Unwrap timestamps in VideoAnalyzer

webrtc/video/video_quality_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.
  */
 #include <stdio.h>
 
 #include <algorithm>
 #include <deque>
 #include <map>
 #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/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/test/layer_filtering_transport.h"
 #include "webrtc/test/run_loop.h"
 #include "webrtc/test/statistics.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/video_renderer.h"
@@ skipping 78 matching lines @@
 
   DeliveryStatus DeliverPacket(MediaType media_type,
                                const uint8_t* packet,
                                size_t length,
                                const PacketTime& packet_time) override {
     RtpUtility::RtpHeaderParser parser(packet, length);
     RTPHeader header;
     parser.Parse(&header);
     {
       rtc::CritScope lock(&crit_);
-      recv_times_[header.timestamp - rtp_timestamp_delta_] =
+      int64_t timestamp = wrap_handler_.Unwrap(header.timestamp);
+      recv_times_[timestamp - rtp_timestamp_delta_] =
           Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     }
 
     return receiver_->DeliverPacket(media_type, packet, length, packet_time);
   }
 
   // EncodingTimeObserver.
   void OnReportEncodedTime(int64_t ntp_time_ms, int encode_time_ms) override {
     rtc::CritScope crit(&comparison_lock_);
     samples_encode_time_ms_[ntp_time_ms] = encode_time_ms;
@@ skipping 19 matching lines @@
                const PacketOptions& options) override {
     RtpUtility::RtpHeaderParser parser(packet, length);
     RTPHeader header;
     parser.Parse(&header);
 
     int64_t current_time =
         Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     bool result = transport_->SendRtp(packet, length, options);
     {
       rtc::CritScope lock(&crit_);
+      int64_t timestamp = wrap_handler_.Unwrap(header.timestamp);
+
       if (rtp_timestamp_delta_ == 0) {
-        rtp_timestamp_delta_ = header.timestamp - first_send_frame_.timestamp();
+        rtp_timestamp_delta_ = timestamp - first_send_frame_.timestamp();
         first_send_frame_.Reset();
       }
-      uint32_t timestamp = header.timestamp - rtp_timestamp_delta_;
+      timestamp -= rtp_timestamp_delta_;
       send_times_[timestamp] = current_time;
       if (!transport_->DiscardedLastPacket() &&
           header.ssrc == ssrc_to_analyze_) {
         encoded_frame_sizes_[timestamp] +=
             length - (header.headerLength + header.paddingLength);
       }
     }
     return result;
   }
 
@@ skipping 145 matching lines @@
     size_t encoded_frame_size;
     double psnr;
     double ssim;
   };
 
   void AddFrameComparison(const VideoFrame& reference,
                           const VideoFrame& render,
                           bool dropped,
                           int64_t render_time_ms)
       EXCLUSIVE_LOCKS_REQUIRED(crit_) {
-    int64_t send_time_ms = send_times_[reference.timestamp()];
-    send_times_.erase(reference.timestamp());
-    int64_t recv_time_ms = recv_times_[reference.timestamp()];
-    recv_times_.erase(reference.timestamp());
+    int64_t reference_timestamp = wrap_handler_.Unwrap(reference.timestamp());
+    int64_t send_time_ms = send_times_[reference_timestamp];
+    send_times_.erase(reference_timestamp);
+    int64_t recv_time_ms = recv_times_[reference_timestamp];
+    recv_times_.erase(reference_timestamp);
 
     // TODO(ivica): Make this work for > 2 streams.
-    auto it = encoded_frame_sizes_.find(reference.timestamp());
+    auto it = encoded_frame_sizes_.find(reference_timestamp);
     if (it == encoded_frame_sizes_.end())
-      it = encoded_frame_sizes_.find(reference.timestamp() - 1);
+      it = encoded_frame_sizes_.find(reference_timestamp - 1);
     size_t encoded_size = it == encoded_frame_sizes_.end() ? 0 : it->second;
     if (it != encoded_frame_sizes_.end())
       encoded_frame_sizes_.erase(it);
 
     VideoFrame reference_copy;
     VideoFrame render_copy;
     reference_copy.CopyFrame(reference);
     render_copy.CopyFrame(render);
 
     rtc::CritScope crit(&comparison_lock_);
@@ skipping 226 matching lines @@
   const int frames_to_process_;
   int frames_recorded_;
   int frames_processed_;
   int dropped_frames_;
   int64_t last_render_time_;
   uint32_t rtp_timestamp_delta_;
 
   rtc::CriticalSection crit_;
   std::deque<VideoFrame> frames_ GUARDED_BY(crit_);
   VideoFrame last_rendered_frame_ GUARDED_BY(crit_);
-  std::map<uint32_t, int64_t> send_times_ GUARDED_BY(crit_);
-  std::map<uint32_t, int64_t> recv_times_ GUARDED_BY(crit_);
-  std::map<uint32_t, size_t> encoded_frame_sizes_ GUARDED_BY(crit_);
+  rtc::TimestampWrapAroundHandler wrap_handler_ GUARDED_BY(crit_);
+  std::map<int64_t, int64_t> send_times_ GUARDED_BY(crit_);
+  std::map<int64_t, int64_t> recv_times_ GUARDED_BY(crit_);
+  std::map<int64_t, size_t> encoded_frame_sizes_ GUARDED_BY(crit_);
   VideoFrame first_send_frame_ GUARDED_BY(crit_);
   const double avg_psnr_threshold_;
   const double avg_ssim_threshold_;
 
   rtc::CriticalSection comparison_lock_;
   std::vector<rtc::PlatformThread*> comparison_thread_pool_;
   rtc::PlatformThread stats_polling_thread_;
   rtc::Event comparison_available_event_;
   std::deque<FrameComparison> comparisons_ GUARDED_BY(comparison_lock_);
   rtc::Event done_;
@@ skipping 463 matching lines @@
   video_send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc