Refactor WVoE DTMF handling #2

talk/media/webrtc/fakewebrtccall.h

Index: talk/media/webrtc/fakewebrtccall.h
diff --git a/talk/media/webrtc/fakewebrtccall.h b/talk/media/webrtc/fakewebrtccall.h
index 2e7039014cb898ab78e120730e11ae11ae7d470c..024c50dd194fa10b6a7a3150b41dcfe1c114b942 100644
--- a/talk/media/webrtc/fakewebrtccall.h
+++ b/talk/media/webrtc/fakewebrtccall.h
@@ -49,10 +49,17 @@
 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.
@@ -64,8 +71,11 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
   }
 
   // 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_;
 };