Device
sysdefault:CARD=Tegra
HDA NVIDIA Tegra, ALC898 Analog
Default Audio Device
front:CARD=Tegra,DEV=0
HDA NVIDIA Tegra, ALC898 Analog
Front speakers
surround21:CARD=Tegra,DEV=0
HDA NVIDIA Tegra, ALC898 Analog
2.1 Surround output to Front and Subwoofer speakers
surround40:CARD=Tegra,DEV=0
HDA NVIDIA Tegra, ALC898 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=Tegra,DEV=0
HDA NVIDIA Tegra, ALC898 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=Tegra,DEV=0
HDA NVIDIA Tegra, ALC898 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=Tegra,DEV=0
HDA NVIDIA Tegra, ALC898 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=Tegra,DEV=0
HDA NVIDIA Tegra, ALC898 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=Tegra,DEV=0
HDA NVIDIA Tegra, ALC898 Digital
IEC958 (S/PDIF) Digital Audio Output
hdmi:CARD=Tegra,DEV=0
HDA NVIDIA Tegra, HDMI 0
HDMI Audio Output
root@apalis-t30:~# speaker-test --device hw:1,3 --rate 48000 --channels 2 --format S16_LE --test wav
speaker-test 1.0.29
Playback device is hw:1,3
Stream parameters are 48000Hz, S16_LE, 2 channels
WAV file(s)
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 64 to 16384
Period size range from 32 to 8192
Using max buffer size 16384
Periods = 4
was set period_size = 4096
was set buffer_size = 16384
0 - Front Left
1 - Front Right
Time per period = 2.745482
^CTransfer failed: Bad address
Please note that not all pixel clock frequencies allow for HDA being streamed through HDMI. E.g. at VESA VGA resolution, you may get an error reported as follows:
[ 16.245677] tegradc tegradc.1: hdmi: can't set audio to 44100 at 25175000 pix_clock
Plus the number of available channels depends on the resp. HDMI device is connected to.
root@apalis-t30:~# amixer set 'Capture' 12
Simple mixer control 'Capture',0
Capabilities: cvolume
Capture channels: Front Left - Front Right
Limits: Capture 0 - 15
Front Left: Capture 12 [80%]
Front Right: Capture 12 [80%]
root@apalis-t30:~# amixer set 'Capture Mux' 'LINE_IN'
Simple mixer control 'Capture Mux',0
Capabilities: enum
Items: 'MIC_IN' 'LINE_IN'
Item0: 'LINE_IN'
root@apalis-t30:~# arecord -D hw:0,0 -V stereo -c 2 -f S16_LE -r 48000 -t wav linein.wav
Recording WAVE 'linein.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
+00%|00%+ ^CAborted by signal Interrupt...
root@apalis-t30:~# amixer set 'Mic' 1
Simple mixer control 'Mic',0
Capabilities: volume volume-joined
Playback channels: Mono
Capture channels: Mono
Limits: 0 - 3
Mono: 1 [33%] [20.00dB]
root@apalis-t30:~# amixer set 'Capture' 12
Simple mixer control 'Capture',0
Capabilities: cvolume
Capture channels: Front Left - Front Right
Limits: Capture 0 - 15
Front Left: Capture 12 [80%]
Front Right: Capture 12 [80%]
root@apalis-t30:~# amixer set 'Capture Mux' 'MIC_IN'
Simple mixer control 'Capture Mux',0
Capabilities: enum
Items: 'MIC_IN' 'LINE_IN'
Item0: 'MIC_IN'
root@apalis-t30:~# arecord -D hw:0,0 -V mono -c 2 -f S16_LE -r 48000 -t wav mic.wav
Recording WAVE 'mic.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
# + | 59%^CAborted by signal Interrupt...
root@apalis-t30:~# amixer set 'Headphone' 113
Simple mixer control 'Headphone',0
Capabilities: pvolume
Playback channels: Front Left - Front Right
Limits: Playback 0 - 127
Mono:
Front Left: Playback 113 [89%] [5.00dB]
Front Right: Playback 113 [89%] [5.00dB]
root@apalis-t30:~# amixer set 'Headphone Mux' 'DAC'
Simple mixer control 'Headphone Mux',0
Capabilities: enum
Items: 'DAC' 'LINE_IN'
Item0: 'DAC'
root@apalis-t30:~# amixer set 'PCM' 144
Simple mixer control 'PCM',0
Capabilities: pvolume
Playback channels: Front Left - Front Right
Limits: Playback 0 - 192
Mono:
Front Left: Playback 144 [75%]
Front Right: Playback 144 [75%]
root@apalis-t30:~# aplay -D hw:0,0 linein.wav
Playing WAVE 'linein.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
The Colibri T20 provides SPDIF signals and Audio through HDMI.
