Fixing typo

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 "webrtc/video/video_quality_test.h"
@@ skipping 212 matching lines @@
                                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);
     }
 
     RtpUtility::RtpHeaderParser parser(packet, length);
     RTPHeader header;
     parser.Parse(&header);
-    if (!IsFlexfec(header.payloadType) && header.ssrc != ssrc_to_analyze_) {
+    if (!IsFlexfec(header.payloadType) && header.ssrc == ssrc_to_analyze_) {
       // Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
       // (FlexFEC and media are sent on different SSRCs, which have different
       // timestamps spaces.)
       // Also ignore packets from wrong SSRC.
       rtc::CritScope lock(&crit_);
       int64_t timestamp =
           wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
       recv_times_[timestamp] =
           Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     }
@@ skipping 34 matching lines @@
     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_);
       if (rtp_timestamp_delta_ == 0 && header.ssrc == ssrc_to_analyze_) {
+        RTC_CHECK(static_cast<bool>(first_sent_timestamp_));
         rtp_timestamp_delta_ = header.timestamp - *first_sent_timestamp_;
       }
 
       if (!IsFlexfec(header.payloadType) && header.ssrc == ssrc_to_analyze_) {
         // Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
         // (FlexFEC and media are sent on different SSRCs, which have different
         // timestamps spaces.)
         // Also ignore packets from wrong SSRC.
         int64_t timestamp =
             wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
@@ skipping 1291 matching lines @@
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 
 }  // namespace webrtc