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-rw-r--r--third_party/libwebrtc/moz-patch-stack/0103.patch134
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);