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

webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc

 /*
  *  Copyright (c) 2015 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 189 matching lines @@
   while (true) {
     Packet packet;
     {
       webrtc::CriticalSectionScoped lock(pq_crit_.get());
       if (packet_queue_.empty())
         break;
       packet = packet_queue_.front();
       packet_queue_.pop_front();
     }
 
-    int32 elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_);
-    int32 sleep_ms = rtt_ms_ / 2 - elapsed_time_ms;
+    int32_t elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_);
+    int32_t sleep_ms = rtt_ms_ / 2 - elapsed_time_ms;
     if (sleep_ms > 0) {
       // Every packet should be delayed by half of RTT.
       webrtc::SleepMs(sleep_ms);
     }
 
     SendPacket(packet);
   }
   return true;
 }
 
@@ skipping 59 matching lines @@
 bool ConferenceTransport::GetReceiverStatistics(unsigned int id,
                                                 webrtc::CallStatistics* stats) {
   int dst = GetReceiverChannelForSsrc(id);
   if (dst == -1) {
     return false;
   }
   EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
   return true;
 }
 }  // namespace voetest