Persist RTP state for FlexFEC.

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 828 matching lines @@
 
 TEST_P(RtpSenderTest, SendFlexfecPackets) {
   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,
-                               &fake_clock_);
+                               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);
@@ skipping 36 matching lines @@
 
 TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
   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,
-                               &fake_clock_);
+                               nullptr /* rtp_state */, &fake_clock_);
 
   // Reset |rtp_sender_| to use FlexFEC.
   rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
                                   &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);
@@ skipping 20 matching lines @@
 
 TEST_P(RtpSenderTest, FecOverheadRate) {
   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,
-                               &fake_clock_);
+                               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);
@@ skipping 583 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