Note that the HDMI sound card is only available if the video output is configured for HDMI and a monitor is attached.
root@colibri-t20:~# speaker-test --device hw:0,1 --rate 48000 --channels 2 --format S16_LE --test wav
Please note that not all pixel clock frequencies allow for SPDIF being streamed through HDMI. E.g. at VESA VGA resolution, you may get an error reported as follows:
[ 16.245677] tegradc tegradc.1: hdmi: can't set audio to 44100 at 25175000 pix_clock
First ALSA needs to be configured correctly using amixer.
root@colibri_t20:~# amixer set 'Left Capture Select' 'Line'
Simple mixer control 'Left Capture Select',0
Capabilities: enum
Items: 'Mic' 'NC' 'NC' 'Speaker Mixer' 'Line' 'Headphone Mixer' 'Phone Mixer' 'Phone'
Item0: 'Line'
root@colibri_t20:~# amixer set 'Right Capture Select' 'Line'
Simple mixer control 'Right Capture Select',0
Capabilities: enum
Items: 'Mic' 'NC' 'NC' 'Speaker Mixer' 'Line' 'Headphone Mixer' 'Phone Mixer' 'Phone'
Item0: 'Line'
root@colibri_t20:~# amixer set 'Capture ADC' on
Simple mixer control 'Capture ADC',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [on]
root@colibri_t20:~# amixer set 'Left HP Mixer Line Bypass' on
Simple mixer control 'Left HP Mixer Line Bypass',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [on]
root@colibri_t20:~# amixer set 'Right HP Mixer Line Bypass' on
Simple mixer control 'Right HP Mixer Line Bypass',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [on]
Recording can then be done easily using GStreamer.
root@colibri_t20:~# gst-launch alsasrc device=hw:0,0 num-buffers=1000 ! audioconvert ! audio/x-raw-int,channels=2,width=16,depth=16 ! audioresample ! wavenc ! filesink location=linein.wav
Setting pipeline to PAUSED ...
[ 1954.087441] i2s_set_channel_bit_count: enabling non-symmetric mode
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstAudioSrcClock
Got EOS from element "pipeline0".
Execution ended after 11613432000 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
First ALSA needs to be configured correctly using amixer.
root@apalis_t30:~# amixer -c 1 set 'Capture' 100 on
Simple mixer control 'Capture',0
Capabilities: cvolume cswitch
Capture channels: Front Left - Front Right
Limits: Capture 0 - 63
Front Left: Capture 63 [100%] [30.00dB] [on]
Front Right: Capture 63 [100%] [30.00dB] [on]
root@apalis_t30:~# amixer -c 1 set 'Input Source' 'Line'
Simple mixer control 'Input Source',0
Capabilities: cenum
Items: 'Rear Mic' 'Front Mic' 'Line' 'CD'
Item0: 'Line'
root@apalis_t30:~# amixer -c 1 set 'Line' 100 on
Simple mixer control 'Line',0
Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right
Limits: Playback 0 - 31
Mono:
Front Left: Playback 31 [100%] [12.00dB] [on]
Front Right: Playback 31 [100%] [12.00dB] [on]
Recording can then be done easily using GStreamer.
root@apalis_t30:~# gst-launch alsasrc device=hw:1,0 num-buffers=1000 ! audioconvert ! audio/x-raw-int,channels=2,width=16,depth=16 ! audioresample ! wavenc ! filesink location=linein.wav
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstAudioSrcClock
^CCaught interrupt -- handling interrupt.
Interrupt: Stopping pipeline ...
