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;