Audio Pipe with using the Sink Clock

[Maik Scholz]

Hi,
I like to build an audio pipeline with several internal app sources and a
single alsa sink.
The pipeline shall be clocked from the sink, with using the
alsasink.slave_mothod=2.
Unfortunately, the AudioClock is not selected by the pipeline.
Instead, the SystemClock is used.

How can I force the pipeline using the clock from the alsasink?

GStreamer Version: 1.10.5

>export GST_DEBUG=2,audioclock:5,pipeline:5,GST_CLOCK:3
>gst-launch-1.0 audiotestsrc freq=330 is-live=true do-timestamp=true !
alsasink provide-clock=true sync=true slave-method=2 can-activate-pull=true
async=true

0:00:00.008763468  7290      0x1832780 DEBUG             audioclock
gstaudioclock.c:69:gst_audio_clock_init:<GstAudioClock at 0x18440d0> init
0:00:00.008899430  7290      0x1832780 DEBUG               pipeline
gstpipeline.c:236:gst_pipeline_init:<GstPipeline at 0x18461d0> set bus <bus1>
on pipeline
Setting pipeline to PAUSED ...
0:00:00.009070571  7290      0x1832780 DEBUG             audioclock
gstaudioclock.c:141:gst_audio_clock_reset:<GstAudioSinkClock> reset clock to
0:00:00.000000000, last 0:00:00.000000000, offset +0:00:00.000000000
0:00:00.017725425  7290      0x1832780 DEBUG               pipeline
gstpipeline.c:305:reset_start_time:<pipeline0> reset start_time to 0
Pipeline is live and does not need PREROLL ...
0:00:00.018167469  7290      0x17d2630 WARN                    alsa
conf.c:4694:snd_config_expand: alsalib error: Unknown parameters {AES0 0x02
AES1 0x82 AES2 0x00 AES3 0x02}
0:00:00.018216937  7290      0x17d2630 WARN                    alsa
pcm.c:2239:snd_pcm_open_noupdate: alsalib error: Unknown PCM default:{AES0
0x02 AES1 0x82 AES2 0x00 AES3 0x02}
Setting pipeline to PLAYING ...
0:00:00.018238070  7290      0x1832780 DEBUG               pipeline
gstpipeline.c:403:gst_pipeline_change_state:<pipeline0> selecting clock and
base_time
0:00:00.018244559  7290      0x1832780 DEBUG               pipeline
gstpipeline.c:424:gst_pipeline_change_state:<pipeline0> Need to update
start_time
0:00:00.018247938  7290      0x1832780 DEBUG               pipeline
gstpipeline.c:429:gst_pipeline_change_state:<pipeline0> Need to update
clock.
0:00:00.018288847  7290      0x1832780 DEBUG               pipeline
gstpipeline.c:469:gst_pipeline_change_state:<pipeline0>
start_time=0:00:00.000000000, now=20:55:58.945104958, base_time
20:55:58.945104958
New clock: GstSystemClock
0:00:00.018347844  7290      0x17d2630 DEBUG             audioclock
gstaudioclock.c:172:gst_audio_clock_get_internal_time:<GstAudioSinkClock>
result 0:00:00.000000000, last_time 0:00:00.000000000
0:00:00.018359345  7290      0x17d2630 DEBUG             audioclock
gstaudioclock.c:172:gst_audio_clock_get_internal_time:<GstAudioSinkClock>
result 0:00:00.000000000, last_time 0:00:00.000000000
0:00:00.022769254  7290      0x17d2630 DEBUG             audioclock
gstaudioclock.c:141:gst_audio_clock_reset:<GstAudioSinkClock> reset clock to
0:00:00.000000000, last 0:00:00.000000000, offset +0:00:00.000000000
Redistribute latency...
0:00:00.023340481  7290 0x7fc7f8003320 DEBUG               pipeline
gstpipeline.c:403:gst_pipeline_change_state:<pipeline0> selecting clock and
base_time
0:00:00.023351966  7290 0x7fc7f8003320 DEBUG               pipeline
gstpipeline.c:478:gst_pipeline_change_state:<pipeline0> NOT adjusting
base_time because we selected one before
0:00:00.042286706  7290      0x17d2630 DEBUG             audioclock
gstaudioclock.c:202:gst_audio_clock_get_time:<GstAudioSinkClock> result
0:00:00.000000000, last_time 0:00:00.000000000

... from here, "gst_audio_clock" is not called anymore.






--
Sent from: http://gstreamer-devel.966125.n4.nabble.com/