Delete method FrameObject::GetBitstream.

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.
  */
 
 #ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_OBJECT_H_
 #define WEBRTC_MODULES_VIDEO_CODING_FRAME_OBJECT_H_
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/encoded_frame.h"
 
 namespace webrtc {
 namespace video_coding {
 
 class FrameObject : public webrtc::VCMEncodedFrame {
  public:
   static const uint8_t kMaxFrameReferences = 5;
 
   FrameObject();
   virtual ~FrameObject() {}
 
-  virtual bool GetBitstream(uint8_t* destination) const = 0;
-
   // 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;
 
+  size_t size() { return _size; }
 
   // 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;
-
-  size_t size;
 };
 
 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;
   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;
   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_