RTCRtpSender.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 _RTCRtpSender_h_

#define _RTCRtpSender_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/RTCStatsReportBinding.h"

#include "mozilla/dom/RTCRtpCapabilitiesBinding.h"

#include "mozilla/dom/RTCRtpParametersBinding.h"

#include "RTCStatsReport.h"

#include "jsep/JsepTrack.h"

#include "transportbridge/MediaPipeline.h"

class nsPIDOMWindowInner;

namespace mozilla {

class MediaSessionConduit;

class MediaTransportHandler;

class JsepTransceiver;

class PeerConnectionImpl;

class DOMMediaStream;

namespace dom {

class MediaStreamTrack;

class Promise;

class RTCDtlsTransport;

class RTCDTMFSender;

struct RTCRtpCapabilities;

class RTCRtpTransceiver;

class RTCRtpScriptTransform;

class RTCRtpSender : public nsISupports,

                     public nsWrapperCache,

                     public MediaPipelineTransmitControlInterface {

 public:

  RTCRtpSender(nsPIDOMWindowInner* aWindow, PeerConnectionImpl* aPc,

               MediaTransportHandler* aTransportHandler,

               AbstractThread* aCallThread, nsISerialEventTarget* aStsThread,

               MediaSessionConduit* aConduit, dom::MediaStreamTrack* aTrack,

               const Sequence<RTCRtpEncodingParameters>& aEncodings,

               RTCRtpTransceiver* aTransceiver);

  // nsISupports

  NS_DECL_CYCLE_COLLECTING_ISUPPORTS

  NS_DECL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpSender)

  JSObject* WrapObject(JSContext* aCx,

                       JS::Handle<JSObject*> aGivenProto) override;

  // webidl

  MediaStreamTrack* GetTrack() const { return mSenderTrack; }

  RTCDtlsTransport* GetTransport() const;

  RTCDTMFSender* GetDtmf() const;

  MOZ_CAN_RUN_SCRIPT

  already_AddRefed<Promise> ReplaceTrack(MediaStreamTrack* aWithTrack,

                                         ErrorResult& aError);

  already_AddRefed<Promise> GetStats(ErrorResult& aError);

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

                              Nullable<dom::RTCRtpCapabilities>& result);

  already_AddRefed<Promise> SetParameters(

      const dom::RTCRtpSendParameters& aParameters, ErrorResult& aError);

  // Not a simple getter, so not const

  // See https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-getparameters

  void GetParameters(RTCRtpSendParameters& aParameters);

  static void CheckAndRectifyEncodings(

      Sequence<RTCRtpEncodingParameters>& aEncodings, bool aVideo,

      ErrorResult& aRv);

  RTCRtpScriptTransform* GetTransform() const { return mTransform; }

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

  bool GenerateKeyFrame(const Maybe<std::string>& aRid);

  nsPIDOMWindowInner* GetParentObject() const;

  nsTArray<RefPtr<RTCStatsPromise>> GetStatsInternal(

      bool aSkipIceStats = false);

  void SetStreams(const Sequence<OwningNonNull<DOMMediaStream>>& aStreams,

                  ErrorResult& aRv);

  // ChromeOnly webidl

  void GetStreams(nsTArray<RefPtr<DOMMediaStream>>& aStreams);

  // ChromeOnly webidl

  void SetStreamsImpl(const Sequence<OwningNonNull<DOMMediaStream>>& aStreams);

  // ChromeOnly webidl

  void SetTrack(const RefPtr<MediaStreamTrack>& aTrack);

  void Shutdown();

  void BreakCycles();

  void Unlink();

  // Terminal state, reached through stopping RTCRtpTransceiver.

  void Stop();

  bool HasTrack(const dom::MediaStreamTrack* aTrack) const;

  bool IsMyPc(const PeerConnectionImpl* aPc) const { return mPc.get() == aPc; }

  RefPtr<MediaPipelineTransmit> GetPipeline() const;

  already_AddRefed<dom::Promise> MakePromise(ErrorResult& aError) const;

  bool SeamlessTrackSwitch(const RefPtr<MediaStreamTrack>& aWithTrack);

  bool SetSenderTrackWithClosedCheck(const RefPtr<MediaStreamTrack>& aTrack);

