Remove webrtc/stream.h and unutilized inheritance.

webrtc/audio_send_stream.h

Index: webrtc/audio_send_stream.h
diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h
index 18e71253f1158dfe5681c79000d77f2f7faedc46..d5d16f2711eed09f1df5d359196aed0a61f39e28 100644
--- a/webrtc/audio_send_stream.h
+++ b/webrtc/audio_send_stream.h
@@ -18,7 +18,6 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/stream.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
@@ -29,7 +28,7 @@ namespace webrtc {
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
-class AudioSendStream : public SendStream {
+class AudioSendStream {
  public:
   struct Stats {
     // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
@@ -90,10 +89,20 @@ class AudioSendStream : public SendStream {
     int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.
   };
 
+  // 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;
+
   // TODO(solenberg): Make payload_type a config property instead.
   virtual bool SendTelephoneEvent(int payload_type, int event,
                                   int duration_ms) = 0;
   virtual Stats GetStats() const = 0;
+
+ protected:
+  virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc