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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/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 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);
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;
}
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;
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_
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