Remove webrtc/stream.h and unutilized inheritance.

webrtc/media/engine/fakewebrtccall.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.
  */
 
@@ skipping 30 matching lines @@
   };
 
   explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config);
 
   const webrtc::AudioSendStream::Config& GetConfig() const;
   void SetStats(const webrtc::AudioSendStream::Stats& stats);
   TelephoneEvent GetLatestTelephoneEvent() const;
   bool IsSending() const { return sending_; }
 
  private:
-  // webrtc::SendStream implementation.
+  // webrtc::AudioSendStream implementation.
   void Start() override { sending_ = true; }
   void Stop() override { sending_ = false; }
-  void SignalNetworkState(webrtc::NetworkState state) override {}
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override {
-    return true;
-  }
 
-  // webrtc::AudioSendStream implementation.
   bool SendTelephoneEvent(int payload_type, int event,
                           int duration_ms) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   TelephoneEvent latest_telephone_event_;
   webrtc::AudioSendStream::Config config_;
   webrtc::AudioSendStream::Stats stats_;
   bool sending_ = false;
 };
 
 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
  public:
   explicit FakeAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config);
 
   const webrtc::AudioReceiveStream::Config& GetConfig() const;
   void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
   int received_packets() const { return received_packets_; }
   bool VerifyLastPacket(const uint8_t* data, size_t length) const;
   const webrtc::AudioSinkInterface* sink() const { return sink_.get(); }
-
   bool DeliverRtp(const uint8_t* packet,
                   size_t length,
-                  const webrtc::PacketTime& packet_time) override;
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override;
+                  const webrtc::PacketTime& packet_time);
+
  private:
-  // webrtc::ReceiveStream implementation.
+  // webrtc::AudioReceiveStream implementation.
   void Start() override {}
   void Stop() override {}
-  void SignalNetworkState(webrtc::NetworkState state) override {}
 
-  // webrtc::AudioReceiveStream implementation.
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
 
   webrtc::AudioReceiveStream::Config config_;
   webrtc::AudioReceiveStream::Stats stats_;
   int received_packets_;
   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
   rtc::Buffer last_packet_;
 };
 
@@ skipping 15 matching lines @@
   int GetLastHeight() const;
   int64_t GetLastTimestamp() const;
   void SetStats(const webrtc::VideoSendStream::Stats& stats);
   int num_encoder_reconfigurations() const {
     return num_encoder_reconfigurations_;
   }
 
  private:
   void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override;
 
-  // webrtc::SendStream implementation.
+  // webrtc::VideoSendStream implementation.
   void Start() override;
   void Stop() override;
-  void SignalNetworkState(webrtc::NetworkState state) override {}
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override {
-    return true;
-  }
-
-  // webrtc::VideoSendStream implementation.
   webrtc::VideoSendStream::Stats GetStats() override;
   void ReconfigureVideoEncoder(
       const webrtc::VideoEncoderConfig& config) override;
   webrtc::VideoCaptureInput* Input() override;
 
   bool sending_;
   webrtc::VideoSendStream::Config config_;
   webrtc::VideoEncoderConfig encoder_config_;
   bool codec_settings_set_;
   union VpxSettings {
@@ skipping 13 matching lines @@
 
   webrtc::VideoReceiveStream::Config GetConfig();
 
   bool IsReceiving() const;
 
   void InjectFrame(const webrtc::VideoFrame& frame);
 
   void SetStats(const webrtc::VideoReceiveStream::Stats& stats);
 
  private:
-  // webrtc::ReceiveStream implementation.
+  // webrtc::VideoReceiveStream implementation.
   void Start() override;
   void Stop() override;
-  void SignalNetworkState(webrtc::NetworkState state) override {}
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override {
-    return true;
-  }
-  bool DeliverRtp(const uint8_t* packet,
-                  size_t length,
-                  const webrtc::PacketTime& packet_time) override {
-    return true;
-  }
 
-  // webrtc::VideoReceiveStream implementation.
   webrtc::VideoReceiveStream::Stats GetStats() const override;
 
   webrtc::VideoReceiveStream::Config config_;
   bool receiving_;
   webrtc::VideoReceiveStream::Stats stats_;
 };
 
 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
  public:
   explicit FakeCall(const webrtc::Call::Config& config);
@@ skipping 59 matching lines @@
   std::vector<FakeAudioSendStream*> audio_send_streams_;
   std::vector<FakeVideoReceiveStream*> video_receive_streams_;
   std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 };
 
 }  // namespace cricket
 #endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_