EndToEndTest.ReceivesAndRetransmitsNack now handles reordering.

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 453 matching lines @@
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
       // Never drop retransmitted packets.
       if (dropped_packets_.find(header.sequenceNumber) !=
           dropped_packets_.end()) {
         retransmitted_packets_.insert(header.sequenceNumber);
-        if (nacks_left_ <= 0 &&
-            retransmitted_packets_.size() == dropped_packets_.size()) {
-          observation_complete_.Set();
-        }
         return SEND_PACKET;
       }
 
+      if (nacks_left_ <= 0 &&
+          retransmitted_packets_.size() == dropped_packets_.size()) {
+        observation_complete_.Set();
+      }
+
       ++sent_rtp_packets_;
 
       // Enough NACKs received, stop dropping packets.
       if (nacks_left_ <= 0)
         return SEND_PACKET;
 
       // Check if it's time for a new loss burst.
       if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
         packets_left_to_drop_ = kLossBurstSize;
 
@@ skipping 3573 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc