FAQs
English (US) 日本語 | Japanese 中文 | Chinese (Simplified)

imx8 yocto pulseaudio doesn't work for gstreamer

cancel
Turn on suggestions
Auto-suggest helps you quickly narrow down your search results by suggesting possible matches as you type.
Showing results for 
Show  only  | Search instead for 
Did you mean: 

imx8 yocto pulseaudio doesn't work for gstreamer

‎09-04-2023 03:26 AM
3,445 Views
Anton_K
Contributor III

Hello!
I have some problems with pulseaudio for gst-launch-1.0.

Actually I can use simple command line test with following logs:

 

$> GST_DEBUG=3,pulse*:9 gst-launch-1.0 pulsesrc ! autoaudiosink
Setting pipeline to PAUSED ...
0:00:00.146182796   732 0xaaaab6d3fb30 DEBUG                  pulse pulsesink.c:2226:gst_pulsesink_query_getcaps:<autoaudiosink0-actual-sink-pulse> caps audio/x-raw, format=(string){ S16LE, S16BE, F32LE, F32BE, S32LE, S32BE, S24LE, S24BE, S24_32LE, S24_32BE, U8 }, layout=(string)interleaved, rate=(int)[ 1, 384000 ], channels=(int)[ 1, 32 ]; audio/x-alaw, rate=(int)[ 1, 384000 ], channels=(int)[ 1, 32 ]; audio/x-mulaw, rate=(int)[ 1, 384000 ], channels=(int)[ 1, 32 ]; audio/x-ac3, framed=(boolean)true; audio/x-eac3, framed=(boolean)true; audio/x-dts, framed=(boolean)true, block-size=(int){ 512, 1024, 2048 }; audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)[ 1, 3 ], parsed=(boolean)true; audio/mpeg, mpegversion=(int){ 2, 4 }, framed=(boolean)true, stream-format=(string)adts
0:00:00.146336422   732 0xaaaab6d3fb30 INFO                   pulse pulsesink.c:3254:gst_pulsesink_change_state:<autoaudiosink0-actual-sink-pulse> new pa main loop thread
0:00:00.146838551   732 0xaaaab6d3fb30 DEBUG                  pulse pulsesink.c:1836:gst_pulsesink_create_ringbuffer:<autoaudiosink0-actual-sink-pulse> creating ringbuffer
0:00:00.147029803   732 0xaaaab6d3fb30 DEBUG                  pulse pulsesink.c:1838:gst_pulsesink_create_ringbuffer:<autoaudiosink0-actual-sink-pulse> created ringbuffer @0xaaaab6e40030
0:00:00.147105553   732 0xaaaab6d3fb30 INFO                   pulse pulsesink.c:533:gst_pulseringbuffer_open_device:<autoaudiosink0-actual-sink-pulse> new context with name gst-launch-1.0, pbuf=0xaaaab6e40030, pctx=0xaaaab6e26980
0:00:00.148039936   732 0xaaaab6d3fb30 LOG                    pulse pulsesink.c:550:gst_pulseringbuffer_open_device:<autoaudiosink0-actual-sink-pulse> connect to server (NULL)
0:00:00.148443189   732 0xaaaab6d3fb30 LOG                    pulse pulsesink.c:427:gst_pulsering_context_state_cb: got new context state 1
0:00:00.152361345   732 0xaaaab6d3fb30 LOG                    pulse pulsesink.c:427:gst_pulsering_context_state_cb: got new context state 5
0:00:00.152458346   732 0xaaaab6d3fb30 LOG                    pulse pulsesink.c:433:gst_pulsering_context_state_cb: signaling
0:00:00.152554472   732 0xaaaab6d3fb30 WARN                   pulse pulsesink.c:617:gst_pulseringbuffer_open_device:<autoaudiosink0-actual-sink-pulse> error: Failed to connect: Connection refused
0:00:00.152844599   732 0xaaaab6d3fb30 DEBUG                  pulse pulsesink.c:340:gst_pulsering_destroy_context:<pulseringbuffer0> destroying ringbuffer 0xaaaab6e40030
0:00:00.152908100   732 0xaaaab6d3fb30 DEBUG                  pulse pulsesink.c:354:gst_pulsering_destroy_context:<pulseringbuffer0> releasing context with name gst-launch-1.0, pbuf=0xaaaab6e40030, pctx=0xaaaab6e26980
0:00:00.152959725   732 0xaaaab6d3fb30 DEBUG                  pulse pulsesink.c:360:gst_pulsering_destroy_context:<pulseringbuffer0> destroying final context with name gst-launch-1.0, pbuf=0xaaaab6e40030, pctx=0xaaaab6e26980
0:00:00.153154977   732 0xaaaab6d3fb30 DEBUG                  pulse pulsesink.c:340:gst_pulsering_destroy_context:<pulseringbuffer0> destroying ringbuffer 0xaaaab6e40030
0:00:00.153214352   732 0xaaaab6d3fb30 INFO                   pulse pulsesink.c:3236:gst_pulsesink_release_mainloop:<autoaudiosink0-actual-sink-pulse> terminating pa main loop thread
0:00:00.231180477   732 0xaaaab6d3fb30 DEBUG                  pulse pulsesrc.c:997:gst_pulsesrc_open:<pulsesrc0> opening device
0:00:00.231943233   732 0xaaaab6d3fb30 DEBUG                  pulse pulsesrc.c:1012:gst_pulsesrc_open:<pulsesrc0> connect to server (NULL)
0:00:00.236367893   732 0xaaaab6d3fb30 WARN                   pulse pulsesrc.c:1016:gst_pulsesrc_open:<pulsesrc0> error: Failed to connect: Connection refused
ERROR: from element /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: Failed to connect: Connection refused
Additional debug info:
../git/ext/pulse/pulsesrc.c(1016): gst_pulsesrc_open (): /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0
ERROR: pipeline doesn't want to preroll.
Failed to set pipeline to PAUSED.
Setting pipeline to NULL ...
Freeing pipeline ...

 

 I have no idea what audio server needs for pulsesrc, probably it needs some /dev/audioX, but on this machine there is no /dev/audioX available, may be there is some issue here.

Also I have this audio devices on gst-device-monitor-1.0:

root@imx8qxp-var-som:~# gst-device-monitor-1.0 
Probing devices...


(gst-device-monitor-1.0:520): GStreamer-CRITICAL **: 08:36:22.486: gst_element_message_full_with_details: assertion 'GST_IS_ELEMENT (element)' failed

(gst-device-monitor-1.0:520): GStreamer-CRITICAL **: 08:36:22.499: gst_element_message_full_with_details: assertion 'GST_IS_ELEMENT (element)' failed

...

Device found:

	name  : wm8904-audio
	class : Audio/Sink
	caps  : audio/x-raw, format={ (string)S32LE, (string)S24_32LE, (string)S16LE }, layout=interleaved, rate=[ 8000, 96000 ], channels=2, channel-mask=0x0000000000000003
	properties:
		device.api = alsa
		device.class = sound
		alsa.card = 0
		alsa.card_name = wm8904-audio
		alsa.driver_name = simple-card
		alsa.name = wm8904-audio
		alsa.id = wm8904audio
		alsa.mixername = ""
		alsa.components = ""
	gst-launch-1.0 ... ! alsasink device="hw:0\,0"


Device found:

	name  : wm8904-audio
	class : Audio/Source
	caps  : audio/x-raw, format={ (string)S32LE, (string)S24_32LE, (string)S16LE }, layout=interleaved, rate=[ 8000, 96000 ], channels=2, channel-mask=0x0000000000000003
	properties:
		device.api = alsa
		device.class = sound
		alsa.card = 0
		alsa.card_name = wm8904-audio
		alsa.driver_name = simple-card
		alsa.name = wm8904-audio
		alsa.id = wm8904audio
		alsa.mixername = ""
		alsa.components = ""
	gst-launch-1.0 alsasrc device="hw:0\,0" ! ...


I have tried also with alsasrc record and play:

