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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 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);
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();
// 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();
AbstractCanonical<Ssrcs>* CanonicalSsrcs() { return &mSsrcs; }
AbstractCanonical<Ssrcs>* CanonicalVideoRtxSsrcs() { return &mVideoRtxSsrcs; }
AbstractCanonical<RtpExtList>* CanonicalLocalRtpExtensions() {
return &mLocalRtpExtensions;
}
AbstractCanonical<Maybe<AudioCodecConfig>>* CanonicalAudioCodec() {
return &mAudioCodec;
}
AbstractCanonical<Maybe<VideoCodecConfig>>* CanonicalVideoCodec() {
return &mVideoCodec;
}
AbstractCanonical<Maybe<RtpRtcpConfig>>* CanonicalVideoRtpRtcpConfig() {
return &mVideoRtpRtcpConfig;
}
AbstractCanonical<webrtc::VideoCodecMode>* CanonicalVideoCodecMode() {
return &mVideoCodecMode;
}
AbstractCanonical<std::string>* CanonicalCname() { return &mCname; }
AbstractCanonical<bool>* CanonicalTransmitting() override {
return &mTransmitting;
}
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 mAddTrackCalled = 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;
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;
bool mHaveWarnedBecauseStaleTransactionId = 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;
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;
};
} // namespace dom
} // namespace mozilla
#endif // _RTCRtpSender_h_
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