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Reland "Add ability to set RTCP sender ssrc at construction time"

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This reverts commit 8b3e4e2d1166464f6b309f4fc533a29607d2771f.

Reason for revert: The culprit was https://webrtc-review.googlesource.com/c/src/+/133169.

Original change's description:
> Revert "Reland "Add ability to set RTCP sender ssrc at construction time""
> 
> This reverts commit 6f420e424885dab1d9f885365ea9abea5cc4a901.
> 
> Reason for revert: Speculative revert (some perf test are failing)
> 
> Original change's description:
> > Reland "Add ability to set RTCP sender ssrc at construction time"
> >
> > This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
> >
> > Patch set 1 is the original CL.
> > Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
> > if either current SSRC is 0 or if the SSRC is identical to the current
> > one. If so, don't schedule an early report.
> > This prevents a regression in which audio jitter became too low(?)
> >
> > Original change's description:
> > > Add ability to set RTCP sender ssrc at construction time
> > >
> > > Bug: webrtc:10774
> > > Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> > > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#28506}
> >
> > Bug: webrtc:10774
> > Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28520}
> 
> TBR=asapersson@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:10774
> Change-Id: I39238d942b2bbe0a9c8ca752387a35ed9dd70650
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145327
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28555}

TBR=mbonadei@webrtc.org,ilnik@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org

Change-Id: I2e5c17e8edfd938424f901222158643baa04866e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145400
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28562}
This commit is contained in:
Mirko Bonadei
2019-07-12 17:32:28 +00:00
committed by Commit Bot
parent 24192c267a
commit 9b1f24d552
6 changed files with 112 additions and 64 deletions
Download Patch File Download Diff File Expand all files Collapse all files

98
modules/rtp_rtcp/source/rtcp_sender_unittest.cc
View File

@ -75,22 +75,25 @@ class RtcpSenderTest : public ::testing::Test {
: clock_(1335900000),
receive_statistics_(ReceiveStatistics::Create(&clock_)),
retransmission_rate_limiter_(&clock_, 1000) {
RtpRtcp::Configuration configuration = GetDefaultConfig();
rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
rtcp_sender_.reset(new RTCPSender(configuration));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds(),
/*payload_type=*/0);
}
RtpRtcp::Configuration GetDefaultConfig() {
RtpRtcp::Configuration configuration;
configuration.audio = false;
configuration.clock = &clock_;
configuration.outgoing_transport = &test_transport_;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
configuration.rtcp_report_interval_ms = 1000;
rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
nullptr, nullptr, &test_transport_,
configuration.rtcp_report_interval_ms));
rtcp_sender_->SetSSRC(kSenderSsrc);
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds(),
/*payload_type=*/0);
configuration.receive_statistics = receive_statistics_.get();
configuration.media_send_ssrc = kSenderSsrc;
return configuration;
}
void InsertIncomingPacket(uint32_t remote_ssrc, uint16_t seq_num) {
@ -187,9 +190,13 @@ TEST_F(RtcpSenderTest, SendConsecutiveSrWithExactSlope) {
}
TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
nullptr, nullptr, &test_transport_, 1000));
rtcp_sender_->SetSSRC(kSenderSsrc);
RtpRtcp::Configuration config;
config.clock = &clock_;
config.receive_statistics = receive_statistics_.get();
config.outgoing_transport = &test_transport_;
config.rtcp_report_interval_ms = 1000;
config.media_send_ssrc = kSenderSsrc;
rtcp_sender_.reset(new RTCPSender(config));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender_->SetSendingStatus(feedback_state(), true);
@ -205,9 +212,13 @@ TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
}
TEST_F(RtcpSenderTest, DoNotSendCompundBeforeRtp) {
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
nullptr, nullptr, &test_transport_, 1000));
rtcp_sender_->SetSSRC(kSenderSsrc);
RtpRtcp::Configuration config;
config.clock = &clock_;
config.receive_statistics = receive_statistics_.get();
config.outgoing_transport = &test_transport_;
config.rtcp_report_interval_ms = 1000;
config.media_send_ssrc = kSenderSsrc;
rtcp_sender_.reset(new RTCPSender(config));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
rtcp_sender_->SetSendingStatus(feedback_state(), true);
@ -551,9 +562,14 @@ TEST_F(RtcpSenderTest, TestNoXrRrtrSentIfNotEnabled) {
TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
RtcpPacketTypeCounterObserverImpl observer;
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
&observer, nullptr, &test_transport_,
1000));
RtpRtcp::Configuration config;
config.clock = &clock_;
config.receive_statistics = receive_statistics_.get();
config.outgoing_transport = &test_transport_;
config.rtcp_packet_type_counter_observer = &observer;
config.rtcp_report_interval_ms = 1000;
rtcp_sender_.reset(new RTCPSender(config));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
@ -674,9 +690,14 @@ TEST_F(RtcpSenderTest, ByeMustBeLast) {
}));
// Re-configure rtcp_sender_ with mock_transport_
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
nullptr, nullptr, &mock_transport, 1000));
rtcp_sender_->SetSSRC(kSenderSsrc);
RtpRtcp::Configuration config;
config.clock = &clock_;
config.receive_statistics = receive_statistics_.get();
config.outgoing_transport = &mock_transport;
config.rtcp_report_interval_ms = 1000;
config.media_send_ssrc = kSenderSsrc;
rtcp_sender_.reset(new RTCPSender(config));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds(),
@ -795,4 +816,37 @@ TEST_F(RtcpSenderTest, SendTargetBitrateExplicitZeroOnStreamRemoval) {
EXPECT_EQ(bitrates[1].target_bitrate_kbps, 0u);
}
TEST_F(RtcpSenderTest, DoesntSchedulesInitialReportWhenSsrcSetOnConstruction) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
// New report should not have been scheduled yet.
clock_.AdvanceTimeMilliseconds(100);
EXPECT_FALSE(rtcp_sender_->TimeToSendRTCPReport(false));
}
TEST_F(RtcpSenderTest, DoesntSchedulesInitialReportOnFirstSetSsrc) {
// Set up without first SSRC not set at construction.
RtpRtcp::Configuration configuration = GetDefaultConfig();
configuration.media_send_ssrc = absl::nullopt;
rtcp_sender_.reset(new RTCPSender(configuration));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds(),
/*payload_type=*/0);
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
// Set SSRC for the first time. New report should not be scheduled.
rtcp_sender_->SetSSRC(kSenderSsrc);
clock_.AdvanceTimeMilliseconds(100);
EXPECT_FALSE(rtcp_sender_->TimeToSendRTCPReport(false));
}
TEST_F(RtcpSenderTest, SchedulesReportOnSsrcChange) {
rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender_->SetSSRC(kSenderSsrc + 1);
clock_.AdvanceTimeMilliseconds(100);
EXPECT_TRUE(rtcp_sender_->TimeToSendRTCPReport(false));
}
} // namespace webrtc