Remove webrtc/stream.h and unutilized inheritance.

webrtc/audio_receive_stream.h

Index: webrtc/audio_receive_stream.h
diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h
index 5254c41780abebbba4f887419fb5d1f08f6070f5..fef684de54235cb54f8cb3d0c8abdae7437cbfce 100644
--- a/webrtc/audio_receive_stream.h
+++ b/webrtc/audio_receive_stream.h
@@ -16,8 +16,8 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/common_types.h"
 #include "webrtc/config.h"
-#include "webrtc/stream.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
@@ -31,7 +31,7 @@ class AudioSinkInterface;
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
-class AudioReceiveStream : public ReceiveStream {
+class AudioReceiveStream {
  public:
   struct Stats {
     uint32_t remote_ssrc = 0;
@@ -104,6 +104,19 @@ class AudioReceiveStream : public ReceiveStream {
     std::map<uint8_t, AudioDecoder*> decoder_map;
   };
 
+  // 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]

These don't even belong here as a virtual methods. It could be a non-virtual on
the impl class since packets are always delivered through i.e.
call_->Receiver()->DeliverPacket(...)

The client never pushes packets directly to the stream.

[pbos-webrtc, 2016/04/28 07:10:13]

Done.

+  // Deliver an incoming RTP packet.
+  virtual bool DeliverRtp(const uint8_t* packet,
+                          size_t length,
+                          const PacketTime& packet_time) = 0;
+
   virtual Stats GetStats() const = 0;
 
   // Sets an audio sink that receives unmixed audio from the receive stream.
@@ -115,6 +128,9 @@ class AudioReceiveStream : public ReceiveStream {
   // is being pulled+rendered and/or if audio is being pulled for the purposes
   // of feeding to the AEC.
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
+
+ protected:
+  virtual ~AudioReceiveStream() {}
 };
 }  // namespace webrtc