root@imx8qxp-var-som:~# GST_DEBUG=3,alsa*:9 gst-launch-1.0 -v alsasrc device="hw:0,0" ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
0:00:00.123746529   864 0xaaaafccd5c70 DEBUG                   alsa gstalsaelement.c:70:alsa_element_init: binding text domain gst-plugins-base-1.0 to locale dir /usr/share/locale
0:00:00.125168673   864 0xaaaafccd5c70 DEBUG                   alsa gstalsasrc.c:300:gst_alsasrc_init:<GstAlsaSrc@0xaaaafcdb8270> initializing
0:00:00.140407715   864 0xaaaafccd5c70 DEBUG                   alsa gstalsasrc.c:328:gst_alsasrc_getcaps:<alsasrc0> device not open, using template caps
0:00:00.140810881   864 0xaaaafccd5c70 DEBUG                   alsa gstalsasrc.c:328:gst_alsasrc_getcaps:<alsasrc0> device not open, using template caps
0:00:00.141142165   864 0xaaaafccd5c70 DEBUG                   alsa gstalsasrc.c:328:gst_alsasrc_getcaps:<alsasrc0> device not open, using template caps
0:00:00.141795106   864 0xaaaafccd5c70 DEBUG                   alsa gstalsasrc.c:328:gst_alsasrc_getcaps:<alsasrc0> device not open, using template caps
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
0:00:00.155190712   864 0xaaaafcb23b60 FIXME                default gstutils.c:4025:gst_pad_create_stream_id_internal:<alsasrc0:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id
0:00:00.155554499   864 0xaaaafcb23b60 INFO                    alsa gstalsasrc.c:346:gst_alsasrc_getcaps:<alsasrc0> template caps audio/x-raw, format=(string){ F64LE, F64BE, F32LE, F32BE, S32LE, S32BE, U32LE, U32BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, S16LE, S16BE, U16LE, U16BE, S8, U8 }, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-iec958, format=(string)U32LE, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)2
0:00:00.155851654   864 0xaaaafcb23b60 DEBUG                   alsa gstalsa.c:196:gst_alsa_detect_formats:<alsasrc0> audio/x-iec958
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
0:00:00.155952789   864 0xaaaafcb23b60 LOG                     alsa gstalsa.c:30:gst_alsa_detect_rates:<alsasrc0> probing sample rates ...
0:00:00.156008295   864 0xaaaafcb23b60 DEBUG                   alsa gstalsa.c:49:gst_alsa_detect_rates:<alsasrc0> Min. rate = 8000 (8000)
0:00:00.156054924   864 0xaaaafcb23b60 DEBUG                   alsa gstalsa.c:50:gst_alsa_detect_rates:<alsasrc0> Max. rate = 96000 (96000)
0:00:00.156100804   864 0xaaaafcb23b60 LOG                     alsa gstalsa.c:372:gst_alsa_detect_channels:<alsasrc0> probing channels ...
0:00:00.156161685   864 0xaaaafcb23b60 DEBUG                   alsa gstalsa.c:416:gst_alsa_detect_channels:<alsasrc0> Min. channels = 2 (2)
0:00:00.156208315   864 0xaaaafcb23b60 DEBUG                   alsa gstalsa.c:417:gst_alsa_detect_channels:<alsasrc0> Max. channels = 2 (2)
0:00:00.156328827   864 0xaaaafcb23b60 INFO                    alsa gstalsasrc.c:356:gst_alsasrc_getcaps:<alsasrc0> returning caps audio/x-raw, format=(string){ S32LE, S24_32LE, S16LE }, layout=(string)interleaved, rate=(int)[ 8000, 96000 ], channels=(int)2, channel-mask=(bitmask)0x0000000000000003
New clock: GstAudioSrcClock
0:00:00.158278400   864 0xaaaafcb23b60 DEBUG                   alsa gstalsasrc.c:406:set_hwparams:<alsasrc0> buffer time 200000, min 2993, max 2972155
0:00:00.158365408   864 0xaaaafcb23b60 DEBUG                   alsa gstalsasrc.c:412:set_hwparams:<alsasrc0> period time 10000, min 1496, max 185715
0:00:00.158448292   864 0xaaaafcb23b60 DEBUG                   alsa gstalsasrc.c:418:set_hwparams:<alsasrc0> periods min 2, max 1985
0:00:00.158584931   864 0xaaaafcb23b60 DEBUG                   alsa gstalsasrc.c:433:set_hwparams:<alsasrc0> period time 10068 buffer time 201360 set correctly
0:00:00.182139690   864 0xaaaafcb23b60 DEBUG                   alsa gstalsasrc.c:478:set_hwparams:<alsasrc0> buffer size : 8880
0:00:00.182233075   864 0xaaaafcb23b60 DEBUG                   alsa gstalsasrc.c:481:set_hwparams:<alsasrc0> period size : 444
0:00:00.182459348   864 0xaaaafcb23b60 DEBUG                   alsa gstalsasrc.c:834:gst_alsasrc_prepare:<alsasrc0> Hardware setup: 
  stream       : CAPTURE
  access       : RW_INTERLEAVED
  format       : S16_LE
  subformat    : STD
  channels     : 2
  rate         : 44100
  exact rate   : 44100 (44100/1)
  msbits       : 16
  buffer_size  : 8880
  period_size  : 444
  period_time  : 10068

