- Clean up unused voice engine DTMF code.

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 49d867bac5019df3744d45e2edafe6adf2ed3bb4..725a1561423c54e87b90ba2578eb60bbf642fad2 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -117,7 +117,7 @@ bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
 }
 
 bool AudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
-                                         uint32_t duration_ms) {
+                                         uint16_t duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
          channel_proxy_->SendTelephoneEventOutband(event, duration_ms);