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Diffstat (limited to '')
| -rw-r--r-- | dom/media/webrtc/jsapi/RTCRtpTransceiver.cpp | 1080 |
1 files changed, 1080 insertions, 0 deletions
diff --git a/dom/media/webrtc/jsapi/RTCRtpTransceiver.cpp b/dom/media/webrtc/jsapi/RTCRtpTransceiver.cpp new file mode 100644 index 0000000000..bf4f74dd5d --- /dev/null +++ b/dom/media/webrtc/jsapi/RTCRtpTransceiver.cpp @@ -0,0 +1,1080 @@ +/* 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/. */ + +#include "jsapi/RTCRtpTransceiver.h" +#include "mozilla/UniquePtr.h" +#include <algorithm> +#include <string> +#include <vector> +#include "libwebrtcglue/AudioConduit.h" +#include "libwebrtcglue/VideoConduit.h" +#include "MediaTrackGraph.h" +#include "transportbridge/MediaPipeline.h" +#include "transportbridge/MediaPipelineFilter.h" +#include "jsep/JsepTrack.h" +#include "sdp/SdpHelper.h" +#include "MediaTrackGraphImpl.h" +#include "transport/logging.h" +#include "MediaEngine.h" +#include "nsIPrincipal.h" +#include "MediaSegment.h" +#include "RemoteTrackSource.h" +#include "libwebrtcglue/RtpRtcpConfig.h" +#include "MediaTransportHandler.h" +#include "mozilla/dom/RTCRtpReceiverBinding.h" +#include "mozilla/dom/RTCRtpSenderBinding.h" +#include "mozilla/dom/RTCRtpTransceiverBinding.h" +#include "mozilla/dom/Promise.h" +#include "RTCDtlsTransport.h" +#include "RTCRtpReceiver.h" +#include "RTCRtpSender.h" +#include "RTCDTMFSender.h" +#include "systemservices/MediaUtils.h" +#include "libwebrtcglue/WebrtcCallWrapper.h" +#include "libwebrtcglue/WebrtcGmpVideoCodec.h" +#include "utils/PerformanceRecorder.h" + +namespace mozilla { + +using namespace dom; + +namespace { +struct ConduitControlState : public AudioConduitControlInterface, + public VideoConduitControlInterface { + ConduitControlState(RTCRtpTransceiver* aTransceiver, RTCRtpSender* aSender, + RTCRtpReceiver* aReceiver) + : mTransceiver(new nsMainThreadPtrHolder<RTCRtpTransceiver>( + "ConduitControlState::mTransceiver", aTransceiver, false)), + mSender(new nsMainThreadPtrHolder<dom::RTCRtpSender>( + "ConduitControlState::mSender", aSender, false)), + mReceiver(new nsMainThreadPtrHolder<dom::RTCRtpReceiver>( + "ConduitControlState::mReceiver", aReceiver, false)) {} + + const nsMainThreadPtrHandle<RTCRtpTransceiver> mTransceiver; + const nsMainThreadPtrHandle<RTCRtpSender> mSender; + const nsMainThreadPtrHandle<RTCRtpReceiver> mReceiver; + + // MediaConduitControlInterface + AbstractCanonical<bool>* CanonicalReceiving() override { + return mReceiver->CanonicalReceiving(); + } + AbstractCanonical<bool>* CanonicalTransmitting() override { + return mSender->CanonicalTransmitting(); + } + AbstractCanonical<Ssrcs>* CanonicalLocalSsrcs() override { + return mSender->CanonicalSsrcs(); + } + AbstractCanonical<std::string>* CanonicalLocalCname() override { + return mSender->CanonicalCname(); + } + AbstractCanonical<std::string>* CanonicalMid() override { + return mTransceiver->CanonicalMid(); + } + AbstractCanonical<Ssrc>* CanonicalRemoteSsrc() override { + return mReceiver->CanonicalSsrc(); + } + AbstractCanonical<std::string>* CanonicalSyncGroup() override { + return mTransceiver->CanonicalSyncGroup(); + } + AbstractCanonical<RtpExtList>* CanonicalLocalRecvRtpExtensions() override { + return mReceiver->CanonicalLocalRtpExtensions(); + } + AbstractCanonical<RtpExtList>* CanonicalLocalSendRtpExtensions() override { + return mSender->CanonicalLocalRtpExtensions(); + } + + // AudioConduitControlInterface + AbstractCanonical<Maybe<AudioCodecConfig>>* CanonicalAudioSendCodec() + override { + return mSender->CanonicalAudioCodec(); + } + AbstractCanonical<std::vector<AudioCodecConfig>>* CanonicalAudioRecvCodecs() + override { + return mReceiver->CanonicalAudioCodecs(); + } + MediaEventSource<DtmfEvent>& OnDtmfEvent() override { + return mSender->GetDtmf()->OnDtmfEvent(); + } + + // VideoConduitControlInterface + AbstractCanonical<Ssrcs>* CanonicalLocalVideoRtxSsrcs() override { + return mSender->CanonicalVideoRtxSsrcs(); + } + AbstractCanonical<Ssrc>* CanonicalRemoteVideoRtxSsrc() override { + return mReceiver->CanonicalVideoRtxSsrc(); + } + AbstractCanonical<Maybe<VideoCodecConfig>>* CanonicalVideoSendCodec() + override { + return mSender->CanonicalVideoCodec(); + } + AbstractCanonical<Maybe<RtpRtcpConfig>>* CanonicalVideoSendRtpRtcpConfig() + override { + return mSender->CanonicalVideoRtpRtcpConfig(); + } + AbstractCanonical<std::vector<VideoCodecConfig>>* CanonicalVideoRecvCodecs() + override { + return mReceiver->CanonicalVideoCodecs(); + } + AbstractCanonical<Maybe<RtpRtcpConfig>>* CanonicalVideoRecvRtpRtcpConfig() + override { + return mReceiver->CanonicalVideoRtpRtcpConfig(); + } + AbstractCanonical<webrtc::VideoCodecMode>* CanonicalVideoCodecMode() + override { + return mSender->CanonicalVideoCodecMode(); + } +}; +} // namespace + +MOZ_MTLOG_MODULE("RTCRtpTransceiver") + +NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpTransceiver) +NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(RTCRtpTransceiver) + if (tmp->mHandlingUnlink) { + tmp->BreakCycles(); + tmp->mHandlingUnlink = false; + } + NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER +NS_IMPL_CYCLE_COLLECTION_UNLINK_END +NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(RTCRtpTransceiver) + NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mWindow, mPc, mSendTrack, mReceiver, + mSender, mDtlsTransport, + mLastStableDtlsTransport) +NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END + +NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCRtpTransceiver) +NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCRtpTransceiver) +NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCRtpTransceiver) + NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY + NS_INTERFACE_MAP_ENTRY(nsISupports) +NS_INTERFACE_MAP_END + +#define INIT_CANONICAL(name, val) \ + name(AbstractThread::MainThread(), val, \ + "RTCRtpTransceiver::" #name " (Canonical)") + +RTCRtpTransceiver::RTCRtpTransceiver( + nsPIDOMWindowInner* aWindow, bool aPrivacyNeeded, PeerConnectionImpl* aPc, + MediaTransportHandler* aTransportHandler, JsepSession* aJsepSession, + const std::string& aTransceiverId, bool aIsVideo, + nsISerialEventTarget* aStsThread, dom::MediaStreamTrack* aSendTrack, + WebrtcCallWrapper* aCallWrapper, RTCStatsIdGenerator* aIdGenerator) + : mWindow(aWindow), + mPc(aPc), + mTransportHandler(aTransportHandler), + mTransceiverId(aTransceiverId), + mJsepTransceiver(*aJsepSession->GetTransceiver(mTransceiverId)), + mStsThread(aStsThread), + mCallWrapper(aCallWrapper), + mSendTrack(aSendTrack), + mIdGenerator(aIdGenerator), + mPrincipalPrivacy(aPrivacyNeeded ? PrincipalPrivacy::Private + : PrincipalPrivacy::NonPrivate), + mIsVideo(aIsVideo), + INIT_CANONICAL(mMid, std::string()), + INIT_CANONICAL(mSyncGroup, std::string()) {} + +#undef INIT_CANONICAL + +RTCRtpTransceiver::~RTCRtpTransceiver() = default; + +SdpDirectionAttribute::Direction ToSdpDirection( + RTCRtpTransceiverDirection aDirection) { + switch (aDirection) { + case dom::RTCRtpTransceiverDirection::Sendrecv: + return SdpDirectionAttribute::Direction::kSendrecv; + case dom::RTCRtpTransceiverDirection::Sendonly: + return SdpDirectionAttribute::Direction::kSendonly; + case dom::RTCRtpTransceiverDirection::Recvonly: + return SdpDirectionAttribute::Direction::kRecvonly; + case dom::RTCRtpTransceiverDirection::Inactive: + return SdpDirectionAttribute::Direction::kInactive; + case dom::RTCRtpTransceiverDirection::EndGuard_:; + } + MOZ_CRASH("Invalid transceiver direction!"); +} + +static uint32_t sRemoteSourceId = 0; + +// TODO(bug 1401592): Once we implement the sendEncodings stuff, there will +// need to be validation code in here. +void RTCRtpTransceiver::Init(const RTCRtpTransceiverInit& aInit, + ErrorResult& aRv) { + TrackingId trackingId(TrackingId::Source::RTCRtpReceiver, sRemoteSourceId++, + TrackingId::TrackAcrossProcesses::Yes); + if (IsVideo()) { + InitVideo(trackingId); + } else { + InitAudio(); + } + + if (!IsValid()) { + aRv = NS_ERROR_UNEXPECTED; + return; + } + + mReceiver = new RTCRtpReceiver(mWindow, mPrincipalPrivacy, mPc, + mTransportHandler, mCallWrapper->mCallThread, + mStsThread, mConduit, this, trackingId); + + mSender = new RTCRtpSender(mWindow, mPc, mTransportHandler, + mCallWrapper->mCallThread, mStsThread, mConduit, + mSendTrack, aInit.mSendEncodings, this); + + if (mConduit) { + InitConduitControl(); + } + + mSender->SetStreamsImpl(aInit.mStreams); + mDirection = aInit.mDirection; +} + +void RTCRtpTransceiver::SetDtlsTransport(dom::RTCDtlsTransport* aDtlsTransport, + bool aStable) { + mDtlsTransport = aDtlsTransport; + if (aStable) { + mLastStableDtlsTransport = mDtlsTransport; + } +} + +void RTCRtpTransceiver::RollbackToStableDtlsTransport() { + mDtlsTransport = mLastStableDtlsTransport; +} + +void RTCRtpTransceiver::InitAudio() { + mConduit = AudioSessionConduit::Create(mCallWrapper, mStsThread); + + if (!mConduit) { + MOZ_MTLOG(ML_ERROR, mPc->GetHandle() + << "[" << mMid.Ref() << "]: " << __FUNCTION__ + << ": Failed to create AudioSessionConduit"); + // TODO(bug 1422897): We need a way to record this when it happens in the + // wild. + } +} + +void RTCRtpTransceiver::InitVideo(const TrackingId& aRecvTrackingId) { + VideoSessionConduit::Options options; + options.mVideoLatencyTestEnable = + Preferences::GetBool("media.video.test_latency", false); + options.mMinBitrate = std::max( + 0, + Preferences::GetInt("media.peerconnection.video.min_bitrate", 0) * 1000); + options.mStartBitrate = std::max( + 0, Preferences::GetInt("media.peerconnection.video.start_bitrate", 0) * + 1000); + options.mPrefMaxBitrate = std::max( + 0, + Preferences::GetInt("media.peerconnection.video.max_bitrate", 0) * 1000); + if (options.mMinBitrate != 0 && + options.mMinBitrate < kViEMinCodecBitrate_bps) { + options.mMinBitrate = kViEMinCodecBitrate_bps; + } + if (options.mStartBitrate < options.mMinBitrate) { + options.mStartBitrate = options.mMinBitrate; + } + if (options.mPrefMaxBitrate && + options.mStartBitrate > options.mPrefMaxBitrate) { + options.mStartBitrate = options.mPrefMaxBitrate; + } + // XXX We'd love if this was a live param for testing adaptation/etc + // in automation + options.mMinBitrateEstimate = + std::max(0, Preferences::GetInt( + "media.peerconnection.video.min_bitrate_estimate", 0) * + 1000); + options.mSpatialLayers = std::max( + 1, Preferences::GetInt("media.peerconnection.video.svc.spatial", 0)); + options.mTemporalLayers = std::max( + 1, Preferences::GetInt("media.peerconnection.video.svc.temporal", 0)); + options.mDenoising = + Preferences::GetBool("media.peerconnection.video.denoising", false); + options.mLockScaling = + Preferences::GetBool("media.peerconnection.video.lock_scaling", false); + + mConduit = + VideoSessionConduit::Create(mCallWrapper, mStsThread, std::move(options), + mPc->GetHandle(), aRecvTrackingId); + + if (!mConduit) { + MOZ_MTLOG(ML_ERROR, mPc->GetHandle() + << "[" << mMid.Ref() << "]: " << __FUNCTION__ + << ": Failed to create VideoSessionConduit"); + // TODO(bug 1422897): We need a way to record this when it happens in the + // wild. + } +} + +void RTCRtpTransceiver::InitConduitControl() { + MOZ_ASSERT(NS_IsMainThread()); + MOZ_ASSERT(mConduit); + ConduitControlState control(this, mSender, mReceiver); + mCallWrapper->mCallThread->Dispatch(NS_NewRunnableFunction( + __func__, [conduit = mConduit, control = std::move(control)]() mutable { + conduit->AsVideoSessionConduit().apply( + [&](VideoSessionConduit* aConduit) { + aConduit->InitControl(&control); + }); + conduit->AsAudioSessionConduit().apply( + [&](AudioSessionConduit* aConduit) { + aConduit->InitControl(&control); + }); + })); +} + +void RTCRtpTransceiver::Close() { + // Called via PCImpl::Close -> PCImpl::CloseInt -> PCImpl::ShutdownMedia -> + // PCMedia::SelfDestruct. Satisfies step 7 of + // https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-close + mShutdown = true; + if (mDtlsTransport) { + mDtlsTransport->UpdateState(TransportLayer::TS_CLOSED); + } + StopImpl(); +} + +void RTCRtpTransceiver::BreakCycles() { + mSender->BreakCycles(); + mReceiver->BreakCycles(); + mWindow = nullptr; + mSendTrack = nullptr; + mSender = nullptr; + mReceiver = nullptr; + mDtlsTransport = nullptr; + mLastStableDtlsTransport = nullptr; + mPc = nullptr; +} + +// TODO: Only called from one place in PeerConnectionImpl, synchronously, when +// the JSEP engine has successfully completed an offer/answer exchange. This is +// a bit squirrely, since identity validation happens asynchronously in +// PeerConnection.jsm. This probably needs to happen once all the "in parallel" +// steps have succeeded, but before we queue the task for JS observable state +// updates. +nsresult RTCRtpTransceiver::UpdateTransport() { + if (!mHasTransport) { + return NS_OK; + } + + mReceiver->UpdateTransport(); + mSender->UpdateTransport(); + return NS_OK; +} + +nsresult RTCRtpTransceiver::UpdateConduit() { + if (mStopped) { + return NS_OK; + } + + mReceiver->UpdateConduit(); + mSender->MaybeUpdateConduit(); + + return NS_OK; +} + +void RTCRtpTransceiver::UpdatePrincipalPrivacy(PrincipalPrivacy aPrivacy) { + if (mPrincipalPrivacy == aPrivacy) { + return; + } + + mPrincipalPrivacy = aPrivacy; + mReceiver->UpdatePrincipalPrivacy(mPrincipalPrivacy); +} + +void RTCRtpTransceiver::ResetSync() { mSyncGroup = std::string(); } + +// TODO: Only called from one place in PeerConnectionImpl, synchronously, when +// the JSEP engine has successfully completed an offer/answer exchange. This is +// a bit squirrely, since identity validation happens asynchronously in +// PeerConnection.jsm. This probably needs to happen once all the "in parallel" +// steps have succeeded, but before we queue the task for JS observable state +// updates. +nsresult RTCRtpTransceiver::SyncWithMatchingVideoConduits( + nsTArray<RefPtr<RTCRtpTransceiver>>& transceivers) { + if (mStopped) { + return NS_OK; + } + + if (IsVideo()) { + MOZ_MTLOG(ML_ERROR, mPc->GetHandle() + << "[" << mMid.Ref() << "]: " << __FUNCTION__ + << " called when transceiver is not " + "video! This should never happen."); + MOZ_CRASH(); + return NS_ERROR_UNEXPECTED; + } + + std::set<std::string> myReceiveStreamIds; + myReceiveStreamIds.insert(mReceiver->GetStreamIds().begin(), + mReceiver->GetStreamIds().end()); + + for (RefPtr<RTCRtpTransceiver>& transceiver : transceivers) { + if (!transceiver->IsValid()) { + continue; + } + + if (!transceiver->IsVideo()) { + // |this| is an audio transceiver, so we skip other audio transceivers + continue; + } + + // Maybe could make this more efficient by cacheing this set, but probably + // not worth it. + for (const std::string& streamId : + transceiver->Receiver()->GetStreamIds()) { + if (myReceiveStreamIds.count(streamId)) { + // Ok, we have one video, one non-video - cross the streams! + mSyncGroup = streamId; + transceiver->mSyncGroup = streamId; + + MOZ_MTLOG(ML_DEBUG, mPc->GetHandle() + << "[" << mMid.Ref() << "]: " << __FUNCTION__ + << " Syncing " << mConduit.get() << " to " + << transceiver->mConduit.get()); + + // The sync code in call.cc only permits sync between audio stream and + // one video stream. They take the first match, so there's no point in + // continuing here. If we want to change the default, we should sort + // video streams here and only call SetSyncGroup on the chosen stream. + break; + } + } + } + + return NS_OK; +} + +bool RTCRtpTransceiver::ConduitHasPluginID(uint64_t aPluginID) { + return mConduit && mConduit->HasCodecPluginID(aPluginID); +} + +void RTCRtpTransceiver::SyncFromJsep(const JsepSession& aSession) { + MOZ_MTLOG(ML_DEBUG, mPc->GetHandle() + << "[" << mMid.Ref() << "]: " << __FUNCTION__ + << " Syncing from JSEP transceiver"); + if (mShutdown) { + // Shutdown_m has already been called, probably due to pc.close(). Just + // nod and smile. + return; + } + + mJsepTransceiver = *aSession.GetTransceiver(mTransceiverId); + + // Transceivers can stop due to JSEP negotiation, so we need to check that + if (mJsepTransceiver.IsStopped()) { + StopImpl(); + } + + mReceiver->SyncFromJsep(mJsepTransceiver); + mSender->SyncFromJsep(mJsepTransceiver); + + // mid from JSEP + if (mJsepTransceiver.IsAssociated()) { + mMid = mJsepTransceiver.GetMid(); + } else { + mMid = std::string(); + } + + // currentDirection from JSEP, but not if "this transceiver has never been + // represented in an offer/answer exchange" + if (mJsepTransceiver.HasLevel() && mJsepTransceiver.IsNegotiated()) { + if (mJsepTransceiver.mRecvTrack.GetActive()) { + if (mJsepTransceiver.mSendTrack.GetActive()) { + mCurrentDirection.SetValue(dom::RTCRtpTransceiverDirection::Sendrecv); + mHasBeenUsedToSend = true; + } else { + mCurrentDirection.SetValue(dom::RTCRtpTransceiverDirection::Recvonly); + } + } else { + if (mJsepTransceiver.mSendTrack.GetActive()) { + mCurrentDirection.SetValue(dom::RTCRtpTransceiverDirection::Sendonly); + mHasBeenUsedToSend = true; + } else { + mCurrentDirection.SetValue(dom::RTCRtpTransceiverDirection::Inactive); + } + } + } + + mShouldRemove = mJsepTransceiver.IsRemoved(); + mHasTransport = mJsepTransceiver.HasLevel() && !mJsepTransceiver.IsStopped(); +} + +void RTCRtpTransceiver::SyncToJsep(JsepSession& aSession) const { + MOZ_MTLOG(ML_DEBUG, mPc->GetHandle() + << "[" << mMid.Ref() << "]: " << __FUNCTION__ + << " Syncing to JSEP transceiver"); + + aSession.ApplyToTransceiver( + mTransceiverId, [this, self = RefPtr<const