Remove cast to LocalAudioSource from AudioRtpSender.

talk/app/webrtc/rtpsender.cc

 /*
  * libjingle
  * Copyright 2015 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 166 matching lines @@
     if (stats_) {
       stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
     }
   }
   stopped_ = true;
 }
 
 void AudioRtpSender::SetAudioSend() {
   RTC_DCHECK(!stopped_ && can_send_track());
   cricket::AudioOptions options;
+#if !defined(WEBRTC_CHROMIUM_BUILD)
+  // TODO(tommi): Remove this hack when we move CreateAudioSource out of
+  // PeerConnection.  This is a bit of a strange way to apply local audio
+  // options since it is also applied to all streams/channels, local or remote.
   if (track_->enabled() && track_->GetSource() &&
       !track_->GetSource()->remote()) {
     // TODO(xians): Remove this static_cast since we should be able to connect
     // a remote audio track to a peer connection.
     options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
   }
+#endif
 
   // Use the renderer if the audio track has one, otherwise use the sink
   // adapter owned by this class.
   cricket::AudioRenderer* renderer =
       track_->GetRenderer() ? track_->GetRenderer() : sink_adapter_.get();
   ASSERT(renderer != nullptr);
   provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
 }
 
 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
@@ skipping 108 matching lines @@
   RTC_DCHECK(!stopped_ && can_send_track());
   const cricket::VideoOptions* options = nullptr;
   VideoSourceInterface* source = track_->GetSource();
   if (track_->enabled() && source) {
     options = source->options();
   }
   provider_->SetVideoSend(ssrc_, track_->enabled(), options);
 }
 
 }  // namespace webrtc