MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

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 eeedd12180310543cbbefeda97903c921e7a30aa..c8f9ffac3082459fb72e2dda5d0348123abad0f9 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_dtmf_sender_handler.h"
 #include "content/renderer/media/rtc_media_constraints.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"
@@ -1364,17 +1365,22 @@ blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender(
   TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDTMFSender");
   DVLOG(1) << "createDTMFSender.";
 
-  MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::GetTrack(track);
+  MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::Get(track);
   if (!native_track || !native_track->is_local_track() ||
       track.source().type() != blink::WebMediaStreamSource::TypeAudio) {
     DLOG(ERROR) << "The DTMF sender requires a local audio track.";
     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()));
   if (!sender) {
     DLOG(ERROR) << "Could not create native DTMF sender.";
     return nullptr;