Switch voice transport to use Call and Stream instead of VoENetwork.

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.
  */
 
 // This file contains fake implementations, for use in unit tests, of the
 // following classes:
 //
 //   webrtc::Call
 //   webrtc::AudioSendStream
 //   webrtc::AudioReceiveStream
 //   webrtc::VideoSendStream
 //   webrtc::VideoReceiveStream
 
 #ifndef WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_
 #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_
 
 #include <memory>
 #include <vector>
 
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_send_stream.h"
+#include "webrtc/base/buffer.h"
 #include "webrtc/call.h"
 #include "webrtc/video_frame.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace cricket {
 class FakeAudioSendStream final : public webrtc::AudioSendStream {
  public:
   struct TelephoneEvent {
     int payload_type = -1;
@@ skipping 29 matching lines @@
 };
 
 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_; }
-  void IncrementReceivedPackets();
+  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;
  private:
   // webrtc::ReceiveStream 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::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_;
 };
 
 class FakeVideoSendStream final : public webrtc::VideoSendStream,
                                   public webrtc::VideoCaptureInput {
  public:
   FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
                       const webrtc::VideoEncoderConfig& encoder_config);
   webrtc::VideoSendStream::Config GetConfig() const;
   webrtc::VideoEncoderConfig GetEncoderConfig() const;
   std::vector<webrtc::VideoStream> GetVideoStreams();
@@ skipping 142 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_