diff options
Diffstat (limited to 'third_party/libwebrtc/moz-patch-stack/0103.patch')
| -rw-r--r-- | third_party/libwebrtc/moz-patch-stack/0103.patch | 134 |
1 files changed, 30 insertions, 104 deletions
diff --git a/third_party/libwebrtc/moz-patch-stack/0103.patch b/third_party/libwebrtc/moz-patch-stack/0103.patch index d232dcb897..a6da56ee72 100644 --- a/third_party/libwebrtc/moz-patch-stack/0103.patch +++ b/third_party/libwebrtc/moz-patch-stack/0103.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); |