From 065e819f18cf272019fe4bd52c4d5c81d71f262f Mon Sep 17 00:00:00 2001 From: Dmitry Sharshakov Date: Sat, 16 Dec 2023 21:29:19 +0300 Subject: [PATCH] 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 --- src/modules/module-rtp/stream.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/modules/module-rtp/stream.c b/src/modules/module-rtp/stream.c index e767c8ecc..40daaf2ce 100644 --- a/src/modules/module-rtp/stream.c +++ b/src/modules/module-rtp/stream.c @@ -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 {