From: Sebastian Dröge <sebastian@centricular.com>
Date: Mon, 9 Mar 2015 15:00:07 +0000 (+0100)
Subject: rtsp-stream: Get the seqnum-base and other information from the last buffer in the... 
X-Git-Tag: 1.6.0~67
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=852cc09f542af5cadd79ffd7fe79d6475cf57e14;p=platform%2Fupstream%2Fgst-rtsp-server.git

rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink

This gives more accurate values than asking the payloader. There might be
queueing happening between the payloader and the sink.

https://bugzilla.gnome.org/show_bug.cgi?id=745704
---

diff --git a/gst/rtsp-server/rtsp-stream.c b/gst/rtsp-server/rtsp-stream.c
index abdeeaa..873ca5d 100644
--- a/gst/rtsp-server/rtsp-stream.c
+++ b/gst/rtsp-server/rtsp-stream.c
@@ -55,6 +55,8 @@
 #include <gst/app/gstappsrc.h>
 #include <gst/app/gstappsink.h>
 
+#include <gst/rtp/gstrtpbuffer.h>
+
 #include "rtsp-stream.h"
 
 #define GST_RTSP_STREAM_GET_PRIVATE(obj)  \
@@ -2457,6 +2459,60 @@ gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
 
   g_mutex_lock (&priv->lock);
 
+  /* First try to extract the information from the last buffer on the sinks.
+   * This will have a more accurate sequence number and timestamp, as between
+   * the payloader and the sink there can be some queues
+   */
+  if (priv->udpsink[0] || priv->appsink[0]) {
+    GstSample *last_sample;
+
+    if (priv->udpsink[0])
+      g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
+    else
+      g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
+
+    if (last_sample) {
+      GstCaps *caps;
+      GstBuffer *buffer;
+      GstSegment *segment;
+      GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
+
+      caps = gst_sample_get_caps (last_sample);
+      buffer = gst_sample_get_buffer (last_sample);
+      segment = gst_sample_get_segment (last_sample);
+
+      if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
+        if (seq) {
+          *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
+          gst_rtp_buffer_unmap (&rtp_buffer);
+        }
+
+        if (rtptime) {
+          *rtptime = GST_BUFFER_TIMESTAMP (buffer);
+        }
+
+        if (running_time) {
+          *running_time =
+              gst_segment_to_running_time (segment, GST_FORMAT_TIME,
+              GST_BUFFER_TIMESTAMP (buffer));
+        }
+
+        if (clock_rate) {
+          GstStructure *s = gst_caps_get_structure (caps, 0);
+
+          gst_structure_get_uint (s, "clock-rate", clock_rate);
+          if (*clock_rate == 0 && running_time)
+            *running_time = GST_CLOCK_TIME_NONE;
+        }
+        gst_sample_unref (last_sample);
+
+        goto done;
+      } else {
+        gst_sample_unref (last_sample);
+      }
+    }
+  }
+
   if (g_object_class_find_property (payobjclass, "stats")) {
     g_object_get (priv->payloader, "stats", &stats, NULL);
     if (stats == NULL)
@@ -2491,6 +2547,8 @@ gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
     if (running_time)
       *running_time = GST_CLOCK_TIME_NONE;
   }
+
+done:
   g_mutex_unlock (&priv->lock);
 
   return TRUE;