commit e525d6aba634b8442a0b9c247a89a6060d225ca3
author stefan <stefan@webrtc.org> Wed Feb 08 05:25:42 2017 -0800
committer Commit bot <commit-bot@chromium.org> Wed Feb 08 13:25:42 2017 +0000
tree 9a69a393b95add39d977722c45c592eb9aac34d5
parent 498ee8e816a9f900f0904319d899cecde8d62b14

Revert Make the new jitter buffer the default jitter buffer.

Speculative revert of https://codereview.chromium.org/2656983002/ to see if it fixes a downstream bug.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2682073003
Cr-Commit-Position: refs/heads/master@{#16492}

webrtc/modules/video_coding/video_receiver.cc

diff --git a/webrtc/modules/video_coding/video_receiver.cc b/webrtc/modules/video_coding/video_receiver.cc
index 14f1265..129a1b5 100644
--- a/webrtc/modules/video_coding/video_receiver.cc
+++ b/webrtc/modules/video_coding/video_receiver.cc
@@ -56,14 +56,31 @@
 
 void VideoReceiver::Process() {
   // Receive-side statistics
-
-  // TODO(philipel): Remove this if block when we know what to do with
-  //                 ReceiveStatisticsProxy::QualitySample.
   if (_receiveStatsTimer.TimeUntilProcess() == 0) {
     _receiveStatsTimer.Processed();
     rtc::CritScope cs(&process_crit_);
     if (_receiveStatsCallback != nullptr) {
-      _receiveStatsCallback->OnReceiveRatesUpdated(0, 0);
+      uint32_t bitRate;
+      uint32_t frameRate;
+      _receiver.ReceiveStatistics(&bitRate, &frameRate);
+      _receiveStatsCallback->OnReceiveRatesUpdated(bitRate, frameRate);
+    }
+
+    if (_decoderTimingCallback != nullptr) {
+      int decode_ms;
+      int max_decode_ms;
+      int current_delay_ms;
+      int target_delay_ms;
+      int jitter_buffer_ms;
+      int min_playout_delay_ms;
+      int render_delay_ms;
+      if (_timing->GetTimings(&decode_ms, &max_decode_ms, &current_delay_ms,
+                              &target_delay_ms, &jitter_buffer_ms,
+                              &min_playout_delay_ms, &render_delay_ms)) {
+        _decoderTimingCallback->OnDecoderTiming(
+            decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
+            jitter_buffer_ms, min_playout_delay_ms, render_delay_ms);
+      }
     }
   }
 
@@ -275,7 +292,7 @@
   return ret;
 }
 
-// Used for the new jitter buffer.
+// Used for the WebRTC-NewVideoJitterBuffer experiment.
 // TODO(philipel): Clean up among the Decode functions as we replace
 //                 VCMEncodedFrame with FrameObject.
 int32_t VideoReceiver::Decode(const webrtc::VCMEncodedFrame* frame) {