Refactor WVoE DTMF handling #2

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 4dd952295667f7a231497b1202cd70ab10f5ed53..2ff388bbca10821249e27542c6867164e4e99e1f 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -102,6 +102,13 @@ bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return false;
 }
 
+bool AudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
+                                         uint32_t duration_ms) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
+         channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
+}
+
 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   webrtc::AudioSendStream::Stats stats;