Replace ASSERT by RTC_DCHECK in all non-test code.

webrtc/api/peerconnection.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 2054 matching lines @@
     // VideoRtpReceiver::track().
     DestroyReceiver(track_id);
     rtc::scoped_refptr<VideoTrackInterface> video_track =
         stream->FindVideoTrack(track_id);
     if (video_track) {
       // There's no guarantee the track is still available, e.g. the track may
       // have been removed from the stream by an application.
       stream->RemoveTrack(video_track);
     }
   } else {
-    ASSERT(false && "Invalid media type");
+    RTC_NOTREACHED() << "Invalid media type";
   }
 }
 
 void PeerConnection::UpdateEndedRemoteMediaStreams() {
   std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
   for (size_t i = 0; i < remote_streams_->count(); ++i) {
     MediaStreamInterface* stream = remote_streams_->at(i);
     if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
       streams_to_remove.push_back(stream);
     }
@@ skipping 478 matching lines @@
 
 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
                                         int64_t max_size_bytes) {
   return event_log_->StartLogging(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   event_log_->StopLogging();
 }
 }  // namespace webrtc