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 47e3782265625bd5cffa2597b80839696c62d9ce..23c731d095616583429ea479007915f40ea8b237 100644
--- a/content/renderer/media/rtc_peer_connection_handler.cc
+++ b/content/renderer/media/rtc_peer_connection_handler.cc
@@ -1371,10 +1371,14 @@ blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender(
     return nullptr;
   }
 
-  scoped_refptr<webrtc::AudioTrackInterface> audio_track =
+  webrtc::AudioTrackInterface* const audio_adapter =
       native_track->GetAudioAdapter();
+  if (!audio_adapter) {
+    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(audio_adapter));
   if (!sender) {
     DLOG(ERROR) << "Could not create native DTMF sender.";
     return nullptr;