commit e3a10e1b435d5d54ee411b9cc80d8d7dc404a6a4
author Erik Språng <sprang@webrtc.org> Mon Aug 19 13:45:00 2019
committer Commit Bot <commit-bot@chromium.org> Mon Aug 19 14:20:30 2019
tree bbe5c4900603e987a67b2d5e456bb5255d817e78
parent 185243b335221d910798f8a5f55ff12fbc362558

Remove usage of RtpRtcp::SetSSRC() in video/

That method is going away in favor in construction time setting.

Bug: webrtc:10774
Change-Id: I2aba5a2537e5846a3c9438a5b376b230e84c5f32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149826
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28901}

video/end_to_end_tests/bandwidth_tests.cc

diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc
index 4312c0e..e9b4131 100644
--- a/video/end_to_end_tests/bandwidth_tests.cc
+++ b/video/end_to_end_tests/bandwidth_tests.cc
@@ -201,9 +201,9 @@
       config.clock = clock_;
       config.outgoing_transport = receive_transport_;
       config.retransmission_rate_limiter = &retransmission_rate_limiter_;
+      config.media_send_ssrc = (*receive_configs)[0].rtp.local_ssrc;
       rtp_rtcp_ = RtpRtcp::Create(config);
       rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
-      rtp_rtcp_->SetSSRC((*receive_configs)[0].rtp.local_ssrc);
       rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
     }
 

video/rtp_video_stream_receiver.cc

diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 9e7ae23..696aa2c 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -56,7 +56,8 @@
     ReceiveStatistics* receive_statistics,
     Transport* outgoing_transport,
     RtcpRttStats* rtt_stats,
-    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer) {
+    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
+    uint32_t local_ssrc) {
   RtpRtcp::Configuration configuration;
   configuration.clock = clock;
   configuration.audio = false;
@@ -66,6 +67,7 @@
   configuration.rtt_stats = rtt_stats;
   configuration.rtcp_packet_type_counter_observer =
       rtcp_packet_type_counter_observer;
+  configuration.media_send_ssrc = local_ssrc;
 
   std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::Create(configuration);
   rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
@@ -183,7 +185,8 @@
                                     rtp_receive_statistics_,
                                     transport,
                                     rtt_stats,
-                                    receive_stats_proxy)),
+                                    receive_stats_proxy,
+                                    config_.rtp.local_ssrc)),
       complete_frame_callback_(complete_frame_callback),
       keyframe_request_sender_(keyframe_request_sender),
       // TODO(bugs.webrtc.org/10336): Let |rtcp_feedback_buffer_| communicate
@@ -204,7 +207,6 @@
   RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
 
   rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
-  rtp_rtcp_->SetSSRC(config_.rtp.local_ssrc);
   rtp_rtcp_->SetRemoteSSRC(config_.rtp.remote_ssrc);
 
   static const int kMaxPacketAgeToNack = 450;