Don't update the jitter estimate with frames containing retransmitted packets.

webrtc/modules/video_coding/frame_object.h

Index: webrtc/modules/video_coding/frame_object.h
diff --git a/webrtc/modules/video_coding/frame_object.h b/webrtc/modules/video_coding/frame_object.h
index 915224ede8de70bdd45ec7c27962e8d8a58ae4e1..3fd5713e23fdca4c2eda87abf3beb002d96c17a0 100644
--- a/webrtc/modules/video_coding/frame_object.h
+++ b/webrtc/modules/video_coding/frame_object.h
@@ -37,6 +37,12 @@ class FrameObject : public webrtc::VCMEncodedFrame {
   // When this frame should be rendered.
   virtual int64_t RenderTime() const = 0;
 
+  // This is a RtpFrameObject specific function that is needed by the current
+  // timing calculation class.
+  // TODO(philipel): Remove this function when new a new timing class has

[stefan-webrtc, 2017/01/24 13:03:14]

when a new

[philipel, 2017/01/25 15:04:17]

Done.

+  //                 been implemented.
+  virtual int times_nacked() const { return 0; }

[stefan-webrtc, 2017/01/24 13:03:14]

Should we call it "delayed_by_retransmission()" instead and make it a bool?

[philipel, 2017/01/25 15:04:16]

Done.

+
   size_t size() { return _length; }
 
   // The tuple (|picture_id|, |spatial_layer|) uniquely identifies a frame
@@ -65,7 +71,7 @@ class RtpFrameObject : public FrameObject {
   ~RtpFrameObject();
   uint16_t first_seq_num() const;
   uint16_t last_seq_num() const;
-  int times_nacked() const;
+  int times_nacked() const override;
   enum FrameType frame_type() const;
   VideoCodecType codec_type() const;
   bool GetBitstream(uint8_t* destination) const override;