Refactor WVoE DTMF handling #2

talk/media/webrtc/fakewebrtccall.h

 /*
  * libjingle
  * Copyright 2015 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 31 matching lines @@
 #include "webrtc/call.h"
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_send_stream.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;
+    uint8_t event_code = 0;
+    uint32_t duration_ms = 0;
+  };
+
   explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config);
 
   const webrtc::AudioSendStream::Config& GetConfig() const;
   void SetStats(const webrtc::AudioSendStream::Stats& stats);
+  TelephoneEvent GetLatestTelephoneEvent() const;
 
  private:
   // webrtc::SendStream 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::AudioSendStream implementation.
+  bool SendTelephoneEvent(int payload_type, uint8_t event,
+                          uint32_t duration_ms) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
+  TelephoneEvent latest_telephone_event_;
   webrtc::AudioSendStream::Config config_;
   webrtc::AudioSendStream::Stats stats_;
 };
 
 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
  public:
   explicit FakeAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config);
 
   const webrtc::AudioReceiveStream::Config& GetConfig() const;
@@ skipping 168 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_