MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/rtc_peer_connection_handler.cc

Index: content/renderer/media/rtc_peer_connection_handler.cc
diff --git a/content/renderer/media/rtc_peer_connection_handler.cc b/content/renderer/media/rtc_peer_connection_handler.cc
index 171a38868b2963049303ccb57406f61a734b1c77..4f649c740e0a2c13c24cb83b2d4d6ce75e90ec19 100644
--- a/content/renderer/media/rtc_peer_connection_handler.cc
+++ b/content/renderer/media/rtc_peer_connection_handler.cc
@@ -30,8 +30,9 @@
 #include "content/renderer/media/rtc_data_channel_handler.h"
 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
+#include "content/renderer/media/webrtc/processed_local_audio_track.h"
+#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_uma_histograms.h"
 #include "content/renderer/render_thread_impl.h"
@@ -1500,10 +1501,16 @@ blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender(
     return nullptr;
   }
 
-  scoped_refptr<webrtc::AudioTrackInterface> audio_track =
-      native_track->GetAudioAdapter();
+  ProcessedLocalAudioTrack* const rtc_audio_track =
+      ProcessedLocalAudioTrack::From(native_track);
+  if (!rtc_audio_track) {
+    DLOG(ERROR) << "WebRTC features are not available on this audio track.";
+    return nullptr;
+  }
+
   rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender(
-      native_peer_connection_->CreateDtmfSender(audio_track.get()));
+      native_peer_connection_->CreateDtmfSender(
+          rtc_audio_track->adapter().get()));
   if (!sender) {
     DLOG(ERROR) << "Could not create native DTMF sender.";
     return nullptr;