Remove some usage of RtpRtcp::SetSSRC()

Bug: webrtc:10774
Change-Id: Ib8fa84f5d70ceb7e715405eae2901bcd7bdfebfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146984
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28895}

call/flexfec_receive_stream_impl.cc

@@ -122,15 +122,16 @@ std::unique_ptr<FlexfecReceiver> MaybeCreateFlexfecReceiver(
 std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
     Clock* clock,
     ReceiveStatistics* receive_statistics,
-    Transport* rtcp_send_transport,
+    const FlexfecReceiveStreamImpl::Config& config,
     RtcpRttStats* rtt_stats) {
   RtpRtcp::Configuration configuration;
   configuration.audio = false;
   configuration.receiver_only = true;
   configuration.clock = clock;
   configuration.receive_statistics = receive_statistics;
-  configuration.outgoing_transport = rtcp_send_transport;
+  configuration.outgoing_transport = config.rtcp_send_transport;
   configuration.rtt_stats = rtt_stats;
+  configuration.media_send_ssrc = config.local_ssrc;
   return RtpRtcp::Create(configuration);
 }
 
@@ -150,14 +151,13 @@ FlexfecReceiveStreamImpl::FlexfecReceiveStreamImpl(
       rtp_receive_statistics_(ReceiveStatistics::Create(clock)),
       rtp_rtcp_(CreateRtpRtcpModule(clock,
                                     rtp_receive_statistics_.get(),
-                                    config_.rtcp_send_transport,
+                                    config_,
                                     rtt_stats)),
       process_thread_(process_thread) {
   RTC_LOG(LS_INFO) << "FlexfecReceiveStreamImpl: " << config_.ToString();
 
   // RTCP reporting.
   rtp_rtcp_->SetRTCPStatus(config_.rtcp_mode);
-  rtp_rtcp_->SetSSRC(config_.local_ssrc);
   process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
 
   // Register with transport.

call/rtp_video_sender.cc

@@ -110,15 +110,23 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
   std::vector<RtpStreamSender> rtp_streams;
   const std::vector<uint32_t>& flexfec_protected_ssrcs =
       rtp_config.flexfec.protected_media_ssrcs;
-  for (uint32_t ssrc : rtp_config.ssrcs) {
+  RTC_DCHECK(rtp_config.rtx.ssrcs.empty() ||
+             rtp_config.rtx.ssrcs.size() == rtp_config.rtx.ssrcs.size());
+  for (size_t i = 0; i < rtp_config.ssrcs.size(); ++i) {
+    configuration.media_send_ssrc = rtp_config.ssrcs[i];
     bool enable_flexfec = flexfec_sender != nullptr &&
                           std::find(flexfec_protected_ssrcs.begin(),
                                     flexfec_protected_ssrcs.end(),
-                                    ssrc) != flexfec_protected_ssrcs.end();
+                                    *configuration.media_send_ssrc) !=
+                              flexfec_protected_ssrcs.end();
     configuration.flexfec_sender = enable_flexfec ? flexfec_sender : nullptr;
     auto playout_delay_oracle = absl::make_unique<PlayoutDelayOracle>();
 
     configuration.ack_observer = playout_delay_oracle.get();
+    if (rtp_config.rtx.ssrcs.size() > i) {
+      configuration.rtx_send_ssrc = rtp_config.rtx.ssrcs[i];
+    }
+
     auto rtp_rtcp = RtpRtcp::Create(configuration);
     rtp_rtcp->SetSendingStatus(false);
     rtp_rtcp->SetSendingMediaStatus(false);
@@ -571,7 +579,6 @@ void RtpVideoSender::ConfigureSsrcs() {
   for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) {
     uint32_t ssrc = rtp_config_.ssrcs[i];
     RtpRtcp* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
-    rtp_rtcp->SetSSRC(ssrc);
 
     // Restore RTP state if previous existed.
     auto it = suspended_ssrcs_.find(ssrc);
@@ -587,12 +594,10 @@ void RtpVideoSender::ConfigureSsrcs() {
   if (rtp_config_.rtx.ssrcs.empty())
     return;
 
-  // Configure RTX SSRCs.
   RTC_DCHECK_EQ(rtp_config_.rtx.ssrcs.size(), rtp_config_.ssrcs.size());
   for (size_t i = 0; i < rtp_config_.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = rtp_config_.rtx.ssrcs[i];
     RtpRtcp* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
-    rtp_rtcp->SetRtxSsrc(ssrc);
     auto it = suspended_ssrcs_.find(ssrc);
     if (it != suspended_ssrcs_.end())
       rtp_rtcp->SetRtxState(it->second);

modules/rtp_rtcp/source/nack_rtx_unittest.cc

@@ -135,11 +135,11 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
     configuration.receive_statistics = receive_statistics_.get();
     configuration.outgoing_transport = &transport_;
     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
+    configuration.media_send_ssrc = kTestSsrc;
     rtp_rtcp_module_ = RtpRtcp::Create(configuration);
     rtp_sender_video_ = absl::make_unique<RTPSenderVideo>(
         &fake_clock, rtp_rtcp_module_->RtpSender(), nullptr,
         &playout_delay_oracle_, nullptr, false, false, FieldTrialBasedConfig());
-    rtp_rtcp_module_->SetSSRC(kTestSsrc);
     rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
     rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
     EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true));

modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc

@@ -180,7 +180,6 @@ class RtpRtcpImplTest : public ::testing::Test {
 
   void SetUp() override {
     // Send module.
-    sender_.impl_->SetSSRC(kSenderSsrc);
     EXPECT_EQ(0, sender_.impl_->SetSendingStatus(true));
     sender_.impl_->SetSendingMediaStatus(true);
     sender_.SetRemoteSsrc(kReceiverSsrc);