Fix timing frames and FEC conflict

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 445 matching lines @@
                                 fake_clock_.TimeInMilliseconds(), false,
                                 PacedPacketInfo());
 
   const auto& packet = transport_.last_sent_packet();
   uint16_t transport_seq_no;
   EXPECT_TRUE(packet.GetExtension<TransportSequenceNumber>(&transport_seq_no));
   EXPECT_EQ(kTransportSequenceNumber, transport_seq_no);
   EXPECT_EQ(transport_.last_packet_id_, transport_seq_no);
 }
 
-// Disabled due to webrtc:7859. Until issues with FEC resolved, pacer exit
-// timstamp is not updated in the pacer.
-TEST_P(RtpSenderTestWithoutPacer, DISABLED_WritesTimestampToTimingExtension) {
+TEST_P(RtpSenderTestWithoutPacer, WritesTimestampToTimingExtension) {
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoTiming, kVideoTimingExtensionId));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   auto packet = rtp_sender_->AllocatePacket();
   packet->SetPayloadType(kPayload);
   packet->SetMarker(true);
   packet->SetTimestamp(kTimestamp);
   packet->set_capture_time_ms(capture_time_ms);
   const VideoTiming kVideoTiming = {0u, 0u, 0u, 0u, 0u, 0u, true};
@@ skipping 1139 matching lines @@
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderTestWithoutPacer,
                         ::testing::Bool());
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderVideoTest,
                         ::testing::Bool());
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderAudioTest,
                         ::testing::Bool());
 }  // namespace webrtc