mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-18 20:25:25 +00:00
The original BUNDLE support commit placed a queue after the rtpfunnel that combines streams, but I don't see a good reason for it. It has default settings, so if network output is slow might accidentally store up to 1 second of pending data, increasing latency. Remove it in favour of doing any necessary buffering before webrtcbin. If it turns out that there is a reason for it to exist, the limits should probably be configurable and small. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3437> |
||
---|---|---|
.. | ||
fwd.h | ||
gstwebrtc.c | ||
gstwebrtcbin.c | ||
gstwebrtcbin.h | ||
gstwebrtcstats.c | ||
gstwebrtcstats.h | ||
meson.build | ||
transportreceivebin.c | ||
transportreceivebin.h | ||
transportsendbin.c | ||
transportsendbin.h | ||
transportstream.c | ||
transportstream.h | ||
utils.c | ||
utils.h | ||
webrtcdatachannel.c | ||
webrtcdatachannel.h | ||
webrtcsctptransport.c | ||
webrtcsctptransport.h | ||
webrtcsdp.c | ||
webrtcsdp.h | ||
webrtctransceiver.c | ||
webrtctransceiver.h |