RTCRtpTransceiver>(this)]( + JsepTransceiver& aTransceiver) { + mReceiver->SyncToJsep(aTransceiver); + mSender->SyncToJsep(aTransceiver); + aTransceiver.mJsDirection = ToSdpDirection(mDirection); + if (mStopped) { + aTransceiver.Stop(); + } + }); +} + +void RTCRtpTransceiver::GetKind(nsAString& aKind) const { + // The transceiver kind of an RTCRtpTransceiver is defined by the kind of the + // associated RTCRtpReceiver's MediaStreamTrack object. + MOZ_ASSERT(mReceiver && mReceiver->Track()); + mReceiver->Track()->GetKind(aKind); +} + +void RTCRtpTransceiver::GetMid(nsAString& aMid) const { + if (!mMid.Ref().empty()) { + aMid = NS_ConvertUTF8toUTF16(mMid.Ref()); + } else { + aMid.SetIsVoid(true); + } +} + +std::string RTCRtpTransceiver::GetMidAscii() const { + if (mMid.Ref().empty()) { + return std::string(); + } + + return mMid.Ref(); +} + +void RTCRtpTransceiver::SetDirection(RTCRtpTransceiverDirection aDirection, + ErrorResult& aRv) { + if (mStopped) { + aRv.ThrowInvalidStateError("Transceiver is stopped!"); + return; + } + + if (aDirection == mDirection) { + return; + } + + SetDirectionInternal(aDirection); + + mPc->UpdateNegotiationNeeded(); +} + +void RTCRtpTransceiver::SetDirectionInternal( + RTCRtpTransceiverDirection aDirection) { + // We do not update the direction on the JsepTransceiver until sync + mDirection = aDirection; +} + +bool RTCRtpTransceiver::ShouldRemove() const { return mShouldRemove; } + +bool RTCRtpTransceiver::CanSendDTMF() const { + // Spec says: "If connection's RTCPeerConnectionState is not "connected" + // return false." We don't support that right now. This is supposed to be + // true once ICE is complete, and _all_ DTLS handshakes are also complete. We + // don't really have access to the state of _all_ of our DTLS states either. + // Our pipeline _does_ know whether SRTP/SRTCP is ready, which happens + // immediately after our transport finishes DTLS (unless there was an error), + // so this is pretty close. + // TODO (bug 1265827): Base this on RTCPeerConnectionState instead. + // TODO (bug 1623193): Tighten this up + if (!IsSending() || !mSender->GetTrack()) { + return false; + } + + // Ok, it looks like the connection is up and sending. Did we negotiate + // telephone-event? + const JsepTrackNegotiatedDetails* details = + mJsepTransceiver.mSendTrack.GetNegotiatedDetails(); + if (NS_WARN_IF(!details || !details->GetEncodingCount())) { + // What? + return false; + } + + for (size_t i = 0; i < details->GetEncodingCount(); ++i) { + const auto& encoding = details->GetEncoding(i); + for (const auto& codec : encoding.GetCodecs()) { + if (codec->mName == "telephone-event") { + return true; + } + } + } + + return false; +} + +JSObject* RTCRtpTransceiver::WrapObject(JSContext* aCx, + JS::Handle<JSObject*> aGivenProto) { + return dom::RTCRtpTransceiver_Binding::Wrap(aCx, this, aGivenProto); +} + +nsPIDOMWindowInner* RTCRtpTransceiver::GetParentObject() const { + return mWindow; +} + +static void JsepCodecDescToAudioCodecConfig( + const JsepAudioCodecDescription& aCodec, Maybe<AudioCodecConfig>* aConfig) { + uint16_t pt; + + // TODO(bug 1761272): Getting the pt for a JsepAudioCodecDescription should be + // infallible. + if (NS_WARN_IF(!aCodec.GetPtAsInt(&pt))) { + MOZ_MTLOG(ML_ERROR, "Invalid payload type: " << aCodec.mDefaultPt); + MOZ_ASSERT(false); + return; + } + + // libwebrtc crashes if we attempt to configure a mono recv codec + bool sendMono = aCodec.mForceMono && aCodec.mDirection == sdp::kSend; + + *aConfig = Some(AudioCodecConfig( + pt, aCodec.mName, static_cast<int>(aCodec.mClock), + sendMono ? 