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b7820a0de7
When using the following setup (the error can be reproduced using simpler sender pipelines), the receiver resynchronises the clock on RTCP packets. The effect was that a couple seconds were cut out of the playback because an initial RTCP packet was dropped. When sending out all RTCP packets (setting sync=FALSE on the RTCP updsink), the playback is fine. This syncs rtpsink with rtpsrc (where this property was already set). gst-launch-1.0 filesrc location=899-en.mp3 \ ! mpegaudioparse \ ! mpg123audiodec \ ! audioconvert \ ! audioresample \ ! avenc_g722 \ ! rtpg722pay ! rtpsink uri=rtp://239.1.2.3:1234 gst-launch-1.0 uridecodebin rtp://239.1.2.3:1234?encoding-name=G722 \ ! autoaudiosink Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/993> |
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gstrtp-utils.c | ||
gstrtp-utils.h | ||
gstrtpsink.c | ||
gstrtpsink.h | ||
gstrtpsrc.c | ||
gstrtpsrc.h | ||
meson.build | ||
plugin.c |