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examples/webrtc: Configure payload types

Browse source 
MR 2398 broke the webrtc sendrecv example
by not configuring the payload types, so both audio and video streams
get sent on payload 96.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3434>
This commit is contained in:
Jan Schmidt 2022-11-19 04:37:00 +11:00
parent f6327e25a7
commit f2ae481a69
1 changed files with 2 additions and 2 deletions
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4
subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
View file

@ -442,7 +442,7 @@ start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
audio_desc =
g_strdup_printf
("audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample"
"! queue ! opusenc ! rtpopuspay name=audiopay ! queue");
"! queue ! opusenc ! rtpopuspay name=audiopay pt=%u ! queue", opus_pt);
audio_bin = gst_parse_bin_from_description (audio_desc, TRUE, &audio_error);
g_free (audio_desc);
if (audio_error) {
@ -461,7 +461,7 @@ start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
"vp8enc deadline=1 keyframe-max-dist=2000 ! "
/* picture-id-mode=15-bit seems to make TWCC stats behave better, and
* fixes stuttery video playback in Chrome */
"rtpvp8pay name=videopay picture-id-mode=15-bit ! queue");
"rtpvp8pay name=videopay picture-id-mode=15-bit pt=%u ! queue", vp8_pt);
video_bin = gst_parse_bin_from_description (video_desc, TRUE, &video_error);
g_free (video_desc);
if (video_error) {