Execution ended after 3256371004 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
root@colibri_t20:~# amixer set 'Left Capture Select' 'Mic'
Simple mixer control 'Left Capture Select',0
Capabilities: enum
Items: 'Mic' 'NC' 'NC' 'Speaker Mixer' 'Line' 'Headphone Mixer' 'Phone Mixer' 'Phone'
Item0: 'Mic'
root@colibri_t20:~# amixer set 'Right Capture Select' 'Mic'
Simple mixer control 'Right Capture Select',0
Capabilities: enum
Items: 'Mic' 'NC' 'NC' 'Speaker Mixer' 'Line' 'Headphone Mixer' 'Phone Mixer' 'Phone'
Item0: 'Mic'
root@colibri_t20:~# amixer set 'Capture ADC' on
Simple mixer control 'Capture ADC',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [on]
root@colibri_t20:~# amixer set 'Left HP Mixer Mic Sidetone' on
Simple mixer control 'Left HP Mixer Mic Sidetone',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [on]
root@colibri_t20:~# amixer set 'Right HP Mixer Mic Sidetone' on
Simple mixer control 'Right HP Mixer Mic Sidetone',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [on]
root@colibri_t20:~# gst-launch alsasrc device=hw:0,0 num-buffers=1000 ! audioconvert ! audio/x-raw-int,channels=2,width=16,depth=16 ! audioresample ! wavenc ! filesink location=mic.wav
Setting pipeline to PAUSED ...
[ 2037.976550] i2s_set_channel_bit_count: enabling non-symmetric mode
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstAudioSrcClock
Got EOS from element "pipeline0".
Execution ended after 11613453000 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
Apalis Evaluation Board: Audio jack X27 bottom pink
root@apalis_t30:~# amixer -c 1 set 'Capture' 100 on
Simple mixer control 'Capture',0
Capabilities: cvolume cswitch
Capture channels: Front Left - Front Right
Limits: Capture 0 - 63
Front Left: Capture 63 [100%] [30.00dB] [on]
Front Right: Capture 63 [100%] [30.00dB] [on]
root@apalis_t30:~# amixer -c 1 set 'Input Source' 'Rear Mic'
Simple mixer control 'Input Source',0
Capabilities: cenum
Items: 'Rear Mic' 'Front Mic' 'Line' 'CD'
Item0: 'Rear Mic'
root@apalis_t30:~# amixer -c 1 set 'Rear Mic' 100 on
Simple mixer control 'Rear Mic',0
Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right
Limits: Playback 0 - 31
Mono:
Front Left: Playback 31 [100%] [12.00dB] [on]
Front Right: Playback 31 [100%] [12.00dB] [on]
root@apalis_t30:~# amixer -c 1 set 'Rear Mic Boost' 100 on
Simple mixer control 'Rear Mic Boost',0
Capabilities: volume
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: 0 - 3
Front Left: 3 [100%] [30.00dB]
Front Right: 3 [100%] [30.00dB]
root@apalis_t30:~# gst-launch alsasrc device=hw:1,0 num-buffers=1000 ! audioconvert ! audio/x-raw-int,channels=2,width=16,depth=16 ! audioresample ! wavenc ! filesink location=mic.wav
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstAudioSrcClock
^CCaught interrupt -- handling interrupt.
Interrupt: Stopping pipeline ...
Execution ended after 3256371004 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
MECS Tellurium: Audio jack X11 pin 4 & 5 (HEADPHONE_LF/RF)
root@colibri_t20:~# amixer set 'Headphone' 8 on
Simple mixer control 'Headphone',0
Capabilities: pvolume pswitch pswitch-joined penum
Playback channels: Front Left - Front Right
Limits: Playback 0 - 31
Mono:
Front Left: Playback 8 [26%] [on]
Front Right: Playback 8 [26%] [on]
root@colibri_t20:~# amixer set 'Left HP Mixer PCM' on
Simple mixer control 'Left HP Mixer PCM',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [on]
root@colibri_t20:~# amixer set 'Right HP Mixer PCM' on
Simple mixer control 'Right HP Mixer PCM',0
Capabilities: pswitch pswitch-joined penum
Playback channels: Mono
Mono: Playback [on]
root@colibri_t20:~# aplay bach.au
[ 2924.023213] i2s_set_channel_bit_count: enabling non-symmetric mode
Playing Sparc Audio 'bach.au' : Mu-Law, Rate 8000 Hz, Mono
root@colibri_t20:~# gst-launch filesrc location=mic.wav ! wavparse ! audioconvert ! alsasink device=hw:0,0
Setting pipeline to PAUSED ...
