commit a45102f7b4a8df1e56d1c42a15d30576614a7551
author philipel <philipel@webrtc.org> Wed Feb 22 05:30:39 2017 -0800
committer Commit bot <commit-bot@chromium.org> Wed Feb 22 13:30:39 2017 +0000
tree c95b4ad9b1418ae976d93e49f43a600a5cc64e38
parent 5650a7d1c41a264a0da15c3bb17d2c14335fcc27

Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )

Reason for revert:
Fix here: https://codereview.chromium.org/2708593003

Original issue's description:
> 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}
> Committed: https://chromium.googlesource.com/external/webrtc/+/e525d6aba634b8442a0b9c247a89a6060d225ca3

TBR=nisse@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2704183002
Cr-Commit-Position: refs/heads/master@{#16772}

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 129a1b5..14f1265 100644
--- a/webrtc/modules/video_coding/video_receiver.cc
+++ b/webrtc/modules/video_coding/video_receiver.cc
@@ -56,31 +56,14 @@
 
 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) {
-      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);
-      }
+      _receiveStatsCallback->OnReceiveRatesUpdated(0, 0);
     }
   }
 
@@ -292,7 +275,7 @@
   return ret;
 }
 
-// Used for the WebRTC-NewVideoJitterBuffer experiment.
+// Used for the new jitter buffer.
 // TODO(philipel): Clean up among the Decode functions as we replace
 //                 VCMEncodedFrame with FrameObject.
 int32_t VideoReceiver::Decode(const webrtc::VCMEncodedFrame* frame) {