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

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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}
This commit is contained in:
Erik Språng 2019-08-19 15:45:00 +02:00 committed by Commit Bot
parent 185243b335
commit e3a10e1b43
2 changed files with 6 additions and 4 deletions
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2
video/end_to_end_tests/bandwidth_tests.cc
View File

@ -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);
}

8
video/rtp_video_stream_receiver.cc
View File

@ -56,7 +56,8 @@ std::unique_ptr<RtpRtcp> 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<RtpRtcp> 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<RtpRtcp> 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;