0:00:00.182580360   864 0xaaaafcb23b60 DEBUG                   alsa gstalsasrc.c:839:gst_alsasrc_prepare:<alsasrc0> Software setup: 
  tstamp_mode  : ENABLE
  tstamp_type  : MONOTONIC
  period_step  : 1
  avail_min    : 444
  period_event : 0
  start_threshold  : 0
  stop_threshold   : 8880
  silence_threshold: 0
  silence_size : 0
  boundary     : 4998995586381250560

0:00:00.182830135   864 0xaaaafcb23b60 WARN                    alsa pcm_hw.c:1360:snd_pcm_hw_get_chmap: alsalib error: Cannot read Channel Map ctl
: No such file or directory
0:00:00.182935146   864 0xaaaafcb23b60 LOG                     alsa gstalsa.c:860:alsa_detect_channels_mapping:<alsasrc0> ALSA driver does not implement channels mapping API
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 201360
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 10068
Redistribute latency...
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps = audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = audio/x-raw, rate=(int)44100, channels=(int)2, format=(string)F32LE, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0.GstPad:sink: caps = audio/x-raw, rate=(int)44100, channels=(int)2, format=(string)F32LE, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0: last-message = encoding at quality level 0.30
Redistribute latency...
0:00:00.246260682   864 0xaaaafcb23b60 FIXME               basesink gstbasesink.c:3395:gst_base_sink_default_event:<filesink0> stream-start event without group-id. Consider implementing group-id handling in the upstream elements
0:00:00.246644846   864 0xaaaafcb23b60 WARN            audioencoder gstaudioencoder.c:980:gst_audio_encoder_finish_frame:<vorbisenc0> Can't copy metadata because input buffer disappeared

Some scope of issues But it tries to record with time counting, then I'm trying to play and no sound on output:

