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

webrtc/modules/video_coding/frame_object.h

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 19 matching lines @@
 
   // The capture timestamp of this frame.
   virtual uint32_t Timestamp() const = 0;
 
   // When this frame was received.
   virtual int64_t ReceivedTime() const = 0;
 
   // 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
   // object. For codec types that don't necessarily have picture ids they
   // have to be constructed from the header data relevant to that codec.
   uint16_t picture_id;
   uint8_t spatial_layer;
   uint32_t timestamp;
 
   size_t num_references;
   uint16_t references[kMaxFrameReferences];
   bool inter_layer_predicted;
 };
 
 class PacketBuffer;
 
 class RtpFrameObject : public FrameObject {
  public:
   RtpFrameObject(PacketBuffer* packet_buffer,
                  uint16_t first_seq_num,
                  uint16_t last_seq_num,
                  size_t frame_size,
                  int times_nacked,
                  int64_t received_time);
 
   ~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;
   uint32_t Timestamp() const override;
   int64_t ReceivedTime() const override;
   int64_t RenderTime() const override;
   rtc::Optional<RTPVideoTypeHeader> GetCodecHeader() const;
 
  private:
   rtc::scoped_refptr<PacketBuffer> packet_buffer_;
   enum FrameType frame_type_;
   VideoCodecType codec_type_;
   uint16_t first_seq_num_;
   uint16_t last_seq_num_;
   uint32_t timestamp_;
   int64_t received_time_;
 
   // Equal to times nacked of the packet with the highet times nacked
   // belonging to this frame.
   int times_nacked_;
 };
 
 }  // namespace video_coding
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_FRAME_OBJECT_H_