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rtpsession: Only add send latency to the running time if it is actually known

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Otherwise we can't know the running time yet if rtcp-sync-send-time is
set, and have to wait until the latency is known later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
This commit is contained in:
Sebastian Dröge 2022-04-09 11:00:52 +03:00 committed by GStreamer Marge Bot
parent 7ffc830959
commit 7c796b3c05
1 changed files with 10 additions and 2 deletions
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10
subprojects/gst-plugins-good/gst/rtpmanager/gstrtpsession.c
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@ -1309,6 +1309,7 @@ gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_READY_TO_PAUSED: case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_RTP_SESSION_LOCK (rtpsession); GST_RTP_SESSION_LOCK (rtpsession);
rtpsession->priv->wait_send = TRUE; rtpsession->priv->wait_send = TRUE;
rtpsession->priv->send_latency = GST_CLOCK_TIME_NONE;
GST_RTP_SESSION_UNLOCK (rtpsession); GST_RTP_SESSION_UNLOCK (rtpsession);
break; break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING: case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
@ -2429,8 +2430,15 @@ gst_rtp_session_chain_send_rtp_common (GstRtpSession * rtpsession,
running_time = running_time =
gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME, gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME,
timestamp); timestamp);
if (priv->rtcp_sync_send_time) if (priv->rtcp_sync_send_time) {
if (priv->send_latency != GST_CLOCK_TIME_NONE) {
running_time += priv->send_latency; running_time += priv->send_latency;
} else {
GST_WARNING_OBJECT (rtpsession,
"Can't determine running time for this packet without knowing configured latency");
running_time = -1;
}
}
} else { } else {
/* no timestamp. */ /* no timestamp. */
running_time = -1; running_time = -1;