Use suffixed {uint,int}{8,16,32,64}_t types.

talk/app/webrtc/peerconnection.cc

 /*
  * libjingle
  * Copyright 2012 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 823 matching lines @@
   observer_->OnRemoveStream(stream);
 }
 
 void PeerConnection::OnAddDataChannel(DataChannelInterface* data_channel) {
   observer_->OnDataChannel(DataChannelProxy::Create(signaling_thread(),
                                                     data_channel));
 }
 
 void PeerConnection::OnAddRemoteAudioTrack(MediaStreamInterface* stream,
                                            AudioTrackInterface* audio_track,
-                                           uint32 ssrc) {
+                                           uint32_t ssrc) {
   receivers_.push_back(new AudioRtpReceiver(audio_track, ssrc, session_.get()));
 }
 
 void PeerConnection::OnAddRemoteVideoTrack(MediaStreamInterface* stream,
                                            VideoTrackInterface* video_track,
-                                           uint32 ssrc) {
+                                           uint32_t ssrc) {
   receivers_.push_back(new VideoRtpReceiver(video_track, ssrc, session_.get()));
 }
 
 // TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
 // description.
 void PeerConnection::OnRemoveRemoteAudioTrack(
     MediaStreamInterface* stream,
     AudioTrackInterface* audio_track) {
   auto it = FindReceiverForTrack(audio_track);
   if (it == receivers_.end()) {
@@ skipping 13 matching lines @@
     LOG(LS_WARNING) << "RtpReceiver for track with id " << video_track->id()
                     << " doesn't exist.";
   } else {
     (*it)->Stop();
     receivers_.erase(it);
   }
 }
 
 void PeerConnection::OnAddLocalAudioTrack(MediaStreamInterface* stream,
                                           AudioTrackInterface* audio_track,
-                                          uint32 ssrc) {
+                                          uint32_t ssrc) {
   senders_.push_back(new AudioRtpSender(audio_track, ssrc, session_.get()));
   stats_->AddLocalAudioTrack(audio_track, ssrc);
 }
 
 void PeerConnection::OnAddLocalVideoTrack(MediaStreamInterface* stream,
                                           VideoTrackInterface* video_track,
-                                          uint32 ssrc) {
+                                          uint32_t ssrc) {
   senders_.push_back(new VideoRtpSender(video_track, ssrc, session_.get()));
 }
 
 // TODO(deadbeef): Keep RtpSenders around even if track goes away in local
 // description.
 void PeerConnection::OnRemoveLocalAudioTrack(MediaStreamInterface* stream,
                                              AudioTrackInterface* audio_track,
-                                             uint32 ssrc) {
+                                             uint32_t ssrc) {
   auto it = FindSenderForTrack(audio_track);
   if (it == senders_.end()) {
     LOG(LS_WARNING) << "RtpSender for track with id " << audio_track->id()
                     << " doesn't exist.";
     return;
   } else {
     (*it)->Stop();
     senders_.erase(it);
   }
   stats_->RemoveLocalAudioTrack(audio_track, ssrc);
@@ skipping 79 matching lines @@
 std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator
 PeerConnection::FindReceiverForTrack(MediaStreamTrackInterface* track) {
   return std::find_if(
       receivers_.begin(), receivers_.end(),
       [track](const rtc::scoped_refptr<RtpReceiverInterface>& receiver) {
         return receiver->track() == track;
       });
 }
 
 }  // namespace webrtc