Remove webrtc/stream.h and unutilized inheritance.

webrtc/video_receive_stream.h

Index: webrtc/video_receive_stream.h
diff --git a/webrtc/video_receive_stream.h b/webrtc/video_receive_stream.h
index 1b46c17749c161c8318ca60e6fa2c2ba37d69187..c19c073157da017683fc3b73b9cef893553d62a4 100644
--- a/webrtc/video_receive_stream.h
+++ b/webrtc/video_receive_stream.h
@@ -20,14 +20,13 @@
 #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.
@@ -176,8 +175,24 @@ class VideoReceiveStream : public ReceiveStream {
     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