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rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer

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This commit is contained in:
Sebastian Dröge 2015-01-16 20:48:42 +01:00
parent ccf6c6eb53
commit 844add610d
1 changed files with 52 additions and 0 deletions
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52
gst/rtsp-server/rtsp-stream.c
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@ -109,6 +109,7 @@ struct _GstRTSPStreamPrivate
/* for TCP transport */
GstElement *appsrc[2];
GstClockTime appsrc_base_time[2];
GstElement *appqueue[2];
GstElement *appsink[2];
@ -2193,6 +2194,7 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
/* make and add appsrc */
priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
priv->appsrc_base_time[i] = -1;
g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, NULL);
gst_bin_add (bin, priv->appsrc[i]);
/* and link to the funnel */
@ -2544,6 +2546,31 @@ gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
g_mutex_unlock (&priv->lock);
if (element) {
if (priv->appsrc_base_time[0] == -1) {
/* Take current running_time. This timestamp will be put on
* the first buffer of each stream because we are a live source and so we
* timestamp with the running_time. When we are dealing with TCP, we also
* only timestamp the first buffer (using the DISCONT flag) because a server
* typically bursts data, for which we don't want to compensate by speeding
* up the media. The other timestamps will be interpollated from this one
* using the RTP timestamps. */
GST_OBJECT_LOCK (element);
if (GST_ELEMENT_CLOCK (element)) {
GstClockTime now;
GstClockTime base_time;
now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
base_time = GST_ELEMENT_CAST (element)->base_time;
priv->appsrc_base_time[0] = now - base_time;
GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
GST_TIME_ARGS (base_time));
}
GST_OBJECT_UNLOCK (element);
}
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
gst_object_unref (element);
} else {
@ -2587,6 +2614,31 @@ gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
g_mutex_unlock (&priv->lock);
if (element) {
if (priv->appsrc_base_time[1] == -1) {
/* Take current running_time. This timestamp will be put on
* the first buffer of each stream because we are a live source and so we
* timestamp with the running_time. When we are dealing with TCP, we also
* only timestamp the first buffer (using the DISCONT flag) because a server
* typically bursts data, for which we don't want to compensate by speeding
* up the media. The other timestamps will be interpollated from this one
* using the RTP timestamps. */
GST_OBJECT_LOCK (element);
if (GST_ELEMENT_CLOCK (element)) {
GstClockTime now;
GstClockTime base_time;
now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
base_time = GST_ELEMENT_CAST (element)->base_time;
priv->appsrc_base_time[1] = now - base_time;
GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
GST_TIME_ARGS (base_time));
}
GST_OBJECT_UNLOCK (element);
}
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
gst_object_unref (element);
} else {