Merging upstream version 127.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 30 insertions, 104 deletions

diff --git a/third_party/libwebrtc/moz-patch-stack/0102.patch b/third_party/libwebrtc/moz-patch-stack/0102.patch
index d232dcb897..a6da56ee72 100644
--- a/third_party/libwebrtc/moz-patch-stack/0102.patch
+++ b/third_party/libwebrtc/moz-patch-stack/0102.patch
@@ -1,107 +1,33 @@
-From: Michael Froman <mfroman@mozilla.com>
-Date: Mon, 18 Dec 2023 15:00:00 +0000
-Subject: Bug 1867099 - revert libwebrtc 8602f604e0. r=bwc
+From: Andreas Pehrson <apehrson@mozilla.com>
+Date: Fri, 2 Feb 2024 18:43:00 +0000
+Subject: Bug 1878010 - Fix webrtc::VideoCaptureFactory for BSD.
+ r=grulja,gaston,webrtc-reviewers,mjf
 
-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
+Differential Revision: https://phabricator.services.mozilla.com/D200427
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/f890637efe5abc0020fab83ff2224313cd0c8460
 ---
- 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_impl.cc      |  1 +
- modules/rtp_rtcp/source/rtp_rtcp_interface.h  |  2 +-
- 5 files changed, 23 insertions(+), 5 deletions(-)
+ modules/video_capture/video_capture_factory.cc | 4 ++--
+ 1 file changed, 2 insertions(+), 2 deletions(-)
 
-diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
-index 1ace08fa32..4d99c61bb4 100644
---- a/call/rtp_video_sender.cc
-+++ b/call/rtp_video_sender.cc
-@@ -510,6 +510,7 @@ void RtpVideoSender::SetActiveModulesLocked(
-     const bool was_active = rtp_module.Sending();
-     const bool should_be_active = active_modules[i];
- 
-+    // Sends a kRtcpByeCode when going from true to false.
-     rtp_module.SetSendingStatus(active_modules[i]);
- 
-     if (was_active && !should_be_active) {
-diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
-index 099b0be1a3..971f49b949 100644
---- a/modules/rtp_rtcp/source/rtcp_sender.cc
-+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
-@@ -212,8 +212,23 @@ 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 002a5f86f1..1dcb628722 100644
---- a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
-+++ b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
-@@ -328,12 +328,13 @@ 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_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
-index cca9a40250..a63067141d 100644
---- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
-+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
-@@ -296,6 +296,7 @@ RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() {
- 
- int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
-   if (rtcp_sender_.Sending() != sending) {
-+    // Sends RTCP BYE when going from true to false
-     rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending);
-   }
-   return 0;
-diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
-index f196d11b58..bc8da63ab6 100644
---- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
-+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
-@@ -277,7 +277,7 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
-   // Returns the FlexFEC SSRC, if there is one.
-   virtual absl::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;
- 
+diff --git a/modules/video_capture/video_capture_factory.cc b/modules/video_capture/video_capture_factory.cc
+index e085ac2df8..2790fbbe1c 100644
+--- a/modules/video_capture/video_capture_factory.cc
++++ b/modules/video_capture/video_capture_factory.cc
+@@ -24,7 +24,7 @@ rtc::scoped_refptr<VideoCaptureModule> VideoCaptureFactory::Create(
+     const char* deviceUniqueIdUTF8) {
+ // This is only implemented on pure Linux and WEBRTC_LINUX is defined for
+ // Android as well
+-#if !defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID)
++#if (!defined(WEBRTC_LINUX) && !defined(WEBRTC_BSD)) || defined(WEBRTC_ANDROID)
+   return nullptr;
+ #else
+   return videocapturemodule::VideoCaptureImpl::Create(options,
+@@ -40,7 +40,7 @@ VideoCaptureModule::DeviceInfo* VideoCaptureFactory::CreateDeviceInfo(
+     VideoCaptureOptions* options) {
+ // This is only implemented on pure Linux and WEBRTC_LINUX is defined for
+ // Android as well
+-#if !defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID)
++#if (!defined(WEBRTC_LINUX) && !defined(WEBRTC_BSD)) || defined(WEBRTC_ANDROID)
+   return nullptr;
+ #else
+   return videocapturemodule::VideoCaptureImpl::CreateDeviceInfo(options);