From e3a10e1b435d5d54ee411b9cc80d8d7dc404a6a4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= Date: Mon, 19 Aug 2019 15:45:00 +0200 Subject: [PATCH] Remove usage of RtpRtcp::SetSSRC() in video/ MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit 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 Commit-Queue: Erik Språng Cr-Commit-Position: refs/heads/master@{#28901} --- video/end_to_end_tests/bandwidth_tests.cc | 2 +- video/rtp_video_stream_receiver.cc | 8 +++++--- 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc index 4312c0e065..e9b4131d9b 100644 --- a/video/end_to_end_tests/bandwidth_tests.cc +++ b/video/end_to_end_tests/bandwidth_tests.cc @@ -201,9 +201,9 @@ TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) { 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); } diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc index 9e7ae23b2c..696aa2c7a2 100644 --- a/video/rtp_video_stream_receiver.cc +++ b/video/rtp_video_stream_receiver.cc @@ -56,7 +56,8 @@ std::unique_ptr CreateRtpRtcpModule( 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 @@ std::unique_ptr CreateRtpRtcpModule( 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 rtp_rtcp = RtpRtcp::Create(configuration); rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); @@ -183,7 +185,8 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver( 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 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver( 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;