[ 2459.620319] i2s_set_channel_bit_count: enabling non-symmetric mode
Pipeline is PREROLLING ...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstAudioSinkClock
Got EOS from element "pipeline0".
Execution ended after 11611505001 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
Apalis Evaluation Board: Audio jack X27 middle green
root@apalis_t30:~# amixer -c 1 set 'Front' 100 on
Simple mixer control 'Front',0
Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right
Limits: Playback 0 - 87
Mono:
Front Left: Playback 87 [100%] [0.00dB] [on]
Front Right: Playback 87 [100%] [0.00dB] [on]
root@apalis_t30:~# gst-launch filesrc location=mic.wav ! wavparse ! audioconvert ! alsasink device=hw:1,0
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstAudioSinkClock
Got EOS from element "pipeline0".
Execution ended after 3258022000 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
The following example shows how to playback MP3 audio via GStreamer using hardware-accelerated MP3 decoding. Please note that the two numbers at the end specify which ALSA card and device to use for audio (e.g. alsasink device=hw:0,0 for WM9715L AC97 through headphones and alsasink device=hw:1,0 for SPDIF through HDMI).
root@colibri-t20:~# gst-launch filesrc location=/home/root/03_Scar_Tissue.mp3 ! mpegaudioparse ! nv_omx_mp3dec ! alsasink device=hw:0,0
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
[26776.345823] avp_init: Using AVP MMU to relocate AVP kernel
[26776.428892] avp_init: Reading firmware from 'nvrm_avp.bin' (46612 bytes)
[26776.441076] avp_init: Loading AVP kernel at vaddr=d7a00000 paddr=18100000
[26776.488814] avp_reset: Resetting AVP: reset_addr=100000
[26776.506561] process_connect_locked: got connect (111794)
[26776.506746] avp_node_try_connect: trying connect from RPC_AVP_PORT
[26776.518587] avp_svc_thread: got remote peer
[26776.523229] [AVP]: AVP kernel (Nov 2 2012 16:47:31)
[26776.537301] avp_node_try_connect: got conn ack 'RPC_AVP_PORT' (cd049b60 <-> 111758)
[26776.545378] avp_init: avp init done
[26776.548917] avp_lib: loading library 'nvmm_manager.axf'
[26776.596783] avp_lib: Successfully loaded library nvmm_manager.axf (lib_id=118960)
[26776.605064] avp_node_try_connect: trying connect from NVMM_MANAGER_SRV
[26776.621815] avp_node_try_connect: got conn ack 'NVMM_MANAGER_SRV' (cdb48f80 <-> 119c00)
[26776.714101] tegra20_ac97_hw_params(): dai->id=0, play
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstAudioSinkClock
[26776.803786] tegra20_ac97_trigger()
[26776.807247] tegra20_ac97_start_playback()
[26776.811295] ac97_fifo_set_attention_level()
[26776.815502] ac97_slot_enable()
Got EOS from element "pipeline0".
Execution ended after 236898907001 ns.
Setting pipeline to PAUSED ...
[27013.681352] tegra20_ac97_trigger()
[27013.684795] tegra20_ac97_stop_playback()
[27013.688743] ac97_fifo_set_attention_level()
[27013.693968] ac97_slot_enable()
Setting pipeline to READY ...
[27013.719878] process_disconnect_locked: got disconnect (cdb48f80)
[27013.741936] avp_lib: Successfully unloaded 'nvmm_manager.axf'
[27013.748859] avp_lib: unloaded 'nvmm_manager.axf'
[27013.754949] avp_svc_thread: AVP seems to be down; wait for kthread_stop
[27013.762933] avp_svc_thread: exiting
[27013.767639] avp_uninit: avp teardown done
Setting pipeline to NULL ...
Freeing pipeline ...
All modules based on the NXP/Freescale i.MX 6 SoC or i.MX 7 SoC provides an on-module NXP/Freescale SGTL5000 audio codec. The SPDIF signals and Audio through HDMI are available only on modules based on i.MX 6 Soc.
HW accelerated decoding and encoding are provided by GStreamer plugins.
