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 24c3d77ab27a30eac58e351381178ca90c27f832..882281c8244a1b2068f86122bb490ae7982803fe 100644
--- a/webrtc/audio_send_stream.h
+++ b/webrtc/audio_send_stream.h
@@ -17,7 +17,6 @@
 #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"
 
@@ -28,7 +27,7 @@ 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.
@@ -89,10 +88,22 @@ 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;
+  // 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]

same as in AudioReceiveStream - only keep this in impl class. This is the client
interface.

+
   // 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