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

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 cf0a19ca4be36cd31f60855689a21dc7f895348e..d463b3da30fadefcf507f7682e797659338642e3 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -40,8 +40,8 @@ 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;
+  bool SendTelephoneEvent(int payload_type, int event,
+                          int duration_ms) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   const webrtc::AudioSendStream::Config& config() const;