Note that the HDMI sound card is only available if the video output is configured for HDMI and a monitor is attached.
root@apalis-imx6:~# cat /proc/asound/cards
0 [imx6qapalissgtl]: imx6q-apalis-sg - imx6q-apalis-sgtl5000
imx6q-apalis-sgtl5000
1 [imxspdif ]: imx-spdif - imx-spdif
imx-spdif
2 [imxhdmisoc ]: imx-hdmi-soc - imx-hdmi-soc
imx-hdmi-soc
root@apalis-imx6:~# aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
sysdefault:CARD=imx6qapalissgtl
imx6q-apalis-sgtl5000,
Default Audio Device
sysdefault:CARD=imxspdif
imx-spdif,
Default Audio Device
sysdefault:CARD=imxhdmisoc
imx-hdmi-soc,
Default Audio Device
surround40:CARD=imxhdmisoc,DEV=0
imx-hdmi-soc,
4.0 Surround output to Front and Rear speakers
...
The SGTL5000 is the default sound device, thus one does not need to specify it explicitly on the command line.
Playback of a wave file:
root@apalis-imx6:~# aplay Gong.wavPlaying WAVE 'Gong.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereoroot@apalis-imx6:~# aplay -D sysdefault:CARD=imx6qapalissgtl Gong.wavPlaying WAVE 'Gong.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, StereoStereo capture can be achieved using Line-in functionality. First, "Capture Select" needs to be selected correctly in ALSA mixer:
root@apalis-imx6:~# amixer set 'Capture Mux' 'LINE_IN' Simple mixer control 'Capture Mux',0 Capabilities: enum Items: 'MIC_IN' 'LINE_IN' Item0: 'LINE_IN'
Then recording in various sample rates is natively possible:
root@apalis-imx6:~# arecord -r 48000 -f S16_LE -c 2 Capture48kHz.wav Recording WAVE 'Capture48kHz.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo ^C root@apalis-imx6:~# arecord -r 8000 -f S16_LE -c 2 Capture8kHz.wav Recording WAVE 'Capture8kHz.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo ^C
Change the input channel to MIC_IN:
root@apalis-imx6:~# amixer set 'Capture Mux' 'MIC_IN' ...
In contrast to Line-in, Microphone capture is mono anyway, hence we can save some space by only recording one channel. The argument -V offers a VU meter which helps to see whether there is any input and if its volume is at a reasonable level.
root@apalis-imx6:~# arecord -V mono -r 8000 -f S16_LE -c 1 Capture8kHz.wav Recording WAVE 'Capture8kHz.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono ### + | 94% ^C
Apalis features dedicated SPDIF pins and the Apalis Evaluation board provides an Optical Transmitter. Thus the Apalis iMX6 image has the SPDIF feature enabled in our default device tree, while for the Colibri iMX6 and Colibri iMX7 it is not enabled.
root@apalis-imx6:~# aplay -D sysdefault:CARD=imxspdif Gong.wav Playing WAVE 'Gong.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
In addition to the video data stream, HDMI is able to carry audio data as well. Note that depending on the video resolution/framerate used, that feature might not be available. HDMI Playback is currently unavailable for Colibri i.MX7.
root@apalis-imx6:~# aplay -D sysdefault:CARD=imxhdmisoc Gong.wav Playing WAVE 'Gong.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
The following example shows how to playback MP3 audio via GStreamer using hardware-accelerated MP3 decoding. Please note that the two numbers at the end specify which ALSA card and device to use for audio (e.g. alsasink device=hw:0,0 for SGTL5000, alsasink device=hw:1,0 for SPDIF and, alsasink device=hw:2,0 for audio through HDMI, compare with /proc/asound/cards).
Additional GStreamer pipelines can be found on the Freescale community website here.
Additional encoders and decoders can be verified with GStreamer command gst-inspect.
gst-launch filesrc location=Gong.mp3 typefind=true ! beepdec ! alsasink device=hw:0,0 ...
Alternatively, using the playbin plugin, which sets up the pipeline automatically:
root@apalis-imx6:~# gst-launch playbin uri=file:///home/root/Gong.mp3 Setting pipeline to PAUSED ... Pipeline is PREROLLING ... WARNING: from element /GstPlayBin2:playbin20/GstPlaySink:playsink0: No volume control found Additional debug info: /build/linuxdev/oe-core/build/out-glibc/work/armv7at2hf-vfp-neon-mx6qdl-angstrom-linux-gnueabi/gst-plugins-base/0.10.36-r8/gst-plugins-base-0.10.36/gst/playback/gstplaysink.c(1869): gen_audio_chain (): /GstPlayBin2:playbin20/GstPlaySink:playsink0: Volume/mute is not available Beep: 3.0.11 Core: MP3 decoder Wrapper build on Mar 21 2014 15:04:50 mime: audio/mpeg, mpegversion = (int)1 file: /usr/lib/imx-mm/audio-codec/wrap/lib_mp3d_wrap_arm12_elinux.so.3 CODEC: BLN_MAD-MMCODECS_MP3D_ARM_02.13.00_CORTEX-A8 build on Apr 10 2014 15:26:15. Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstAudioSinkClock Got EOS from element "playbin0". Execution ended after 9596381667 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ...
To record and encode to MP3, the following GStreamer command can be used:
root@apalis-imx6:~# gst-launch alsasrc ! imxmp3enc ! filesink location= audio.mp3 ... ^C
The Colibri iMX6ULL does not feature an audio codec on the module. However, it is possible to access all three audio interfaces by placing an external audio codec.
However, the Colibri iMX6ULL features a Medium Quality Sound (MQS) interface that can be used to generate medium-quality audio via a standard GPIO. Please refer to the module datasheet for more information.
Note: This section currently only applies to Apalis iMX8
root@apalis-imx8:~# cat /proc/asound/cards
0 [imxhdmiarc ]: imx-hdmi-arc - imx-hdmi-arc
imx-hdmi-arc
1 [imxspdif ]: imx-spdif - imx-spdif
imx-spdif
2 [amixaudiosai ]: amix-audio-sai - amix-audio-sai
amix-audio-sai
3 [imxaudiohdmitx ]: imx-audio-hdmi- - imx-audio-hdmi-tx
imx-audio-hdmi-tx
4 [apalisimx8qmsgt]: apalis-imx8qm-s - apalis-imx8qm-sgtl5000
apalis-imx8qm-sgtl5000
root@apalis-imx8:~# aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
sysdefault:CARD=imxhdmiarc
imx-hdmi-arc,
Default Audio Device
sysdefault:CARD=imxspdif
imx-spdif,
Default Audio Device
sysdefault:CARD=amixaudiosai
amix-audio-sai,
Default Audio Device
sysdefault:CARD=imxaudiohdmitx
imx-audio-hdmi-tx,
Default Audio Device
sysdefault:CARD=apalisimx8qmsgt
apalis-imx8qm-sgtl5000,
Default Audio Device
root@apalis-imx8:~# aplay Gong.wav Playing WAVE 'Gong.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo root@apalis-imx8:~# aplay -D sysdefault:CARD=apalisimx8qmsgt Gong.wav Playing WAVE 'Gong.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
amixer set 'Capture Mux' 'LINE_IN' arecord -D hw:4,0 -V mono -c 2 -f S16_LE -r 44100 -t wav line.wav
amixer set 'Capture Mux' 'MIC_IN' amixer set 'Mic' '20.00dB' arecord -D hw:4,0 -V mono -c 2 -f S16_LE -r 44100 -t wav mic.wav
aplay -D sysdefault:CARD=imxspdif Gong.wav
This was done with an optical to the electronic medium converter and the signal hooked up to MXM3_217
arecord -D sysdefault:CARD=imxspdif -V mono -c 2 -f S32_LE -r 44100 -t wav spdif_input.wav
aplay -D sysdefault:CARD=imxaudiohdmitx Gong.wav
The Verdin iMX8M Mini and Verdin iMX8M Plus do not feature an audio codec on the module. However, it is possible to access all Synchronous Audio Interfaces (SAI) by placing an external audio codec. The SAI interfaces can be used as I2S (Recommended) or AC97. An I2S is connected to the Wi-Fi module for Bluetooth audio features. Please refer to the module datasheet for more information.
Verdin Development Board V1.0 and Dahlia Carrier Board features an analog audio interface based on the WM8904CGEFL/RV audio codec IC from Cirrus Logic.
To check the available sound cards, use the cat /proc/asound/cards command. The expected result on a Verdin iMX8M Mini:
root@verdin-imx8mm:~# cat /proc/asound/cards
0 [imx8mmwm8904 ]: imx8mm-wm8904 - imx8mm-wm8904
imx8mm-wm8904
To record audio, you might need to run the following command to unmute the capture device:
root@verdin-imx8mm:~# amixer set "Capture" cap
# amixer set 'Left Capture Inverting Mux' 'IN1L'
# amixer set 'Right Capture Inverting Mux' 'IN1R'
# amixer set 'Capture' '31'
# arecord -D hw:0,0 -V mono -c 2 -f S16_LE -r 48000 -t wav mic.wav
# amixer set 'Left Capture Inverting Mux' 'IN2L'
# amixer set 'Right Capture Inverting Mux' 'IN2R'
# amixer set 'Capture' '16'
# arecord -D hw:0,0 -V mono -c 2 -f S16_LE -r 48000 -t wav line.wav
# aplay mic.wav
# aplay line.wav
# gst-launch-1.0 playbin uri=file:///media/sda1/Gong.mp3
# aplay /media/sda1/Gong.wav
# gst-launch-1.0 playbin uri=file:///media/sda1/Gong.mp3 audio-sink="alsasink device=hw:0,0"
# aplay -D sysdefault:CARD=imx8mmwm8904 /media/sda1/Gong.wav
[Verdin Development Board] V1.1 features an analog audio interface based on the NAU8822 24-bit audio codec IC with integrated DSP with specific functions from Nuvoton:
To check the available sound cards, use the cat /proc/asound/cards command. The expected result on a Verdin iMX8M Plus:
# cat /proc/asound/cards
0 [imx8mpnau8822 ]: imx8mp-nau8822 - imx8mp-nau8822
imx8mp-nau8822
Note that without a connected mic on X22 the onboard one is used.
# amixer set 'PGA',0 '63'
# amixer set 'PGA Boost',0 '0'
# amixer set 'Aux Boost',0 '0'
# arecord -D hw:0,0 -V mono -c 2 -f S16_LE -r 48000 -t wav mic.wav
X21 what is played from AUX_OUT, X14. → connect X21 with X14
# amixer set 'PGA',0 '0'
# amixer set 'PGA Boost',0 '0'
# amixer set 'Aux Boost',0 '5'
# gst-launch-1.0 audiotestsrc ! audio/x-raw, rate=48000 ! alsasink -v &
# arecord -D hw:0,0 -V mono -c 2 -f S16_LE -r 48000 -t wav line.wav
<ctrl C>
# killall gst-launch-1.0
The speaker output can be used in a Mono, bridged mode configuration (One loudspeaker on X28/X29) or in a Stereo configuration (Two loudspeakers on X13)
# amixer set 'Speaker RInversion',0 on
# amixer set 'Speaker RInversion',0 off
aplay mic.wav
aplay line.wav
gst-launch-1.0 playbin uri=file:///home/root/sound/Gong.mp3
aplay /home/root/sound/Gong.wav
gst-launch-1.0 playbin uri=file:///home/root/sound/Gong.mp3 audio-sink="alsasink device=hw:0,0"
aplay -D sysdefault:CARD=imx8mpnau8822 /home/root/sound/Gong.wap
Within the Vybrid module family, only Colibri VF61 supports analog audio. Audio is handled by the onboard Wolfson WM9715L audio codec connected via a AC97 audio link to the Vybrid SoC. The audio codec driver is called wm9712-codec, since it was initially created for the almost identical WM9712L codec.
Playback of a WAVE file:
root@colibri-vf:~# aplay Test48kHz.wav Playing WAVE 'Test48kHz.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo root@colibri-vf:~# aplay Test44.1kHz.wav Playing WAVE 'Test44.1kHz.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
Note: Some parts of the AC97 protocol are realized in software. To make Software AC97 efficient, a rather large buffer is used to prepare several AC97 frames in a batch. However, using buffering does not allow the use of the variable sample rate (VRA) feature of the AC97 link during playback since that would require a response that depends on the request flag of the last received frame. Therefore, the hardware is only able to transfer 48kHz sample rate PCM streams. The ALSA audio stack is able to re-sample the sample rate dynamically, the only difference is that a file with a sample rate other than 48kHz requires somewhat more CPU resources while playback (<1% vs ~3%). Implementing the AC97 protocol without buffering would lead to a much higher CPU load than relying on ALSA's re-sampling feature.
The hardware parameter of the PCM stream during playback can be seen via the proc file system:
root@colibri-vf:~# cat /proc/asound/card0/pcm0p/sub0/hw_params access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 1024 buffer_size: 4096
Playback of encoded audio formats can be achieved using GStreamer. The Colibri VF61 root filesystem does not provide GStreamer by default, however it is installable via Angstrom feeds:
root@colibri-vf:~# opkg install gst-plugins-base-playbin gst-plugins-base-alsa gst-plugins-base-audioconvert gst-plugins-base-audioresample gst-plugins-base-vorbis gst-plugins-base-ogg gst-plugins-good-wavenc
The following GStreamer pipeline allows to playback Ogg Vorbis encoded audio files:
root@colibri-vf:~# gst-launch filesrc location=./vorbis.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
The WM9715L has two audio ADCs hence allowing stereo audio capture. The input to these capture ADCS can be individually selected through the ALSA Capture Select controls. Capture volume and boost are handled after the channel selection, hence the volume applies for the Microphone as well as for Line-In capturing.
Two record volume modes can be selected, a standard AC97 1.5dB gain steps and an extended mode using 0.75dB gain steps. The extended mode (0.75dB gain steps) has also a larger gain range (-17.25-+30db), so we recommend using the extended mode.
root@colibri-vf:~# amixer set 'Capture Volume Steps' '+0.75dB Steps'
The record volume can then be set to a value between 0-63 (which mappings linearly to the gain ranges above).
root@colibri-vf:~# amixer set 'Capture' 50
Depending on the input (e.g. microphones) a boost might be necessary to increase the ADC signal by +20dB.
root@colibri-vf:~# amixer set 'Capture Boost' unmute
Stereo capture can be achieved using Line-in functionality. First, the "Capture Select" needs to be selected correctly in ALSA mixer:
root@colibri-vf:~# amixer set 'Left Capture Select' 'Line'
Simple mixer control 'Left Capture Select',0
Capabilities: enum
Items: 'Mic' 'NC' 'NC' 'Speaker Mixer' 'Line' 'Headphone Mixer' 'Phone Mixer' 'Phone'
Item0: 'Line'
root@colibri-vf:~# amixer set 'Right Capture Select' 'Line'
...
Then recording in various sample rates is natively possible:
root@colibri-vf:~# arecord -r 48000 -V mono -f S16_LE -c 2 Capture48kHz.wav
Recording WAVE 'Capture48kHz.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
^C
root@colibri-vf:~# arecord -r 8000 -f S16_LE -c 2 Capture8kHz.wav
Recording WAVE 'Capture8kHz.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo
^C
To record the Ogg Vorbis audio format, the following GStreamer command can be used
root@colibri-vf:~# gst-launch alsasrc do-timestamp=true ! audioconvert ! vorbisenc quality=0.4 ! oggmux ! filesink location=vorbis.ogg
...
root@colibri-vf:~# amixer set 'Left Capture Select' 'Mic'
...
root@colibri-vf:~# amixer set 'Right Capture Select' 'Mic'
...
The Colibri standard has only one microphone input, which is connected to Mic 1 of the audio codec. The codec internally samples the same signal with both ADCs, hence the ALSA stack receives a stereo signal. However, since the signals are the same, we can save some space by only recording one channel. The argument -V offers a VU meter which helps to see whether there is any input and if its volume is on a reasonable level.
root@colibri-vf:~# arecord -V mono -r 8000 -f S16_LE -c 1 Capture8kHz.wav
####### + | 38%
^C
Note: The volume controls "Mic 1" and "Mic 2" do not change the volume when capturing from the microphone. Those volume controls are only active if the sidetone path is enabled (e.g. direct path from microphone to headphone). Use the main capture volume to control the microphone volume.
CONFIG_SND_USB_AUDIO is enabled, so you may use USB audio hardware.
If you get the following error when recording from USB, enable the CONFIG_USB_EHCI_TT_NEWSCHED kernel configuration.
root@colibri_t20:~# arecord -D hw:3,0 -c 2 -f S16_LE -r 44100 -t wav usbin.wav
Recording WAVE 'usbin.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
[ 3187.734697] cannot submit datapipe for urb 0, error -28: not enough bandwidth
