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rtpsession: Only add send latency to the running time if it is actually known
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>
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7ffc830959
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7c796b3c05
1 changed files with 10 additions and 2 deletions
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@ -1309,6 +1309,7 @@ gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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GST_RTP_SESSION_LOCK (rtpsession);
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rtpsession->priv->wait_send = TRUE;
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rtpsession->priv->send_latency = GST_CLOCK_TIME_NONE;
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GST_RTP_SESSION_UNLOCK (rtpsession);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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@ -2429,8 +2430,15 @@ gst_rtp_session_chain_send_rtp_common (GstRtpSession * rtpsession,
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running_time =
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gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME,
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timestamp);
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if (priv->rtcp_sync_send_time)
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running_time += priv->send_latency;
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if (priv->rtcp_sync_send_time) {
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if (priv->send_latency != GST_CLOCK_TIME_NONE) {
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running_time += priv->send_latency;
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} else {
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GST_WARNING_OBJECT (rtpsession,
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"Can't determine running time for this packet without knowing configured latency");
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running_time = -1;
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}
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}
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} else {
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/* no timestamp. */
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running_time = -1;
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