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TODO: module-rtp: buffering for sender 

This should be done to match packet size requirements (e.g. 1 ms) while allowing user's software to run at higher buffer size to not stutter.

This will require scheduling multiple rtp_audio_flush_packets calls per one rtp_audio_process_capture call
This commit is contained in:
Dmitry Sharshakov 2023-12-16 21:29:19 +03:00 committed by Wim Taymans
parent 066e8e1c63
commit 065e819f18
1 changed files with 1 additions and 0 deletions
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src/modules/module-rtp/stream.c
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@ -452,6 +452,7 @@ struct rtp_stream *rtp_stream_new(struct pw_core *core,
pw_properties_setf(props, PW_KEY_NODE_RATE, "1/%d", impl->rate);
if (direction == PW_DIRECTION_INPUT) {
// TODO: make sess.latency.msec work for sender streams
pw_properties_setf(props, PW_KEY_NODE_LATENCY, "%d/%d",
impl->psamples, impl->rate);
} else {