- Clean up unused voice engine DTMF code.

webrtc/media/engine/fakewebrtccall.h

Index: webrtc/media/engine/fakewebrtccall.h
diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h
index 89a644a2960a121e07fb8707ebcd8e9fd8226f3e..41a92dfac069fa576ec4e012b2dbca237094d27f 100644
--- a/webrtc/media/engine/fakewebrtccall.h
+++ b/webrtc/media/engine/fakewebrtccall.h
@@ -35,8 +35,8 @@ 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);
@@ -56,8 +56,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
   }
 
   // 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_;