1 files changed, 170 insertions, 0 deletions
diff --git a/dom/media/webrtc/jsapi/RTCRtpReceiver.h b/dom/media/webrtc/jsapi/RTCRtpReceiver.h
new file mode 100644
index 0000000000..c6e90bcf28
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpReceiver.h
@@ -0,0 +1,170 @@
+/* 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 _RTCRtpReceiver_h_
+#define _RTCRtpReceiver_h_
+
+#include "nsISupports.h"
+#include "nsWrapperCache.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/StateMirroring.h"
+#include "mozilla/Maybe.h"
+#include "js/RootingAPI.h"
+#include "libwebrtcglue/MediaConduitInterface.h"
+#include "libwebrtcglue/RtpRtcpConfig.h"
+#include "nsTArray.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "PerformanceRecorder.h"
+#include "RTCStatsReport.h"
+#include <vector>
+
+class nsPIDOMWindowInner;
+
+namespace mozilla {
+class MediaPipelineReceive;
+class MediaSessionConduit;
+class MediaTransportHandler;
+class JsepTransceiver;
+class PeerConnectionImpl;
+
+namespace dom {
+class MediaStreamTrack;
+class Promise;
+class RTCDtlsTransport;
+struct RTCRtpContributingSource;
+struct RTCRtpSynchronizationSource;
+class RTCRtpTransceiver;
+
+class RTCRtpReceiver : public nsISupports, public nsWrapperCache {
+ public:
+  RTCRtpReceiver(nsPIDOMWindowInner* aWindow, bool aPrivacyNeeded,
+                 PeerConnectionImpl* aPc,
+                 MediaTransportHandler* aTransportHandler,
+                 AbstractThread* aCallThread, nsISerialEventTarget* aStsThread,
+                 MediaSessionConduit* aConduit, RTCRtpTransceiver* aTransceiver,
+                 const TrackingId& aTrackingId);
+
+  // nsISupports
+  NS_DECL_CYCLE_COLLECTING_ISUPPORTS
+  NS_DECL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpReceiver)
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  // webidl
+  MediaStreamTrack* Track() const { return mTrack; }
+  RTCDtlsTransport* GetTransport() const;
+  already_AddRefed<Promise> GetStats(ErrorResult& aError);
+  void GetContributingSources(
+      nsTArray<dom::RTCRtpContributingSource>& aSources);
+  void GetSynchronizationSources(
+      nsTArray<dom::RTCRtpSynchronizationSource>& aSources);
+  // test-only: insert fake CSRCs and audio levels for testing
+  void MozInsertAudioLevelForContributingSource(
+      const uint32_t aSource, const DOMHighResTimeStamp aTimestamp,
+      const uint32_t aRtpTimestamp, const bool aHasLevel, const uint8_t aLevel);
+
+  nsPIDOMWindowInner* GetParentObject() const;
+  nsTArray<RefPtr<RTCStatsPromise>> GetStatsInternal(
+      bool aSkipIceStats = false);
+
+  void Shutdown();
+  void BreakCycles();
+  void Stop();
+  void Start();
+  bool HasTrack(const dom::MediaStreamTrack* aTrack) const;
+  void SyncToJsep(JsepTransceiver& aJsepTransceiver) const;
+  void SyncFromJsep(const JsepTransceiver& aJsepTransceiver);
+  const std::vector<std::string>& GetStreamIds() const { return mStreamIds; }
+
+  struct StreamAssociation {
+    RefPtr<MediaStreamTrack> mTrack;
+    std::string mStreamId;
+  };
+
+  struct TrackEventInfo {
+    RefPtr<RTCRtpReceiver> mReceiver;
+    std::vector<std::string> mStreamIds;
+  };
+
+  struct StreamAssociationChanges {
+    std::vector<RefPtr<MediaStreamTrack>> mTracksToMute;
+    std::vector<StreamAssociation> mStreamAssociationsRemoved;
+    std::vector<StreamAssociation> mStreamAssociationsAdded;
+    std::vector<TrackEventInfo> mTrackEvents;
+  };
+
+  // This is called when we set an answer (ie; when the transport is finalized).
+  void UpdateTransport();
+  void UpdateConduit();
+
+  // This is called when we set a remote description; may be an offer or answer.
+  void UpdateStreams(StreamAssociationChanges* aChanges);
+
+  void OnRtcpBye();
+  void OnRtcpTimeout();
+
+  void SetReceiveTrackMuted(bool aMuted);
+
+  AbstractCanonical<Ssrc>* CanonicalSsrc() { return &mSsrc; }
+  AbstractCanonical<Ssrc>* CanonicalVideoRtxSsrc() { return &mVideoRtxSsrc; }
+  AbstractCanonical<RtpExtList>* CanonicalLocalRtpExtensions() {
+    return &mLocalRtpExtensions;
+  }
+
+  AbstractCanonical<std::vector<AudioCodecConfig>>* CanonicalAudioCodecs() {
+    return &mAudioCodecs;
+  }
+
+  AbstractCanonical<std::vector<VideoCodecConfig>>* CanonicalVideoCodecs() {
+    return &mVideoCodecs;
+  }
+  AbstractCanonical<Maybe<RtpRtcpConfig>>* CanonicalVideoRtpRtcpConfig() {
+    return &mVideoRtpRtcpConfig;
+  }
+  AbstractCanonical<bool>* CanonicalReceiving() { return &mReceiving; }
+
+ private:
+  virtual ~RTCRtpReceiver();
+
+  void UpdateVideoConduit();
+  void UpdateAudioConduit();
+
+  std::string GetMid() const;
+  JsepTransceiver& GetJsepTransceiver();
+  const JsepTransceiver& GetJsepTransceiver() const;
+
+  nsCOMPtr<nsPIDOMWindowInner> mWindow;
+  RefPtr<PeerConnectionImpl> mPc;
+  bool mHaveStartedReceiving = false;
+  bool mHaveSetupTransport = false;
+  RefPtr<AbstractThread> mCallThread;
+  nsCOMPtr<nsISerialEventTarget> mStsThread;
+  RefPtr<dom::MediaStreamTrack> mTrack;
+  RefPtr<MediaPipelineReceive> mPipeline;
+  RefPtr<MediaTransportHandler> mTransportHandler;
+  RefPtr<RTCRtpTransceiver> mTransceiver;
+  // This is [[AssociatedRemoteMediaStreams]], basically. We do not keep the
+  // streams themselves here, because that would require this object to know
+  // where the stream list for the whole RTCPeerConnection lives..
+  std::vector<std::string> mStreamIds;
+  bool mRemoteSetSendBit = false;
+
+  MediaEventListener mRtcpByeListener;
+  MediaEventListener mRtcpTimeoutListener;
+
+  Canonical<Ssrc> mSsrc;
+  Canonical<Ssrc> mVideoRtxSsrc;
+  Canonical<RtpExtList> mLocalRtpExtensions;
+
+  Canonical<std::vector<AudioCodecConfig>> mAudioCodecs;
+
+  Canonical<std::vector<VideoCodecConfig>> mVideoCodecs;
+  Canonical<Maybe<RtpRtcpConfig>> mVideoRtpRtcpConfig;
+  Canonical<bool> mReceiving;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+#endif  // _RTCRtpReceiver_h_