Relanding https://codereview.webrtc.org/1715883002/ in pieces.

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 160a818323b139381c06ca16164b3d8b667e8491..24afcbcf58e94ad2582dd9fc88cf7860f27d3b2d 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -125,8 +125,8 @@ 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) {
+bool AudioSendStream::SendTelephoneEvent(int payload_type, int event,
+                                         int duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
          channel_proxy_->SendTelephoneEventOutband(event, duration_ms);