Remove the audio/video split for the RTCP report intervals.
This is a follow up of a comment in https://webrtc-review.googlesource.com/c/src/+/110105 It was not very useful to split the audio and video report interval since the RTCP module can only either be audio or video. The recent it was written that way in https://webrtc-review.googlesource.com/c/src/+/43201/ was because that was a straightforward transition from two global constants to two variable. Bug: webrtc:8789 Change-Id: I2293de14ba5f363351f379a02022ed5dc7b8d458 Reviewed-on: https://webrtc-review.googlesource.com/c/110824 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#25741}
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@ -79,11 +79,12 @@ class RtcpSenderTest : public ::testing::Test {
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configuration.clock = &clock_;
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configuration.outgoing_transport = &test_transport_;
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configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
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configuration.rtcp_report_interval_ms = 1000;
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rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
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rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
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nullptr, nullptr, &test_transport_,
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configuration.rtcp_interval_config));
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configuration.rtcp_report_interval_ms));
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rtcp_sender_->SetSSRC(kSenderSsrc);
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rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
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rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
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@ -186,8 +187,7 @@ TEST_F(RtcpSenderTest, SendConsecutiveSrWithExactSlope) {
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TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
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rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
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nullptr, nullptr, &test_transport_,
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RtcpIntervalConfig{}));
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nullptr, nullptr, &test_transport_, 1000));
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rtcp_sender_->SetSSRC(kSenderSsrc);
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rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
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rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
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@ -205,8 +205,7 @@ TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
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TEST_F(RtcpSenderTest, DoNotSendCompundBeforeRtp) {
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rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
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nullptr, nullptr, &test_transport_,
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RtcpIntervalConfig{}));
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nullptr, nullptr, &test_transport_, 1000));
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rtcp_sender_->SetSSRC(kSenderSsrc);
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rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
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rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
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@ -507,7 +506,7 @@ TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
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RtcpPacketTypeCounterObserverImpl observer;
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rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
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&observer, nullptr, &test_transport_,
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RtcpIntervalConfig{}));
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1000));
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rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
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rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
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EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
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@ -629,8 +628,7 @@ TEST_F(RtcpSenderTest, ByeMustBeLast) {
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// Re-configure rtcp_sender_ with mock_transport_
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rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
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nullptr, nullptr, &mock_transport,
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RtcpIntervalConfig{}));
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nullptr, nullptr, &mock_transport, 1000));
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rtcp_sender_->SetSSRC(kSenderSsrc);
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rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
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rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
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