Remove webrtc/stream.h and unutilized inheritance.

webrtc/video_receive_stream.h

 /*
  *  Copyright (c) 2013 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_VIDEO_RECEIVE_STREAM_H_
 #define WEBRTC_VIDEO_RECEIVE_STREAM_H_
 
 #include <limits>
 #include <map>
 #include <string>
 #include <vector>
 
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/videosinkinterface.h"
-#include "webrtc/stream.h"
 #include "webrtc/transport.h"
 
 namespace webrtc {
 
 class VideoDecoder;
 
-class VideoReceiveStream : public ReceiveStream {
+class VideoReceiveStream {
  public:
   // TODO(mflodman) Move all these settings to VideoDecoder and move the
   // declaration to common_types.h.
   struct Decoder {
     std::string ToString() const;
 
     // The actual decoder instance.
     VideoDecoder* decoder = nullptr;
 
     // Received RTP packets with this payload type will be sent to this decoder
@@ skipping 128 matching lines @@
     // Called for each decoded frame. E.g. used when adding effects to the
     // decoded
     // stream. 'nullptr' disables the callback.
     I420FrameCallback* pre_render_callback = nullptr;
 
     // Target delay in milliseconds. A positive value indicates this stream is
     // used for streaming instead of a real-time call.
     int target_delay_ms = 0;
   };
 
+  // Starts stream activity.
+  // When a stream is active, it can receive, process and deliver packets.
+  virtual void Start() = 0;
+  // Stops stream activity.
+  // When a stream is stopped, it can't receive, process or deliver packets.
+  virtual void Stop() = 0;
+  // Deliver an incoming RTCP packet.
+  virtual bool DeliverRtcp(const uint8_t* packet, size_t length) = 0;
+  // Deliver an incoming RTP packet.

[The Sun (google.com), 2016/04/27 19:39:46]

No need for these in the client interface, like for audio.

+  virtual bool DeliverRtp(const uint8_t* packet,
+                          size_t length,
+                          const PacketTime& packet_time) = 0;
+
   // TODO(pbos): Add info on currently-received codec to Stats.
   virtual Stats GetStats() const = 0;
+
+ protected:
+  virtual ~VideoReceiveStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RECEIVE_STREAM_H_