Use existing parser in ReceivesAndRetransmitsNack.

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 355 matching lines @@
 }
 
 TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
   static const int kNumberOfNacksToObserve = 2;
   static const int kLossBurstSize = 2;
   static const int kPacketsBetweenLossBursts = 9;
   class NackObserver : public test::EndToEndTest {
    public:
     NackObserver()
         : EndToEndTest(kLongTimeoutMs),
-          rtp_parser_(RtpHeaderParser::Create()),
           sent_rtp_packets_(0),
           packets_left_to_drop_(0),
           nacks_left_(kNumberOfNacksToObserve) {}
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       RTPHeader header;
-      EXPECT_TRUE(rtp_parser_->Parse(packet, length, &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;
@@ skipping 42 matching lines @@
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     }
 
     void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out waiting for packets to be NACKed, retransmitted and "
              "rendered.";
     }
 
     rtc::CriticalSection crit_;
     rtc::scoped_ptr<RtpHeaderParser> rtp_parser_;

[sprang_webrtc, 2015/12/07 14:37:15]

Now this is unused and should be removed as well.

[pbos-webrtc, 2015/12/07 14:55:34]

Sorry about that, https://codereview.webrtc.org/1506743003/

     std::set<uint16_t> dropped_packets_;
     std::set<uint16_t> retransmitted_packets_;
     uint64_t sent_rtp_packets_;
     int packets_left_to_drop_;
     int nacks_left_ GUARDED_BY(&crit_);
   } test;
 
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 
@@ skipping 2766 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
 }  // namespace webrtc