root@imx8qxp-var-som:~# GST_DEBUG=3,alsa*:9 gst-launch-1.0 filesrc location=alsasrc.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! autoaudiosink
Setting pipeline to PAUSED ...
0:00:00.164507605   885 0xaaab1e302810 WARN                   pulse pulsesink.c:617:gst_pulseringbuffer_open_device:<autoaudiosink0-actual-sink-pulse> error: Failed to connect: Connection refused
0:00:00.169726634   885 0xaaab1e302810 DEBUG                   alsa gstalsaelement.c:70:alsa_element_init: binding text domain gst-plugins-base-1.0 to locale dir /usr/share/locale
0:00:00.170570574   885 0xaaab1e302810 DEBUG                   alsa gstalsasink.c:266:gst_alsasink_init:<GstAlsaSink@0xaaab1e325a30> initializing alsasink
0:00:00.170680208   885 0xaaab1e302810 DEBUG                   alsa gstalsasink.c:308:gst_alsasink_getcaps:<autoaudiosink0-actual-sink-alsa> device not open, using template caps
0:00:00.234804418   885 0xaaab1e302810 LOG                     alsa gstalsasink.c:911:gst_alsasink_open:<autoaudiosink0-actual-sink-alsa> Opened device default
0:00:00.241689951   885 0xaaab1e302810 LOG                     alsa gstalsasink.c:911:gst_alsasink_open:<autoaudiosink0-actual-sink-alsa> Opened device default
0:00:00.243218194   885 0xaaab1e302810 WARN                 basesrc gstbasesrc.c:3688:gst_base_src_start_complete:<filesrc0> pad not activated yet
Pipeline is PREROLLING ...
0:00:00.245038710   885 0xaaab1e3144c0 WARN                oggdemux gstoggdemux.c:3417:gst_ogg_demux_do_seek:<'':src_47b83c1f> Locking on pts 0:00:00.480362811
0:00:00.246162796   885 0xaaab1e3144c0 DEBUG                   alsa gstalsa.c:198:gst_alsa_detect_formats:<autoaudiosink0-actual-sink-alsa> skipping non-raw format or iec958 format
0:00:00.246275055   885 0xaaab1e3144c0 DEBUG                   alsa gstalsa.c:198:gst_alsa_detect_formats:<autoaudiosink0-actual-sink-alsa> skipping non-raw format or iec958 format
0:00:00.246317183   885 0xaaab1e3144c0 DEBUG                   alsa gstalsa.c:198:gst_alsa_detect_formats:<autoaudiosink0-actual-sink-alsa> skipping non-raw format or iec958 format
0:00:00.246358937   885 0xaaab1e3144c0 DEBUG                   alsa gstalsa.c:198:gst_alsa_detect_formats:<autoaudiosink0-actual-sink-alsa> skipping non-raw format or iec958 format
0:00:00.246417816   885 0xaaab1e3144c0 LOG                     alsa gstalsa.c:30:gst_alsa_detect_rates:<autoaudiosink0-actual-sink-alsa> probing sample rates ...
0:00:00.246481696   885 0xaaab1e3144c0 DEBUG                   alsa gstalsa.c:49:gst_alsa_detect_rates:<autoaudiosink0-actual-sink-alsa> Min. rate = 4000 (4000)
0:00:00.246528575   885 0xaaab1e3144c0 DEBUG                   alsa gstalsa.c:50:gst_alsa_detect_rates:<autoaudiosink0-actual-sink-alsa> Max. rate = 2147483647 (-1)
0:00:00.246574578   885 0xaaab1e3144c0 LOG                     alsa gstalsa.c:372:gst_alsa_detect_channels:<autoaudiosink0-actual-sink-alsa> probing channels ...
0:00:00.246624707   885 0xaaab1e3144c0 DEBUG                   alsa gstalsa.c:416:gst_alsa_detect_channels:<autoaudiosink0-actual-sink-alsa> Min. channels = 1 (1)
0:00:00.246670836   885 0xaaab1e3144c0 DEBUG                   alsa gstalsa.c:417:gst_alsa_detect_channels:<autoaudiosink0-actual-sink-alsa> Max. channels = 8 (10000)
0:00:00.247879554   885 0xaaab1e3144c0 DEBUG                   alsa gstalsa.c:498:gst_alsa_open_iec958_pcm:<autoaudiosink0-actual-sink-alsa> Generated device string "default"
0:00:00.257634559   885 0xaaab1e3144c0 LOG                     alsa gstalsasink.c:348:gst_alsasink_getcaps:<autoaudiosink0-actual-sink-alsa> Add pass-through capabilities
...


Also I can test output sound and it works, like:

root@imx8qxp-var-som:~# gst-launch-1.0 audiotestsrc ! autoaudiosink

So the issue on the source audio device

Also I can run another Linux image and micro also works

My system configuration:
Custom yocto distro Linux imx8qxp-var-som 5.15.60 Kirkstone 

Let me summarize the yocto configuration:

DISTRO_FEATURES_BACKFILL_CONSIDERED:remove = " pulseaudio"
DISTRO_FEATURES_BACKFILL:append = " pulseaudio"
DISTRO_FEATURES:append = " alsa"


Can someone help me to resolve the problem on audio source solution. It doesn't matter use pulsesrc or alsasrc.
Thanks in advance, Anton

0 Kudos
Reply
5 Replies

‎09-04-2023 08:02 PM
3,414 Views
joanxie
NXP TechSupport
NXP TechSupport

pls try this:

The pipeline described below shows the process to obtain this type of audio file and save it as an Ogg Vorbis file:
$ gst-launch-1.0 -v pulsesrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg

0 Kudos
Reply

‎09-05-2023 02:43 AM
3,405 Views
Anton_K
Contributor III

Hello, the result of command will be the same as on the first code block on my question before. The issue:

<autoaudiosink0-actual-sink-pulse> error: Failed to connect: Connection refused

 

I've made some progress and suppose that I have no configuration for micro.
Can you help me to configure micro input for the yocto build? Totally I have connected micro to the right channel on the LINEIN1_RP pin on the var-som-mx8x

0 Kudos
Reply

‎09-18-2023 12:42 AM
3,341 Views
joanxie
NXP TechSupport
NXP TechSupport

since you use the third party company bsp, suggest that you can compare your bsp with nxp bsp, which can support this already

0 Kudos
Reply

‎09-18-2023 01:21 AM
3,333 Views
Anton_K
Contributor III
Thank you for your time, will do. I just mean, have I done enough in distro recipes to include "pulseaudio" correctly related to this issue?
0 Kudos
Reply