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gstpipewiresrc: Fix re-linking for audio

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For a pipeline like below, we might want to dynamically switch the audio
source.

gst-launch-1.0 -e pipewiresrc autoconnect=false ! queue ! audioconvert ! autoaudiosink

On switching to a different audio source, any one of driver, quantum
or clock rate might change which changes the return `result` value of
gst_pipewire_clock_get_internal_time.

This can result in the basesrc create function incorrectly waiting in
gst_clock_id_wait. We post clock lost message to fix this. In the case
of gst-launch, it will set the pipeline to PAUSED and then PLAYING to
to force a new clock and a new base_time distribution.

Without the clock lost message, the following can be seen

before re-linking to a different source
0:00:30.887602864 79499 0x7fffe8000d40 DEBUG              GST_CLOCK gstsystemclock.c:1158:gst_system_clock_id_wait_jitter_unlocked:<pipewireclock0> entry 0x7fffd803fad0 time 0:00:17.024565416 now 0:00:17.024109144 diff (time-now) 456272

after re-linking to a different source
0:00:45.790843245 79499 0x7fffe8000d40 DEBUG              GST_CLOCK gstsystemclock.c:1158:gst_system_clock_id_wait_jitter_unlocked:<pipewireclock0> entry 0x7fffd803fad0 time 0:00:31.927694059 now 0:00:17.066883864 diff (time-now) 14860810195

With the clock lost message, the following can be seen

before re-linking to a different source
0:01:09.336533552 89461 0x7fffe8000d40 DEBUG              GST_CLOCK gstsystemclock.c:1158:gst_system_clock_id_wait_jitter_unlocked:<pipewireclock0> entry 0x7fffd803fad0 time 0:00:58.198536772 now 0:00:58.197444926 diff (time-now) 1091846

after re-linking to a different source
0:01:21.659827958 89461 0x7fffe8000d40 DEBUG              GST_CLOCK gstsystemclock.c:1158:gst_system_clock_id_wait_jitter_unlocked:<pipewireclock0> entry 0x7fffd803fad0 time 0:28:24.853517646 now 0:28:24.853527204 diff (time-now) -9558

Note the difference in `time` and `now` fields of the above log message.

This is easy to reproduce by using a pipewiresink as the audio source
with a pipeline like below, as one of the sources during switching.

gst-launch-1.0 -e audiotestsrc wave=ticks ! audioconvert ! audio/x-raw,format=F32LE,rate=48000,channels=1 !
pipewiresink stream-properties="props,media.class=Audio/Source,node.description=pwsink" client-name=pwsink

Applications need to handle the GST_MESSAGE_CLOCK_LOST message in their
bus handlers.
This commit is contained in:
Sanchayan Maity 2025-04-01 17:34:15 +05:30
parent dd683176c2
commit 77143e54d9
1 changed files with 16 additions and 0 deletions
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16
src/gst/gstpipewiresrc.c
View file

@ -739,13 +739,29 @@ on_state_changed (void *data,
enum pw_stream_state state, const char *error)
{
GstPipeWireSrc *pwsrc = data;
GstState current_state = GST_ELEMENT_CAST (pwsrc)->current_state;
GST_DEBUG ("got stream state %s", pw_stream_state_as_string (state));
switch (state) {
case PW_STREAM_STATE_UNCONNECTED:
case PW_STREAM_STATE_CONNECTING:
break;
case PW_STREAM_STATE_PAUSED:
/*
* We may see a driver/quantum/clock rate change on switching audio
* sources. The same is not applicable for video.
*
* We post the clock lost message here to take care of a possible
* jump or shift in base_time/clock for the pipeline. Application
* must handle the clock lost message in it's bus handler by pausing
* the pipeline and then setting it back to playing.
*/
if (current_state == GST_STATE_PLAYING && !pwsrc->is_video)
gst_element_post_message (GST_ELEMENT_CAST (pwsrc),
gst_message_new_clock_lost (GST_OBJECT_CAST (pwsrc),
GST_CLOCK_CAST (pwsrc->stream->clock)));
break;
case PW_STREAM_STATE_STREAMING:
break;
case PW_STREAM_STATE_ERROR: