Remove webrtc/stream.h and unutilized inheritance.

webrtc/audio_send_stream.h

 /*
  *  Copyright (c) 2015 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_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_SEND_STREAM_H_
 
 #include <string>
 #include <vector>
 
 #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"
 
 namespace webrtc {
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // 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.
     uint32_t local_ssrc = 0;
     int64_t bytes_sent = 0;
     int32_t packets_sent = 0;
     int32_t packets_lost = -1;
     float fraction_lost = -1.0f;
     std::string codec_name;
     int32_t ext_seqnum = -1;
@@ skipping 40 matching lines @@
     int voe_channel_id = -1;
 
     // Ownership of the encoder object is transferred to Call when the config is
     // passed to Call::CreateAudioSendStream().
     // TODO(solenberg): Implement, once we configure codecs through the new API.
     // rtc::scoped_ptr<AudioEncoder> encoder;
     int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
     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
 
 #endif  // WEBRTC_AUDIO_SEND_STREAM_H_