Remove webrtc/stream.h and unutilized inheritance.

webrtc/video_send_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_SEND_STREAM_H_
 #define WEBRTC_VIDEO_SEND_STREAM_H_
 
 #include <map>
 #include <string>
 
 #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"
 #include "webrtc/media/base/videosinkinterface.h"
 
 namespace webrtc {
 
 class LoadObserver;
 class VideoEncoder;
 
 // Class to deliver captured frame to the video send stream.
 class VideoCaptureInput {
  public:
   // These methods do not lock internally and must be called sequentially.
   // If your application switches input sources synchronization must be done
   // externally to make sure that any old frames are not delivered concurrently.
   virtual void IncomingCapturedFrame(const VideoFrame& video_frame) = 0;
 
  protected:
   virtual ~VideoCaptureInput() {}
 };
 
-class VideoSendStream : public SendStream {
+class VideoSendStream {
  public:
   struct StreamStats {
     FrameCounts frame_counts;
     int width = 0;
     int height = 0;
     // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
     int total_bitrate_bps = 0;
     int retransmit_bitrate_bps = 0;
     int avg_delay_ms = 0;
     int max_delay_ms = 0;
@@ skipping 107 matching lines @@
     // 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;
 
     // True if the stream should be suspended when the available bitrate fall
     // below the minimum configured bitrate. If this variable is false, the
     // stream may send at a rate higher than the estimated available bitrate.
     bool suspend_below_min_bitrate = false;
   };
 
+  // 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;

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

No need to  have this in client interface, like for audio.

+
   // Gets interface used to insert captured frames. Valid as long as the
   // VideoSendStream is valid.
   virtual VideoCaptureInput* Input() = 0;
 
   // Set which streams to send. Must have at least as many SSRCs as configured
   // in the config. Encoder settings are passed on to the encoder instance along
   // with the VideoStream settings.
   virtual void ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
 
   virtual Stats GetStats() = 0;
+
+ protected:
+  virtual ~VideoSendStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_SEND_STREAM_H_