Test RtcpParser rewritten to use rtcp packet classes

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 728 matching lines @@
           break;
       }
       return SEND_PACKET;
     }
 
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock_(&crit_);
       if (state_ == kVerifyFecPacketNotInNackList) {
         test::RtcpPacketParser rtcp_parser;
         rtcp_parser.Parse(packet, length);
-        std::vector<uint16_t> nacks = rtcp_parser.nack_item()->last_nack_list();
+        const std::vector<uint16_t>& nacks = rtcp_parser.nack()->packet_ids();
         EXPECT_TRUE(std::find(nacks.begin(), nacks.end(),
                               fec_sequence_number_) == nacks.end())
             << "Got nack for FEC packet";
         if (!nacks.empty() &&
             IsNewerSequenceNumber(nacks.back(), fec_sequence_number_)) {
           observation_complete_.Set();
         }
       }
       return SEND_PACKET;
     }
@@ skipping 1267 matching lines @@
         return DROP_PACKET;
       }
       VerifyStats();
       return SEND_PACKET;
     }
 
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       test::RtcpPacketParser rtcp_parser;
       rtcp_parser.Parse(packet, length);
-      std::vector<uint16_t> nacks = rtcp_parser.nack_item()->last_nack_list();
+      const std::vector<uint16_t>& nacks = rtcp_parser.nack()->packet_ids();
       if (!nacks.empty() && std::find(
           nacks.begin(), nacks.end(), dropped_rtp_packet_) != nacks.end()) {
         dropped_rtp_packet_requested_ = true;
       }
       return SEND_PACKET;
     }
 
     void VerifyStats() EXCLUSIVE_LOCKS_REQUIRED(&crit_) {
       if (!dropped_rtp_packet_requested_)
         return;
@@ skipping 1100 matching lines @@
         }
       }
 
       return SEND_PACKET;
     }
 
     Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       test::RtcpPacketParser rtcp_parser;
       rtcp_parser.Parse(packet, length);
       if (rtcp_parser.sender_report()->num_packets() > 0) {
-        uint32_t ssrc = rtcp_parser.sender_report()->Ssrc();
-        uint32_t rtcp_timestamp = rtcp_parser.sender_report()->RtpTimestamp();
+        uint32_t ssrc = rtcp_parser.sender_report()->sender_ssrc();
+        uint32_t rtcp_timestamp = rtcp_parser.sender_report()->rtp_timestamp();
 
         rtc::CritScope lock(&crit_);
         ValidateTimestampGap(ssrc, rtcp_timestamp, false);
       }
       return SEND_PACKET;
     }
 
     SequenceNumberUnwrapper seq_numbers_unwrapper_;
     std::map<uint32_t, std::list<int64_t>> last_observed_seq_numbers_;
     std::map<uint32_t, uint32_t> last_observed_timestamp_;
@@ skipping 564 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc