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

webrtc/voice_engine/test/auto_test/voe_conference_test.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 54 matching lines @@
   // Start NTP time is the time when a stream is played out, rather than
   // when it is added.
   webrtc::SleepMs(kDelayMs);
   EXPECT_TRUE(trans.StartPlayout(id_2));
 
   const int kMaxRunTimeMs = 25000;
   const int kNeedSuccessivePass = 3;
   const int kStatsRequestIntervalMs = 1000;
   const int kStatsBufferSize = 3;
 
-  uint32 deadline = rtc::TimeAfter(kMaxRunTimeMs);
+  uint32_t deadline = rtc::TimeAfter(kMaxRunTimeMs);
   // Run the following up to |kMaxRunTimeMs| milliseconds.
   int successive_pass = 0;
   webrtc::CallStatistics stats_1;
   webrtc::CallStatistics stats_2;
   std::queue<Stats> stats_buffer;
 
   while (rtc::TimeIsLater(rtc::Time(), deadline) &&
       successive_pass < kNeedSuccessivePass) {
     webrtc::SleepMs(kStatsRequestIntervalMs);
 
@@ skipping 81 matching lines @@
     // depends on the sleep timer. So we allow a small off from |kPackets|.
     EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2);
     EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2);
     EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2);
   }
 
   remove(silence_file.c_str());
 }
 
 }  // namespace voetest