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>
 #include <list>
 #include <map>
 #include <memory>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
-#include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
@@ skipping 2893 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);
+        // Wraparound handling is unnecessary here as long as an int variable
+        // is used to store the result.
+        int32_t timestamp_gap = 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