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From: Michael Froman <mfroman@mozilla.com>
Date: Mon, 18 Dec 2023 15:00:00 +0000
Subject: Bug 1867099 - revert libwebrtc 8602f604e0. r=bwc
Upstream 8602f604e0 removed code sending BYEs which breaks some of
our wpt. They've opened a bug for a real fix here:
https://bugs.chromium.org/p/webrtc/issues/detail?id=15664
I've opened Bug 1870643 to track the real fix and upstream bug.
Differential Revision: https://phabricator.services.mozilla.com/D196729
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d92a578327f524ec3e1c144c82492a4c76b8266f
---
call/rtp_video_sender.cc | 1 +
modules/rtp_rtcp/source/rtcp_sender.cc | 19 +++++++++++++++++--
.../rtp_rtcp/source/rtcp_sender_unittest.cc | 5 +++--
modules/rtp_rtcp/source/rtp_rtcp_interface.h | 2 +-
4 files changed, 22 insertions(+), 5 deletions(-)
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 25cc81396c..4fa0e8c5e7 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -534,6 +534,7 @@ void RtpVideoSender::SetModuleIsActive(bool sending,
return;
}
+ // Sends a kRtcpByeCode when going from true to false.
rtp_module.SetSendingStatus(sending);
rtp_module.SetSendingMediaStatus(sending);
if (sending) {
diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
index 2b936a151e..43d9fc784d 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -193,8 +193,23 @@ bool RTCPSender::Sending() const {
void RTCPSender::SetSendingStatus(const FeedbackState& /* feedback_state */,
bool sending) {
- MutexLock lock(&mutex_rtcp_sender_);
- sending_ = sending;
+ bool sendRTCPBye = false;
+ {
+ MutexLock lock(&mutex_rtcp_sender_);
+
+ if (method_ != RtcpMode::kOff) {
+ if (sending == false && sending_ == true) {
+ // Trigger RTCP bye
+ sendRTCPBye = true;
+ }
+ }
+ sending_ = sending;
+ }
+ if (sendRTCPBye) {
+ if (SendRTCP(feedback_state, kRtcpBye) != 0) {
+ RTC_LOG(LS_WARNING) << "Failed to send RTCP BYE";
+ }
+ }
}
void RTCPSender::SetNonSenderRttMeasurement(bool enabled) {
diff --git a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index 35a7e96156..fa63ae090f 100644
--- a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -339,12 +339,13 @@ TEST_F(RtcpSenderTest, SendBye) {
EXPECT_EQ(kSenderSsrc, parser()->bye()->sender_ssrc());
}
-TEST_F(RtcpSenderTest, StopSendingDoesNotTriggersBye) {
+TEST_F(RtcpSenderTest, StopSendingTriggersBye) {
auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender->SetSendingStatus(feedback_state(), true);
rtcp_sender->SetSendingStatus(feedback_state(), false);
- EXPECT_EQ(0, parser()->bye()->num_packets());
+ EXPECT_EQ(1, parser()->bye()->num_packets());
+ EXPECT_EQ(kSenderSsrc, parser()->bye()->sender_ssrc());
}
TEST_F(RtcpSenderTest, SendFir) {
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index 5c6a9fcb85..13e7765cea 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -273,7 +273,7 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
// Returns the FlexFEC SSRC, if there is one.
virtual std::optional<uint32_t> FlexfecSsrc() const = 0;
- // Sets sending status.
+ // Sets sending status. Sends kRtcpByeCode when going from true to false.
// Returns -1 on failure else 0.
virtual int32_t SetSendingStatus(bool sending) = 0;
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