Change default timestamp to 64 bits in all webrtc directories.

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <algorithm>
@@ skipping 2912 matching lines @@
         }
       }
 
       static const int32_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
       auto timestamp_it = last_observed_timestamp_.find(ssrc);
       if (timestamp_it == last_observed_timestamp_.end()) {
         last_observed_timestamp_[ssrc] = timestamp;
       } else {
         // Verify timestamps are reasonably close.
         uint32_t latest_observed = timestamp_it->second;
-        int32_t timestamp_gap = rtc::TimeDiff(timestamp, latest_observed);
+        int32_t timestamp_gap = rtc::TimeDiff32(timestamp, latest_observed);
         EXPECT_LE(std::abs(timestamp_gap), kMaxTimestampGap)
             << "Gap in timestamps (" << latest_observed << " -> "
             << timestamp << ") too large for SSRC: " << ssrc << ".";
         timestamp_it->second = timestamp;
       }
 
       rtc::CritScope lock(&crit_);
       // Wait for media packets on all ssrcs.
       if (!ssrc_observed_[ssrc] && !only_padding) {
         ssrc_observed_[ssrc] = true;
@@ skipping 554 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc