Don't disable fec for packets without timing frames extension, even if

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 933 matching lines @@
 
 // TODO(ilnik): because of webrtc:7859. Once FEC moved below pacer, this test
 // should be removed.
 TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
   constexpr int kMediaPayloadType = 127;
   constexpr int kFlexfecPayloadType = 118;
   constexpr uint32_t kMediaSsrc = 1234;
   constexpr uint32_t kFlexfecSsrc = 5678;
   const std::vector<RtpExtension> kNoRtpExtensions;
   const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
+
   FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
                                kNoRtpExtensions, kNoRtpExtensionSizes,
                                nullptr /* rtp_state */, &fake_clock_);
 
   // Reset |rtp_sender_| to use FlexFEC.
   rtp_sender_.reset(new RTPSender(
       false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
       &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
       &mock_rtc_event_log_, &send_packet_observer_,
       &retransmission_rate_limiter_, nullptr));
   rtp_sender_->SetSSRC(kMediaSsrc);
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetSendPayloadType(kMediaPayloadType);
   rtp_sender_->SetStorePacketsStatus(true, 10);
 
+  // Need extension to be registered for timing frames to be sent.
+  ASSERT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
+                   kRtpExtensionVideoTiming, kVideoTimingExtensionId));
+
   // Parameters selected to generate a single FEC packet per media packet.
   FecProtectionParams params;
   params.fec_rate = 15;
   params.max_fec_frames = 1;
   params.fec_mask_type = kFecMaskRandom;
   rtp_sender_->SetFecParameters(params, params);
 
   EXPECT_CALL(mock_paced_sender_,
               InsertPacket(RtpPacketSender::kLowPriority, kMediaSsrc, kSeqNum,
                            _, _, false));
@@ skipping 733 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