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

talk/media/webrtc/fakewebrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2010 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 228 matching lines @@
     bool red;
     bool nack;
     bool rx_agc_enabled;
     webrtc::AgcModes rx_agc_mode;
     webrtc::AgcConfig rx_agc_config;
     int cn8_type;
     int cn16_type;
     int dtmf_type;
     int red_type;
     int nack_max_packets;
-    uint32 send_ssrc;
+    uint32_t send_ssrc;
     int send_audio_level_ext_;
     int receive_audio_level_ext_;
     int send_absolute_sender_time_ext_;
     int receive_absolute_sender_time_ext_;
     int associate_send_channel;
     DtmfInfo dtmf_info;
     std::vector<webrtc::CodecInst> recv_codecs;
     webrtc::CodecInst send_codec;
     webrtc::PacketTime last_rtp_packet_time;
     std::list<std::string> packets;
@@ skipping 32 matching lines @@
     // Ought to have all been deleted by the WebRtcVoiceMediaChannel
     // destructors, but just in case ...
     for (std::map<int, Channel*>::const_iterator i = channels_.begin();
          i != channels_.end(); ++i) {
       delete i->second;
     }
   }
 
   bool IsInited() const { return inited_; }
   int GetLastChannel() const { return last_channel_; }
-  int GetChannelFromLocalSsrc(uint32 local_ssrc) const {
+  int GetChannelFromLocalSsrc(uint32_t local_ssrc) const {
     for (std::map<int, Channel*>::const_iterator iter = channels_.begin();
          iter != channels_.end(); ++iter) {
       if (local_ssrc == iter->second->send_ssrc)
         return iter->first;
     }
     return -1;
   }
   int GetNumChannels() const { return static_cast<int>(channels_.size()); }
   bool GetPlayout(int channel) {
     return channels_[channel]->playout;
@@ skipping 537 matching lines @@
       block.extended_highest_sequence_number = kIntStatValue;
       receive_blocks->push_back(block);
     }
     return 0;
   }
   WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs,
                                  unsigned int& maxJitterMs,
                                  unsigned int& discardedPackets));
   WEBRTC_FUNC(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats)) {
     WEBRTC_CHECK_CHANNEL(channel);
-    stats.fractionLost = static_cast<int16>(kIntStatValue);
+    stats.fractionLost = static_cast<int16_t>(kIntStatValue);
     stats.cumulativeLost = kIntStatValue;
     stats.extendedMax = kIntStatValue;
     stats.jitterSamples = kIntStatValue;
     stats.rttMs = kIntStatValue;
     stats.bytesSent = kIntStatValue;
     stats.packetsSent = kIntStatValue;
     stats.bytesReceived = kIntStatValue;
     stats.packetsReceived = kIntStatValue;
     return 0;
   }
@@ skipping 282 matching lines @@
   int playout_sample_rate_;
   DtmfInfo dtmf_info_;
   FakeAudioProcessing audio_processing_;
 };
 
 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_