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rtpgstpay: Delay pushing of event packets until the next buffer
And also re-timestamp them with the current buffer's PTS. Not doing so keeps the timestamps of event packets as GST_CLOCK_TIME_NONE or the timestamp of the previous buffer, both of which are bogus. Making sure that (especially) the first packet has a valid timestamp allows putting e.g. the NTP timestamp RTP header extension on it. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
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37578454b9
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2 changed files with 21 additions and 17 deletions
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@ -182,7 +182,6 @@ gst_rtp_gst_pay_reset (GstRtpGSTPay * rtpgstpay, gboolean full)
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rtpgstpay->current_CV = 0;
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rtpgstpay->current_CV = 0;
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rtpgstpay->next_CV = 0;
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rtpgstpay->next_CV = 0;
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}
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}
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rtpgstpay->received_buffer = FALSE;
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}
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}
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static void
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static void
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@ -366,6 +365,16 @@ gst_rtp_gst_pay_create_from_adapter (GstRtpGSTPay * rtpgstpay,
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return TRUE;
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return TRUE;
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}
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}
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static gboolean
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retimestamp_buffer (GstBuffer ** buffer, guint idx, gpointer user_data)
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{
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GstClockTime *timestamp = user_data;
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GST_BUFFER_PTS (*buffer) = *timestamp;
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return TRUE;
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}
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static GstFlowReturn
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static GstFlowReturn
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gst_rtp_gst_pay_flush (GstRtpGSTPay * rtpgstpay, GstClockTime timestamp)
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gst_rtp_gst_pay_flush (GstRtpGSTPay * rtpgstpay, GstClockTime timestamp)
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{
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{
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@ -373,13 +382,14 @@ gst_rtp_gst_pay_flush (GstRtpGSTPay * rtpgstpay, GstClockTime timestamp)
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gst_rtp_gst_pay_create_from_adapter (rtpgstpay, timestamp);
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gst_rtp_gst_pay_create_from_adapter (rtpgstpay, timestamp);
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if (!rtpgstpay->received_buffer) {
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GST_DEBUG_OBJECT (rtpgstpay,
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"Can't flush without having received a buffer yet");
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return GST_FLOW_OK;
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}
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if (rtpgstpay->pending_buffers) {
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if (rtpgstpay->pending_buffers) {
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// make sure all buffers in the buffer list have the correct timestamp.
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// If we created packets based on an event they would have
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// GST_CLOCK_TIME_NONE as PTS.
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gst_buffer_list_foreach (rtpgstpay->pending_buffers, retimestamp_buffer,
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×tamp);
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/* push the whole buffer list at once */
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/* push the whole buffer list at once */
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ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpgstpay),
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ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpgstpay),
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rtpgstpay->pending_buffers);
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rtpgstpay->pending_buffers);
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@ -584,12 +594,10 @@ gst_rtp_gst_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
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GST_DEBUG_OBJECT (rtpgstpay, "make event type %d for %s",
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GST_DEBUG_OBJECT (rtpgstpay, "make event type %d for %s",
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etype, GST_EVENT_TYPE_NAME (event));
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etype, GST_EVENT_TYPE_NAME (event));
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gst_rtp_gst_pay_send_event (rtpgstpay, etype, event);
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gst_rtp_gst_pay_send_event (rtpgstpay, etype, event);
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/* Do not send stream-start right away since caps/new-segment were not yet
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// do not flush events here yet as they would get no timestamp at all or
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sent, so our data would be considered invalid */
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// the timestamp of the previous buffer, both of which are bogus. We need
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if (etype != 4) {
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// to wait until the next actual input frame to know the timestamp that
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/* flush the adapter immediately */
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// applies to the event.
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gst_rtp_gst_pay_flush (rtpgstpay, GST_CLOCK_TIME_NONE);
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}
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}
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}
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gst_event_unref (event);
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gst_event_unref (event);
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@ -654,8 +662,6 @@ gst_rtp_gst_pay_handle_buffer (GstRTPBasePayload * basepayload,
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rtpgstpay = GST_RTP_GST_PAY (basepayload);
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rtpgstpay = GST_RTP_GST_PAY (basepayload);
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rtpgstpay->received_buffer = TRUE;
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timestamp = GST_BUFFER_PTS (buffer);
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timestamp = GST_BUFFER_PTS (buffer);
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running_time =
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running_time =
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gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME,
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gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME,
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@ -57,8 +57,6 @@ struct _GstRtpGSTPay
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guint config_interval;
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guint config_interval;
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GstClockTime last_config;
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GstClockTime last_config;
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gboolean force_config;
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gboolean force_config;
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gboolean received_buffer;
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};
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};
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struct _GstRtpGSTPayClass
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struct _GstRtpGSTPayClass
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