Refactor WVoE DTMF handling #2

webrtc/audio/audio_send_stream.h

Index: webrtc/audio/audio_send_stream.h
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index b670efee3cbfad5074cbbc77050ed7e590f5230f..88304fd70216a4f0875a59e1052c4ec25b465f9f 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -37,6 +37,8 @@ class AudioSendStream final : public webrtc::AudioSendStream {
   bool DeliverRtcp(const uint8_t* packet, size_t length) override;
 
   // webrtc::AudioSendStream implementation.
+  bool SendTelephoneEvent(int payload_type, uint8_t event,
+                          uint32_t duration_ms) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   const webrtc::AudioSendStream::Config& config() const;