Merge mozilla-central to autoland on a CLOSED TREE

CLOBBER

@@ -23,4 +23,4 @@
 # don't change CLOBBER for WebIDL changes any more.
 
 
-Merge day clobber 2023-07-31
+Modified build files in third_party/libwebrtc - Bug 1843113 - Vendor libwebrtc from 8b5bf6dd05

dom/media/systemservices/video_engine/desktop_capture_impl.cc

@@ -358,9 +358,6 @@ static DesktopCaptureOptions CreateDesktopCaptureOptions() {
 #if defined(WEBRTC_WIN)
   if (mozilla::StaticPrefs::media_webrtc_capture_allow_directx()) {
     options.set_allow_directx_capturer(true);
-    options.set_allow_use_magnification_api(false);
-  } else {
-    options.set_allow_use_magnification_api(true);
   }
   options.set_allow_cropping_window_capturer(true);
 #  if defined(RTC_ENABLE_WIN_WGC)

dom/media/webrtc/jsapi/RTCEncodedVideoFrame.cpp

@@ -49,28 +49,28 @@ RTCEncodedVideoFrame::RTCEncodedVideoFrame(
   mType = videoFrame.IsKeyFrame() ? RTCEncodedVideoFrameType::Key
                                   : RTCEncodedVideoFrameType::Delta;
 
-  if (videoFrame.GetMetadata().GetFrameId().has_value()) {
-    mMetadata.mFrameId.Construct(*videoFrame.GetMetadata().GetFrameId());
+  if (videoFrame.Metadata().GetFrameId().has_value()) {
+    mMetadata.mFrameId.Construct(*videoFrame.Metadata().GetFrameId());
   }
   mMetadata.mDependencies.Construct();
-  for (const auto dep : videoFrame.GetMetadata().GetFrameDependencies()) {
+  for (const auto dep : videoFrame.Metadata().GetFrameDependencies()) {
     Unused << mMetadata.mDependencies.Value().AppendElement(
         static_cast<unsigned long long>(dep), fallible);
   }
-  mMetadata.mWidth.Construct(videoFrame.GetMetadata().GetWidth());
-  mMetadata.mHeight.Construct(videoFrame.GetMetadata().GetHeight());
-  if (videoFrame.GetMetadata().GetSpatialIndex() >= 0) {
+  mMetadata.mWidth.Construct(videoFrame.Metadata().GetWidth());
+  mMetadata.mHeight.Construct(videoFrame.Metadata().GetHeight());
+  if (videoFrame.Metadata().GetSpatialIndex() >= 0) {
     mMetadata.mSpatialIndex.Construct(
-        videoFrame.GetMetadata().GetSpatialIndex());
+        videoFrame.Metadata().GetSpatialIndex());
   }
-  if (videoFrame.GetMetadata().GetTemporalIndex() >= 0) {
+  if (videoFrame.Metadata().GetTemporalIndex() >= 0) {
     mMetadata.mTemporalIndex.Construct(
-        videoFrame.GetMetadata().GetTemporalIndex());
+        videoFrame.Metadata().GetTemporalIndex());
   }
   mMetadata.mSynchronizationSource.Construct(videoFrame.GetSsrc());
   mMetadata.mPayloadType.Construct(videoFrame.GetPayloadType());
   mMetadata.mContributingSources.Construct();
-  for (const auto csrc : videoFrame.GetMetadata().GetCsrcs()) {
+  for (const auto csrc : videoFrame.Metadata().GetCsrcs()) {
     Unused << mMetadata.mContributingSources.Value().AppendElement(csrc,
                                                                    fallible);
   }

dom/media/webrtc/jsapi/RTCRtpReceiver.cpp

@@ -378,9 +378,9 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal(
               constructCommonInboundRtpStats(local);
               local.mJitter.Construct(audioStats->jitter_ms / 1000.0);
               local.mPacketsLost.Construct(audioStats->packets_lost);
-              local.mPacketsReceived.Construct(audioStats->packets_rcvd);
+              local.mPacketsReceived.Construct(audioStats->packets_received);
               local.mPacketsDiscarded.Construct(audioStats->packets_discarded);
-              local.mBytesReceived.Construct(audioStats->payload_bytes_rcvd);
+              local.mBytesReceived.Construct(audioStats->payload_bytes_received);
               // Always missing from libwebrtc stats
               // if (audioStats->estimated_playout_ntp_timestamp_ms) {
               //   local.mEstimatedPlayoutTimestamp.Construct(
@@ -409,7 +409,7 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal(
                         .ToDom());
               }
               local.mHeaderBytesReceived.Construct(
-                  audioStats->header_and_padding_bytes_rcvd);
+                  audioStats->header_and_padding_bytes_received);
               local.mFecPacketsReceived.Construct(
                   audioStats->fec_packets_received);
               local.mFecPacketsDiscarded.Construct(

dom/media/webrtc/jsapi/RTCRtpScriptTransformer.cpp

@@ -316,7 +316,7 @@ void RTCRtpScriptTransformer::TransformFrame(
         ResolveGenerateKeyFramePromises(videoFrame->GetRid(),
                                         videoFrame->GetTimestamp());
         if (!videoFrame->GetRid().empty() &&
-            videoFrame->GetMetadata().GetSimulcastIdx() == 0) {
+            videoFrame->Metadata().GetSimulcastIdx() == 0) {
           ResolveGenerateKeyFramePromises("", videoFrame->GetTimestamp());
         }
       }

dom/media/webrtc/libwebrtcglue/AudioConduit.cpp

@@ -621,6 +621,32 @@ void WebrtcAudioConduit::OnRtpReceived(webrtc::RtpPacketReceived&& aPacket,
                 aPacket.SequenceNumber(), aPacket.size(), aPacket.Ssrc(),
                 aPacket.Ssrc());
 
+  // Libwebrtc commit cde4b67d9d now expect calls to
+  // SourceTracker::GetSources() to happen on the call thread.  We'll
+  // grab the value now while on the call thread, and dispatch to main
+  // to store the cached value if we have new source information.
+  // See Bug 1845621.
+  std::vector<webrtc::RtpSource> sources;
+  if (mRecvStream) {
+    sources = mRecvStream->GetSources();
+  }
+
+  bool needsCacheUpdate = false;
+  {
+    AutoReadLock lock(mLock);
+    needsCacheUpdate = sources != mRtpSources;
+  }
+
+  // only dispatch to main if we have new data
+  if (needsCacheUpdate) {
+    GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+        __func__, [this, rtpSources = std::move(sources),
+                   self = RefPtr<WebrtcAudioConduit>(this)]() {
+          AutoWriteLock lock(mLock);
+          mRtpSources = rtpSources;
+        }));
+  }
+
   mRtpPacketEvent.Notify();
   if (mCall->Call()) {
     mCall->Call()->Receiver()->DeliverRtpPacket(
@@ -819,14 +845,7 @@ bool WebrtcAudioConduit::IsSamplingFreqSupported(int freq) const {
 std::vector<webrtc::RtpSource> WebrtcAudioConduit::GetUpstreamRtpSources()
     const {
   MOZ_ASSERT(NS_IsMainThread());
-  std::vector<webrtc::RtpSource> sources;
-  {
-    AutoReadLock lock(mLock);
-    if (mRecvStream) {
-      sources = mRecvStream->GetSources();
-    }
-  }
-  return sources;
+  return mRtpSources;
 }
 
 /* Return block-length of 10 ms audio frame in number of samples */

dom/media/webrtc/libwebrtcglue/AudioConduit.h

@@ -289,6 +289,10 @@ class WebrtcAudioConduit : public AudioSessionConduit,
   // To track changes needed to mRtpSendBaseSeqs.
   std::map<uint32_t, uint16_t> mRtpSendBaseSeqs_n;
 
+  // Written only on the main thread.  Guarded by mLock, except for
+  // reads on the main thread.
+  std::vector<webrtc::RtpSource> mRtpSources;
+
   // Thread safe
   Atomic<bool> mTransportActive = Atomic<bool>(false);
   MediaEventProducer<void> mRtcpByeEvent;

dom/media/webrtc/libwebrtcglue/VideoConduit.cpp

@@ -1535,6 +1535,32 @@ void WebrtcVideoConduit::OnRtpReceived(webrtc::RtpPacketReceived&& aPacket,
                 aPacket.SequenceNumber(), aPacket.size(), aPacket.Ssrc(),
                 aPacket.Ssrc());
 
+  // Libwebrtc commit cde4b67d9d now expect calls to
+  // SourceTracker::GetSources() to happen on the call thread.  We'll
+  // grab the value now while on the call thread, and dispatch to main
+  // to store the cached value if we have new source information.
+  // See Bug 1845621.
+  std::vector<webrtc::RtpSource> sources;
+  if (mRecvStream) {
+    sources = mRecvStream->GetSources();
+  }
+
+  bool needsCacheUpdate = false;
+  {
+    MutexAutoLock lock(mMutex);
+    needsCacheUpdate = sources != mRtpSources;
+  }
+
+  // only dispatch to main if we have new data
+  if (needsCacheUpdate) {
+    GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+        __func__, [this, rtpSources = std::move(sources),
+                   self = RefPtr<WebrtcVideoConduit>(this)]() {
+          MutexAutoLock lock(mMutex);
+          mRtpSources = rtpSources;
+        }));
+  }
+
   mRtpPacketEvent.Notify();
   if (mCall->Call()) {
     mCall->Call()->Receiver()->DeliverRtpPacket(
@@ -1928,14 +1954,7 @@ void WebrtcVideoConduit::SetTransportActive(bool aActive) {
 std::vector<webrtc::RtpSource> WebrtcVideoConduit::GetUpstreamRtpSources()
     const {
   MOZ_ASSERT(NS_IsMainThread());
-  std::vector<webrtc::RtpSource> sources;
-  {
-    MutexAutoLock lock(mMutex);
-    if (mRecvStream) {
-      sources = mRecvStream->GetSources();
-    }
-  }
-  return sources;
+  return mRtpSources;
 }
 
 void WebrtcVideoConduit::RequestKeyFrame(FrameTransformerProxy* aProxy) {

dom/media/webrtc/libwebrtcglue/VideoConduit.h

@@ -487,6 +487,10 @@ class WebrtcVideoConduit
   // Protected by mRendererMonitor
   dom::RTCVideoFrameHistoryInternal mReceivedFrameHistory;
 
+  // Written only on the main thread.  Guarded by mMutex, except for
+  // reads on the main thread.
+  std::vector<webrtc::RtpSource> mRtpSources;
+
   // Thread safe
   Atomic<bool> mTransportActive = Atomic<bool>(false);
   MediaEventProducer<void> mRtcpByeEvent;

dom/media/webrtc/third_party_build/default_config_env

@@ -9,34 +9,34 @@ export MOZ_LIBWEBRTC_SRC=$STATE_DIR/moz-libwebrtc
 # the commit summary as each upstream commit is vendored into the
 # mercurial repository.  The bug used for the v106 fast-forward was
 # 1800920.
-export MOZ_FASTFORWARD_BUG="1839451"
+export MOZ_FASTFORWARD_BUG="1843113"
 
 # MOZ_NEXT_LIBWEBRTC_MILESTONE and MOZ_NEXT_FIREFOX_REL_TARGET are
 # not used during fast-forward processing, but facilitate generating this
 # default config.  To generate an default config for the next update, run
 # bash dom/media/webrtc/third_party_build/update_default_config_env.sh
-export MOZ_NEXT_LIBWEBRTC_MILESTONE=113
-export MOZ_NEXT_FIREFOX_REL_TARGET=117
+export MOZ_NEXT_LIBWEBRTC_MILESTONE=114
+export MOZ_NEXT_FIREFOX_REL_TARGET=118
 
 # For Chromium release branches, see:
 # https://chromiumdash.appspot.com/branches
 
-# Chromium's v112 release branch was 5615.  This is used to pre-stack
+# Chromium's v113 release branch was 5672.  This is used to pre-stack
 # the previous release branch's commits onto the appropriate base commit
 # (the first common commit between trunk and the release branch).
-export MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM="5615"
+export MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM="5672"
 
-# New target release branch for v113 is branch-heads/5672.  This is used
+# New target release branch for v114 is branch-heads/5735.  This is used
 # to calculate the next upstream commit.
-export MOZ_TARGET_UPSTREAM_BRANCH_HEAD="branch-heads/5672"
+export MOZ_TARGET_UPSTREAM_BRANCH_HEAD="branch-heads/5735"
 
 # For local development 'mozpatches' is fine for a branch name, but when
 # pushing the patch stack to github, it should be named something like
-# 'moz-mods-chr113-for-rel117'.
+# 'moz-mods-chr114-for-rel118'.
 export MOZ_LIBWEBRTC_BRANCH="mozpatches"
 
 # After elm has been merged to mozilla-central, the patch stack in
 # moz-libwebrtc should be pushed to github.  The script
 # push_official_branch.sh uses this branch name when pushing to the
 # public repo.
-export MOZ_LIBWEBRTC_OFFICIAL_BRANCH="moz-mods-chr113-for-rel117"
+export MOZ_LIBWEBRTC_OFFICIAL_BRANCH="moz-mods-chr114-for-rel118"

third_party/libwebrtc/AUTHORS

@@ -133,6 +133,7 @@ Yura Yaroshevich <[email protected]>
 Yuriy Pavlyshak <[email protected]>
 Yusuke Suzuki <[email protected]>
 Pengfei Han <[email protected]>
+Yingying Ma <[email protected]>
 # END individuals section.
 
 # BEGIN organizations section.

third_party/libwebrtc/BUILD.gn

@@ -503,6 +503,16 @@ if (!build_with_chromium) {
       "api/task_queue",
       "api/task_queue:default_task_queue_factory",
       "api/test/metrics",
+      "api/video_codecs:video_decoder_factory_template",
+      "api/video_codecs:video_decoder_factory_template_dav1d_adapter",
+      "api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
+      "api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
+      "api/video_codecs:video_decoder_factory_template_open_h264_adapter",
+      "api/video_codecs:video_encoder_factory_template",
+      "api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
+      "api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
+      "api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
+      "api/video_codecs:video_encoder_factory_template_open_h264_adapter",
       "audio",
       "call",
       "common_audio",
@@ -527,6 +537,16 @@ if (!build_with_chromium) {
         "api/task_queue",
         "api/task_queue:default_task_queue_factory",
         "api/test/metrics",
+        "api/video_codecs:video_decoder_factory_template",
+        "api/video_codecs:video_decoder_factory_template_dav1d_adapter",
+        "api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
+        "api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
+        "api/video_codecs:video_decoder_factory_template_open_h264_adapter",
+        "api/video_codecs:video_encoder_factory_template",
+        "api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
+        "api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
+        "api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
+        "api/video_codecs:video_encoder_factory_template_open_h264_adapter",
         "logging:rtc_event_log_api",
         "p2p:rtc_p2p",
         "pc:libjingle_peerconnection",
@@ -542,13 +562,6 @@ if (!build_with_chromium) {
       ]
     }
 
-    if (rtc_include_builtin_video_codecs) {
-      deps += [
-        "api/video_codecs:builtin_video_decoder_factory",
-        "api/video_codecs:builtin_video_encoder_factory",
-      ]
-    }
-
     if (build_with_mozilla) {
       deps += [
         "api/video:video_frame",