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author | Wim Taymans <wim.taymans@gmail.com> | 2007-08-29 16:56:27 +0000 |
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committer | Wim Taymans <wim.taymans@gmail.com> | 2007-08-29 16:56:27 +0000 |
commit | 696bf742122af762be421aa1c3638b8a265d8fd8 (patch) | |
tree | abbaa5f8b77889235b11b6d592a8fed6419307ec /gst/rtpmanager/gstrtpjitterbuffer.c | |
parent | 43ead5a174f0485578128afc7ad87de0deda92f9 (diff) | |
download | gst-plugins-bad-696bf742122af762be421aa1c3638b8a265d8fd8.tar.gz gst-plugins-bad-696bf742122af762be421aa1c3638b8a265d8fd8.tar.bz2 gst-plugins-bad-696bf742122af762be421aa1c3638b8a265d8fd8.zip |
gst/rtpmanager/gstrtpjitterbuffer.c: Improve Comments.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Improve Comments.
* gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
(gst_rtp_session_change_state), (gst_rtp_session_parse_caps),
(gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps),
(gst_rtp_session_event_send_rtp_sink), (create_recv_rtp_sink),
(create_send_rtp_sink):
Also parse the sink caps for clock-rate instead of only relying on the
result of the signal.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Make sure we fetch the clock rate for payloads we are sending out so
that we can use it for SR reports.
Diffstat (limited to 'gst/rtpmanager/gstrtpjitterbuffer.c')
-rw-r--r-- | gst/rtpmanager/gstrtpjitterbuffer.c | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c index 2e44425a..f4d59bb6 100644 --- a/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/gst/rtpmanager/gstrtpjitterbuffer.c @@ -942,6 +942,7 @@ again: if (priv->eos) goto do_eos; } + /* wait for packets or flushing now */ JBUF_WAIT_CHECK (priv, flushing); } @@ -1004,7 +1005,6 @@ again: running_time = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, timestamp); - /* correct for sync against the gstreamer clock, add latency */ GST_OBJECT_LOCK (jitterbuffer); clock = GST_ELEMENT_CLOCK (jitterbuffer); if (!clock) { @@ -1013,7 +1013,7 @@ again: goto push_buffer; } - /* add latency */ + /* add latency, this includes our own latency and the peer latency. */ running_time += (priv->latency_ms * GST_MSECOND); running_time += priv->peer_latency; @@ -1050,7 +1050,7 @@ again: if (ret == GST_CLOCK_UNSCHEDULED) { GST_DEBUG_OBJECT (jitterbuffer, "Wait got unscheduled, will retry to push with new buffer"); - /* reinserting popped buffer into queue */ + /* reinsert popped buffer into queue */ if (!rtp_jitter_buffer_insert (priv->jbuf, outbuf)) { GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping", seqnum); |