Fixing issues with timestamps in video_quality_test.cc.

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>
@@ skipping 100 matching lines @@
       delete thread;
     }
   }
 
   virtual void SetReceiver(PacketReceiver* receiver) { receiver_ = receiver; }
 
   DeliveryStatus DeliverPacket(MediaType media_type,
                                const uint8_t* packet,
                                size_t length,
                                const PacketTime& packet_time) override {
+    // Ignore timestamps of RTCP packets. They're not synchronized with
+    // RTP packet timestamps and so they would confuse wrap_handler_.
+    if (RtpHeaderParser::IsRtcp(packet, length)) {
+      return receiver_->DeliverPacket(media_type, packet, length, packet_time);
+    }
+

[sprang_webrtc, 2016/03/18 10:05:29]

Thinking some more about this, perhaps we should even filter on media_type =
video. Though there shouldn't be any other type, right now...

     RtpUtility::RtpHeaderParser parser(packet, length);
     RTPHeader header;
     parser.Parse(&header);
     {
       rtc::CritScope lock(&crit_);
       int64_t timestamp =
           wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
       recv_times_[timestamp] =
           Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     }
@@ skipping 67 matching lines @@
     if (frames_recorded_ < frames_to_process_)
       encoded_frame_size_.AddSample(frame.length_);
   }
 
   void RenderFrame(const VideoFrame& video_frame,
                    int time_to_render_ms) override {
     int64_t render_time_ms =
         Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
 
     rtc::CritScope lock(&crit_);
-    uint32_t send_timestamp =
+    int64_t send_timestamp =
         wrap_handler_.Unwrap(video_frame.timestamp() - rtp_timestamp_delta_);
 
     while (wrap_handler_.Unwrap(frames_.front().timestamp()) < send_timestamp) {
       if (last_rendered_frame_.IsZeroSize()) {
         // No previous frame rendered, this one was dropped after sending but
         // before rendering.
         ++dropped_frames_before_rendering_;
         frames_.pop_front();
         RTC_CHECK(!frames_.empty());
         continue;
@@ skipping 905 matching lines @@
   video_send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc