Adding upstream version 115.8.0esr.upstream/115.8.0esr

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

1 files changed, 194 insertions, 0 deletions

diff --git a/dom/media/webrtc/jsapi/PeerConnectionCtx.h b/dom/media/webrtc/jsapi/PeerConnectionCtx.h
new file mode 100644
index 0000000000..fdd81f6406
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PeerConnectionCtx.h
@@ -0,0 +1,194 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef peerconnectionctx_h___h__
+#define peerconnectionctx_h___h__
+
+#include <map>
+#include <string>
+
+#include "WebrtcGlobalChild.h"
+#include "api/field_trials_view.h"
+#include "api/scoped_refptr.h"
+#include "call/audio_state.h"
+#include "MediaTransportHandler.h"  // Mostly for IceLogPromise
+#include "mozIGeckoMediaPluginService.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/StaticPtr.h"
+#include "nsIRunnable.h"
+#include "PeerConnectionImpl.h"
+
+namespace webrtc {
+class AudioDecoderFactory;
+
+// Used for testing in mediapipeline_unittest.cpp, MockCall.h
+class NoTrialsConfig : public FieldTrialsView {
+ public:
+  NoTrialsConfig() = default;
+  std::string Lookup(absl::string_view key) const override {
+    // Upstream added a new default field trial string for
+    // CongestionWindow, that we don't want.  In
+    // third_party/libwebrtc/rtc_base/experiments/rate_control_settings.cc
+    // they set kCongestionWindowDefaultFieldTrialString to
+    // "QueueSize:350,MinBitrate:30000,DropFrame:true". With QueueSize
+    // set, GoogCcNetworkController::UpdateCongestionWindowSize is
+    // called.  Because negative values are calculated in
+    // feedback_rtt, an assert fires when calculating data_window in
+    // GoogCcNetworkController::UpdateCongestionWindowSize.  We probably
+    // need to figure out why we're calculating negative feedback_rtt.
+    // See Bug 1780620.
+    if ("WebRTC-CongestionWindow" == key) {
+      return std::string("MinBitrate:30000,DropFrame:true");
+    }
+    return std::string();
+  }
+};
+}  // namespace webrtc
+
+namespace mozilla {
+class PeerConnectionCtxObserver;
+
+namespace dom {
+class WebrtcGlobalInformation;
+}
+
+/**
+ * Refcounted class containing state shared across all PeerConnections and all
+ * Call instances. Managed by PeerConnectionCtx, and kept around while there are
+ * registered peer connections.
+ */
+class SharedWebrtcState {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(SharedWebrtcState)
+
+  SharedWebrtcState(RefPtr<AbstractThread> aCallWorkerThread,
+                    webrtc::AudioState::Config&& aAudioStateConfig,
+                    RefPtr<webrtc::AudioDecoderFactory> aAudioDecoderFactory,
+                    UniquePtr<webrtc::FieldTrialsView> aTrials);
+
+  // A global Call worker thread shared between all Call instances. Implements
+  // AbstractThread for running tasks that call into a Call instance through its
+  // webrtc::TaskQueue member, and for using AbstractThread-specific higher
+  // order constructs like StateMirroring.
+  const RefPtr<AbstractThread> mCallWorkerThread;
+
+  // AudioState config containing dummy implementations of the audio stack,
+  // since we use our own audio stack instead. Shared across all Call instances.
+  const webrtc::AudioState::Config mAudioStateConfig;
+
+  // AudioDecoderFactory instance shared between calls, to limit the number of
+  // instances in large calls.
+  const RefPtr<webrtc::AudioDecoderFactory> mAudioDecoderFactory;
+
+  // Trials instance shared between calls, to limit the number of instances in
+  // large calls.
+  const UniquePtr<webrtc::FieldTrialsView> mTrials;
+
+ private:
+  virtual ~SharedWebrtcState();
+};
+
+// A class to hold some of the singleton objects we need:
+// * The global PeerConnectionImpl table and its associated lock.
+// * Stats report objects for PCs that are gone
+// * GMP related state
+// * Upstream webrtc state shared across all Calls (processing thread)
+class PeerConnectionCtx {
+ public:
+  static nsresult InitializeGlobal();
+  static PeerConnectionCtx* GetInstance();
+  static bool isActive();
+  static void Destroy();
+
+  bool isReady() {
+    // If mGMPService is not set, we aren't using GMP.
+    if (mGMPService) {
+      return mGMPReady;
+    }
+    return true;
+  }
+
+  void queueJSEPOperation(nsIRunnable* aJSEPOperation);
+  void onGMPReady();
+
+  bool gmpHasH264();
+
+  static void UpdateNetworkState(bool online);
+
+  RefPtr<MediaTransportHandler> GetTransportHandler() const {
+    return mTransportHandler;
+  }
+
+  SharedWebrtcState* GetSharedWebrtcState() const;
+
+  void RemovePeerConnection(const std::string& aKey);
+  void AddPeerConnection(const std::string& aKey,
+                         PeerConnectionImpl* aPeerConnection);
+  PeerConnectionImpl* GetPeerConnection(const std::string& aKey) const;
+  template <typename Function>
+  void ForEachPeerConnection(Function&& aFunction) const {
+    MOZ_ASSERT(NS_IsMainThread());
+    for (const auto& pair : mPeerConnections) {
+      aFunction(pair.second);
+    }
+  }
+
+  void ClearClosedStats();
+
+ private:
+  std::map<const std::string, PeerConnectionImpl*> mPeerConnections;
+
+  PeerConnectionCtx()
+      : mGMPReady(false),
+        mTransportHandler(
+            MediaTransportHandler::Create(GetMainThreadSerialEventTarget())) {}
+
+  // This is a singleton, so don't copy construct it, etc.
+  PeerConnectionCtx(const PeerConnectionCtx& other) = delete;
+  void operator=(const PeerConnectionCtx& other) = delete;
+  virtual ~PeerConnectionCtx() = default;
+
+  nsresult Initialize();
+  nsresult StartTelemetryTimer();
+  void StopTelemetryTimer();
+  nsresult Cleanup();
+
+  void initGMP();
+
+  static void EverySecondTelemetryCallback_m(nsITimer* timer, void*);
+
+  nsCOMPtr<nsITimer> mTelemetryTimer;
+
+ private:
+  void DeliverStats(UniquePtr<dom::RTCStatsReportInternal>&& aReport);
+
+  std::map<nsString, UniquePtr<dom::RTCStatsReportInternal>> mLastReports;
+  // We cannot form offers/answers properly until the Gecko Media Plugin stuff
+  // has been initted, which is a complicated mess of thread dispatches,
+  // including sync dispatches to main. So, we need to be able to queue up
+  // offer creation (or SetRemote, when we're the answerer) until all of this is
+  // ready to go, since blocking on this init is just begging for deadlock.
+  nsCOMPtr<mozIGeckoMediaPluginService> mGMPService;
+  bool mGMPReady;
+  nsTArray<nsCOMPtr<nsIRunnable>> mQueuedJSEPOperations;
+
+  // Not initted, just for ICE logging stuff
+  RefPtr<MediaTransportHandler> mTransportHandler;
+
+  // State used by libwebrtc that needs to be shared across all PeerConnections
+  // and all Call instances. Set while there is at least one peer connection
+  // registered. CallWrappers can hold a ref to this object to be sure members
+  // are alive long enough.
+  RefPtr<SharedWebrtcState> mSharedWebrtcState;
+
+  static PeerConnectionCtx* gInstance;
+
+ public:
+  static mozilla::StaticRefPtr<mozilla::PeerConnectionCtxObserver>
+      gPeerConnectionCtxObserver;
+};
+
+}  // namespace mozilla
+
+#endif