- Clean up unused voice engine DTMF code.

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 17 matching lines @@
 #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;
-    uint8_t event_code = 0;
-    uint32_t duration_ms = 0;
+    int event_code = 0;
+    int 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;
   bool IsSending() const { return sending_; }
 
  private:
   // webrtc::SendStream 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, uint8_t event,
-                          uint32_t duration_ms) override;
+  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:
@@ skipping 175 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_