  // This is called when we set an answer (ie; when the transport is finalized).

  void UpdateTransport();

  void SyncToJsep(JsepTransceiver& aJsepTransceiver) const;

  void SyncFromJsep(const JsepTransceiver& aJsepTransceiver);

  void MaybeUpdateConduit();

  Canonical<Ssrcs>& CanonicalSsrcs() { return mSsrcs; }

  Canonical<Ssrcs>& CanonicalVideoRtxSsrcs() { return mVideoRtxSsrcs; }

  Canonical<RtpExtList>& CanonicalLocalRtpExtensions() {

    return mLocalRtpExtensions;

  Canonical<Maybe<AudioCodecConfig>>& CanonicalAudioCodec() {

    return mAudioCodec;

  Canonical<Maybe<VideoCodecConfig>>& CanonicalVideoCodec() {

    return mVideoCodec;

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

    return mVideoRtpRtcpConfig;

  Canonical<webrtc::VideoCodecMode>& CanonicalVideoCodecMode() {

    return mVideoCodecMode;

  Canonical<std::string>& CanonicalCname() { return mCname; }

  Canonical<bool>& CanonicalTransmitting() override { return mTransmitting; }

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

    return mFrameTransformerProxy;

  bool HasPendingSetParameters() const { return mPendingParameters.isSome(); }

  void InvalidateLastReturnedParameters() {

    mLastReturnedParameters = Nothing();

 private:

  virtual ~RTCRtpSender();

  std::string GetMid() const;

  JsepTransceiver& GetJsepTransceiver();

  static void ApplyJsEncodingToConduitEncoding(

      const RTCRtpEncodingParameters& aJsEncoding,

      VideoCodecConfig::Encoding* aConduitEncoding);

  void UpdateRestorableEncodings(

      const Sequence<RTCRtpEncodingParameters>& aEncodings);

  Sequence<RTCRtpEncodingParameters> GetMatchingEncodings(

      const std::vector<std::string>& aRids) const;

  Sequence<RTCRtpEncodingParameters> ToSendEncodings(

      const std::vector<std::string>& aRids) const;

  void MaybeGetJsepRids();

  void UpdateDtmfSender();

  void WarnAboutBadSetParameters(const nsCString& aError);

  nsCString GetEffectiveTLDPlus1() const;

  WatchManager<RTCRtpSender> mWatchManager;

  nsCOMPtr<nsPIDOMWindowInner> mWindow;

  RefPtr<PeerConnectionImpl> mPc;

  RefPtr<dom::MediaStreamTrack> mSenderTrack;

  bool mSenderTrackSetByAddTrack = false;

  RTCRtpSendParameters mParameters;

  Maybe<RTCRtpSendParameters> mPendingParameters;

  uint32_t mNumSetParametersCalls = 0;

  // When JSEP goes from simulcast to unicast without a rid, and we started out

  // as unicast without a rid, we are supposed to restore that unicast encoding

  // from before.

  Maybe<RTCRtpEncodingParameters> mUnicastEncoding;

  bool mSimulcastEnvelopeSet = false;

  bool mSimulcastEnvelopeSetByJSEP = false;

  bool mPendingRidChangeFromCompatMode = false;

  Maybe<RTCRtpSendParameters> mLastReturnedParameters;

  RefPtr<MediaPipelineTransmit> mPipeline;

  RefPtr<MediaTransportHandler> mTransportHandler;

  RefPtr<RTCRtpTransceiver> mTransceiver;

  nsTArray<RefPtr<DOMMediaStream>> mStreams;

  RefPtr<RTCRtpScriptTransform> mTransform;

  bool mHaveSetupTransport = false;

  // TODO(bug 1803388): Remove this stuff once it is no longer needed.

  bool mAllowOldSetParameters = false;

  // TODO(bug 1803388): Remove the glean warnings once they are no longer needed

  bool mHaveWarnedBecauseNoGetParameters = false;

  bool mHaveWarnedBecauseEncodingCountChange = false;

  bool mHaveWarnedBecauseNoTransactionId = false;

  // TODO(bug 1803389): Remove the glean errors once they are no longer needed.

  bool mHaveFailedBecauseNoGetParameters = false;

  bool mHaveFailedBecauseEncodingCountChange = false;

  bool mHaveFailedBecauseRidChange = false;

  bool mHaveFailedBecauseNoTransactionId = false;

  bool mHaveFailedBecauseStaleTransactionId = false;

  bool mHaveFailedBecauseNoEncodings = false;

  bool mHaveFailedBecauseOtherError = false;

  // Limits logging of codec information

  bool mHaveLoggedUlpfecInfo = false;

  bool mHaveLoggedOtherFec = false;

  bool mHaveLoggedVideoPreferredCodec = false;

  bool mHaveLoggedAudioPreferredCodec = false;

  RefPtr<dom::RTCDTMFSender> mDtmf;

  class BaseConfig {

   public:

    // TODO(bug 1744116): Use = default here

    bool operator==(const BaseConfig& aOther) const {

      return mSsrcs == aOther.mSsrcs &&

             mLocalRtpExtensions == aOther.mLocalRtpExtensions &&

             mCname == aOther.mCname && mTransmitting == aOther.mTransmitting;

    Ssrcs mSsrcs;

    RtpExtList mLocalRtpExtensions;

    std::string mCname;

    bool mTransmitting = false;

};

  class VideoConfig : public BaseConfig {

   public:

    // TODO(bug 1744116): Use = default here

    bool operator==(const VideoConfig& aOther) const {

      return BaseConfig::operator==(aOther) &&

             mVideoRtxSsrcs == aOther.mVideoRtxSsrcs &&

             mVideoCodec == aOther.mVideoCodec &&

             mVideoRtpRtcpConfig == aOther.mVideoRtpRtcpConfig &&

             mVideoCodecMode == aOther.mVideoCodecMode;

    Ssrcs mVideoRtxSsrcs;

    Maybe<VideoCodecConfig> mVideoCodec;

    Maybe<RtpRtcpConfig> mVideoRtpRtcpConfig;

    webrtc::VideoCodecMode mVideoCodecMode =

        webrtc::VideoCodecMode::kRealtimeVideo;

};

  class AudioConfig : public BaseConfig {

   public:

    // TODO(bug 1744116): Use = default here

    bool operator==(const AudioConfig& aOther) const {

      return BaseConfig::operator==(aOther) &&

             mAudioCodec == aOther.mAudioCodec && mDtmfPt == aOther.mDtmfPt &&

             mDtmfFreq == aOther.mDtmfFreq;

    Maybe<AudioCodecConfig> mAudioCodec;

    int32_t mDtmfPt = -1;

    int32_t mDtmfFreq = 0;

};

  Maybe<VideoConfig> GetNewVideoConfig();

  Maybe<AudioConfig> GetNewAudioConfig();

  void UpdateBaseConfig(BaseConfig* aConfig);

  void ApplyVideoConfig(const VideoConfig& aConfig);

  void ApplyAudioConfig(const AudioConfig& aConfig);

  Canonical<Ssrcs> mSsrcs;

  Canonical<Ssrcs> mVideoRtxSsrcs;

  Canonical<RtpExtList> mLocalRtpExtensions;

  Canonical<Maybe<AudioCodecConfig>> mAudioCodec;

  Canonical<Maybe<VideoCodecConfig>> mVideoCodec;

  Canonical<Maybe<RtpRtcpConfig>> mVideoRtpRtcpConfig;

  Canonical<webrtc::VideoCodecMode> mVideoCodecMode;

  Canonical<std::string> mCname;

  Canonical<bool> mTransmitting;

  Canonical<RefPtr<FrameTransformerProxy>> mFrameTransformerProxy;

};

}  // namespace dom

}  // namespace mozilla

#endif  // _RTCRtpSender_h_