Remove webrtc/stream.h and unutilized inheritance.

webrtc/audio_receive_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_RECEIVE_STREAM_H_
 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_
 
 #include <map>
 #include <memory>
 #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"
 
 namespace webrtc {
 
 class AudioDecoder;
 class AudioSinkInterface;
 
 // 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 AudioReceiveStream : public ReceiveStream {
+class AudioReceiveStream {
  public:
   struct Stats {
     uint32_t remote_ssrc = 0;
     int64_t bytes_rcvd = 0;
     uint32_t packets_rcvd = 0;
     uint32_t packets_lost = 0;
     float fraction_lost = 0.0f;
     std::string codec_name;
     uint32_t ext_seqnum = 0;
     uint32_t jitter_ms = 0;
@@ skipping 52 matching lines @@
     // stream to one audio stream. Tracked by issue webrtc:4762.
     std::string sync_group;
 
     // Decoders for every payload that we can receive. Call owns the
     // AudioDecoder instances once the Config is submitted to
     // Call::CreateReceiveStream().
     // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
     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.
   // Ownership of the sink is passed to the stream and can be used by the
   // caller to do lifetime management (i.e. when the sink's dtor is called).
   // Only one sink can be set and passing a null sink clears an existing one.
   // NOTE: Audio must still somehow be pulled through AudioTransport for audio
   // to stream through this sink. In practice, this happens if mixed audio
   // 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
 
 #endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_