RTCRtpReceiver.h - mozsearch

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/* 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/RtpRtcpConfig.h"

#include "nsTArray.h"

#include "mozilla/dom/RTCRtpCapabilitiesBinding.h"

#include "mozilla/dom/RTCRtpParametersBinding.h"

#include "mozilla/dom/RTCStatsReportBinding.h"

#include "PerformanceRecorder.h"

#include "RTCStatsReport.h"

#include "transportbridge/MediaPipeline.h"

#include <vector>

class nsPIDOMWindowInner;

namespace mozilla {

class MediaSessionConduit;

class MediaTransportHandler;

class JsepTransceiver;

class PeerConnectionImpl;

enum class PrincipalPrivacy : uint8_t;

class RemoteTrackSource;

namespace dom {

class MediaStreamTrack;

class Promise;

class RTCDtlsTransport;

struct RTCRtpCapabilities;

struct RTCRtpContributingSource;

struct RTCRtpSynchronizationSource;

class RTCRtpTransceiver;

class RTCRtpScriptTransform;

class RTCStatsTimestampMaker;

class RTCRtpReceiver : public nsISupports,

                       public nsWrapperCache,

                       public MediaPipelineReceiveControlInterface {

 public:

  RTCRtpReceiver(nsPIDOMWindowInner* aWindow, PrincipalPrivacy aPrivacy,

                 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;

  static void GetCapabilities(const GlobalObject&, const nsAString& aKind,

                              Nullable<dom::RTCRtpCapabilities>& aResult);

  void GetParameters(RTCRtpReceiveParameters& aParameters) 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);

  RTCRtpScriptTransform* GetTransform() const { return mTransform; }

  void SetTransform(RTCRtpScriptTransform* aTransform, ErrorResult& aError);

  nsPIDOMWindowInner* GetParentObject() const;

  nsTArray<RefPtr<RTCStatsPromise>> GetStatsInternal(

      bool aSkipIceStats = false);

  Nullable<DOMHighResTimeStamp> GetJitterBufferTarget(

      ErrorResult& aError) const {

    return mJitterBufferTarget.isSome() ? Nullable(mJitterBufferTarget.value())

                                        : Nullable<DOMHighResTimeStamp>();

  void SetJitterBufferTarget(const Nullable<DOMHighResTimeStamp>& aTargetMs,

                             ErrorResult& aError);

  void Shutdown();

  void BreakCycles();

  void Unlink();

  // Terminal state, reached through stopping RTCRtpTransceiver.

  void Stop();

  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<RTCRtpReceiver>> mReceiversToMute;

    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);

  // Called when the privacy-needed state changes on the fly, as a result of

  // ALPN negotiation.

  void UpdatePrincipalPrivacy(PrincipalPrivacy aPrivacy);

  // Called by FrameTransformerProxy

  void RequestKeyFrame();

  void OnRtcpBye();

  void OnRtcpTimeout();

  void SetTrackMuteFromRemoteSdp();

  void OnRtpPacket();

  void UpdateUnmuteBlockingState();

  void UpdateReceiveTrackMute();

  Canonical<Ssrc>& CanonicalSsrc() { return mSsrc; }

  Canonical<Ssrc>& CanonicalVideoRtxSsrc() { return mVideoRtxSsrc; }

  Canonical<RtpExtList>& CanonicalLocalRtpExtensions() {

    return mLocalRtpExtensions;

  Canonical<std::vector<AudioCodecConfig>>& CanonicalAudioCodecs() {

    return mAudioCodecs;

  Canonical<std::vector<VideoCodecConfig>>& CanonicalVideoCodecs() {

    return mVideoCodecs;

  Canonical<Maybe<RtpRtcpConfig>>& CanonicalVideoRtpRtcpConfig() {

    return mVideoRtpRtcpConfig;

  Canonical<bool>& CanonicalReceiving() override { return mReceiving; }

  Canonical<RefPtr<FrameTransformerProxy>>& CanonicalFrameTransformerProxy() {

    return mFrameTransformerProxy;

  const RTCStatsTimestampMaker* GetTimestampMaker() const;

 private:

  virtual ~RTCRtpReceiver();

  void UpdateVideoConduit();

  void UpdateAudioConduit();

  std::string GetMid() const;

  JsepTransceiver& GetJsepTransceiver();

  const JsepTransceiver& GetJsepTransceiver() const;

  WatchManager<RTCRtpReceiver> mWatchManager;

  nsCOMPtr<nsPIDOMWindowInner> mWindow;

  RefPtr<PeerConnectionImpl> mPc;

  bool mHaveStartedReceiving = false;

  bool mHaveSetupTransport = false;

  RefPtr<AbstractThread> mCallThread;

  nsCOMPtr<nsISerialEventTarget> mStsThread;

  RefPtr<dom::MediaStreamTrack> mTrack;

  RefPtr<RemoteTrackSource> mTrackSource;

  RefPtr<MediaPipelineReceive> mPipeline;

  RefPtr<MediaTransportHandler> mTransportHandler;

  RefPtr<RTCRtpTransceiver> mTransceiver;

  RefPtr<RTCRtpScriptTransform> mTransform;

  // 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;

  Watchable<bool> mReceiveTrackMute{true, "RTCRtpReceiver::mReceiveTrackMute"};

  // This corresponds to the [[Receptive]] slot on RTCRtpTransceiver.

  // Its only purpose is suppressing unmute events if true.

  bool mReceptive = false;

  // This is the [[JitterBufferTarget]] internal slot.

  Maybe<DOMHighResTimeStamp> mJitterBufferTarget;

  // Houses [[ReceiveCodecs]]

  RTCRtpReceiveParameters mParameters;

  MediaEventListener mRtcpByeListener;

  MediaEventListener mRtcpTimeoutListener;

  MediaEventListener mUnmuteListener;

  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;

  Canonical<RefPtr<FrameTransformerProxy>> mFrameTransformerProxy;

};

}  // namespace dom

}  // namespace mozilla

#endif  // _RTCRtpReceiver_h_