1 : static_cast<int>(aCodec.mChannels), aCodec.mFECEnabled)); + (*aConfig)->mMaxPlaybackRate = static_cast<int>(aCodec.mMaxPlaybackRate); + (*aConfig)->mDtmfEnabled = aCodec.mDtmfEnabled; + (*aConfig)->mDTXEnabled = aCodec.mDTXEnabled; + (*aConfig)->mMaxAverageBitrate = aCodec.mMaxAverageBitrate; + (*aConfig)->mFrameSizeMs = aCodec.mFrameSizeMs; + (*aConfig)->mMinFrameSizeMs = aCodec.mMinFrameSizeMs; + (*aConfig)->mMaxFrameSizeMs = aCodec.mMaxFrameSizeMs; + (*aConfig)->mCbrEnabled = aCodec.mCbrEnabled; +} + +// TODO: This and the next function probably should move to JsepTransceiver +Maybe<const std::vector<UniquePtr<JsepCodecDescription>>&> +RTCRtpTransceiver::GetNegotiatedSendCodecs() const { + if (!mJsepTransceiver.mSendTrack.GetActive()) { + return Nothing(); + } + + const auto* details = mJsepTransceiver.mSendTrack.GetNegotiatedDetails(); + if (!details) { + return Nothing(); + } + + if (details->GetEncodingCount() == 0) { + return Nothing(); + } + + return SomeRef(details->GetEncoding(0).GetCodecs()); +} + +Maybe<const std::vector<UniquePtr<JsepCodecDescription>>&> +RTCRtpTransceiver::GetNegotiatedRecvCodecs() const { + if (!mJsepTransceiver.mRecvTrack.GetActive()) { + return Nothing(); + } + + const auto* details = mJsepTransceiver.mRecvTrack.GetNegotiatedDetails(); + if (!details) { + return Nothing(); + } + + if (details->GetEncodingCount() == 0) { + return Nothing(); + } + + return SomeRef(details->GetEncoding(0).GetCodecs()); +} + +// TODO: Maybe move this someplace else? +/*static*/ +void RTCRtpTransceiver::NegotiatedDetailsToAudioCodecConfigs( + const JsepTrackNegotiatedDetails& aDetails, + std::vector<AudioCodecConfig>* aConfigs) { + Maybe<AudioCodecConfig> telephoneEvent; + + if (aDetails.GetEncodingCount()) { + for (const auto& codec : aDetails.GetEncoding(0).GetCodecs()) { + if (NS_WARN_IF(codec->Type() != SdpMediaSection::kAudio)) { + MOZ_ASSERT(false, "Codec is not audio! This is a JSEP bug."); + return; + } + Maybe<AudioCodecConfig> config; + const JsepAudioCodecDescription& audio = + static_cast<const JsepAudioCodecDescription&>(*codec); + JsepCodecDescToAudioCodecConfig(audio, &config); + if (config->mName == "telephone-event") { + telephoneEvent = std::move(config); + } else { + aConfigs->push_back(std::move(*config)); + } + } + } + + // Put telephone event at the back, because webrtc.org crashes if we don't + // If we need to do even more sorting, we should use std::sort. + if (telephoneEvent) { + aConfigs->push_back(std::move(*telephoneEvent)); + } +} + +auto RTCRtpTransceiver::GetActivePayloadTypes() const + -> RefPtr<ActivePayloadTypesPromise> { + if (!mConduit) { + return ActivePayloadTypesPromise::CreateAndResolve(PayloadTypes(), + __func__); + } + + if (!mCallWrapper) { + return ActivePayloadTypesPromise::CreateAndResolve(PayloadTypes(), + __func__); + } + + return InvokeAsync(mCallWrapper->mCallThread, __func__, + [conduit = mConduit]() { + PayloadTypes pts; + pts.mSendPayloadType = conduit->ActiveSendPayloadType(); + pts.mRecvPayloadType = conduit->ActiveRecvPayloadType(); + return ActivePayloadTypesPromise::CreateAndResolve( + std::move(pts), __func__); + }); +} + +static void JsepCodecDescToVideoCodecConfig( + const JsepVideoCodecDescription& aCodec, Maybe<VideoCodecConfig>* aConfig) { + uint16_t pt; + + // TODO(bug 1761272): Getting the pt for a JsepVideoCodecDescription should be + // infallible. + if (NS_WARN_IF(!aCodec.GetPtAsInt(&pt))) { + MOZ_MTLOG(ML_ERROR, "Invalid payload type: " << aCodec.mDefaultPt); + MOZ_ASSERT(false); + return; + } + + UniquePtr<VideoCodecConfigH264> h264Config; + + if (aCodec.mName == "H264") { + h264Config = MakeUnique<VideoCodecConfigH264>(); + size_t spropSize = sizeof(h264Config->sprop_parameter_sets); + strncpy(h264Config->sprop_parameter_sets, + aCodec.mSpropParameterSets.c_str(), spropSize); + h264Config->sprop_parameter_sets[spropSize - 1] = '\0'; + h264Config->packetization_mode = + static_cast<int>(aCodec.mPacketizationMode); + h264Config->profile_level_id = static_cast<int>(aCodec.mProfileLevelId); + h264Config->tias_bw = 0; // TODO(bug 1403206) + } + + *aConfig = Some(VideoCodecConfig(pt, aCodec.mName, aCodec.mConstraints, + h264Config.get())); + + (*aConfig)->mAckFbTypes = aCodec.mAckFbTypes; + (*aConfig)->mNackFbTypes = aCodec.mNackFbTypes; + (*aConfig)->mCcmFbTypes = aCodec.mCcmFbTypes; + (*aConfig)->mRembFbSet = aCodec.RtcpFbRembIsSet(); + (*aConfig)->mFECFbSet = aCodec.mFECEnabled; + (*aConfig)->mTransportCCFbSet = aCodec.RtcpFbTransportCCIsSet(); + if (aCodec.mFECEnabled) { + uint16_t pt; + if (SdpHelper::GetPtAsInt(aCodec.mREDPayloadType, &pt)) { + (*aConfig)->mREDPayloadType = pt; + } + if (SdpHelper::GetPtAsInt(aCodec.mULPFECPayloadType, &pt)) { + (*aConfig)->mULPFECPayloadType = pt; + } + } + if (aCodec.mRtxEnabled) { + uint16_t pt; + if (SdpHelper::GetPtAsInt(aCodec.mRtxPayloadType, &pt)) { + (*aConfig)->mRTXPayloadType = pt; + } + } +} + +// TODO: Maybe move this someplace else? +/*static*/ +void RTCRtpTransceiver::NegotiatedDetailsToVideoCodecConfigs( + const JsepTrackNegotiatedDetails& aDetails, + std::vector<VideoCodecConfig>* aConfigs) { + if (aDetails.GetEncodingCount()) { + for (const auto& codec : aDetails.GetEncoding(0).GetCodecs()) { + if (NS_WARN_IF(codec->Type() != SdpMediaSection::kVideo)) { + MOZ_ASSERT(false, "Codec is not video! This is a JSEP bug."); + return; + } + Maybe<VideoCodecConfig> config; + const JsepVideoCodecDescription& video = + static_cast<const JsepVideoCodecDescription&>(*codec); + + JsepCodecDescToVideoCodecConfig(video, &config); + + config->mTias = aDetails.GetTias(); + + for (size_t i = 0; i < aDetails.GetEncodingCount(); ++i) { + const JsepTrackEncoding& jsepEncoding(aDetails.GetEncoding(i)); + if (jsepEncoding.HasFormat(video.mDefaultPt)) { + VideoCodecConfig::Encoding encoding; + encoding.rid = jsepEncoding.mRid; + config->mEncodings.push_back(encoding); + } + } + + aConfigs->push_back(std::move(*config)); + } + } +} + +void RTCRtpTransceiver::Stop(ErrorResult& aRv) { + if (mPc->IsClosed()) { + aRv.ThrowInvalidStateError("Peer connection is closed"); + return; + } + + StopImpl(); + mPc->UpdateNegotiationNeeded(); +} + +void RTCRtpTransceiver::StopImpl() { + if (mStopped) { + return; + } + + if (mCallWrapper) { + auto conduit = std::move(mConduit); + (conduit ? conduit->Shutdown() + : GenericPromise::CreateAndResolve(true, __func__)) + ->Then(GetMainThreadSerialEventTarget(), __func__, + [sender = mSender, receiver = mReceiver]() mutable { + // Shutdown pipelines when conduits are guaranteed shut down, + // so that all packets sent from conduits can be delivered. + sender->Shutdown(); + receiver->Shutdown(); + }); + mCallWrapper = nullptr; + } + mStopped = true; + mCurrentDirection.SetNull(); + + mSender->Stop(); + mReceiver->Stop(); + + auto self = nsMainThreadPtrHandle<RTCRtpTransceiver>( + new nsMainThreadPtrHolder<RTCRtpTransceiver>( + "RTCRtpTransceiver::StopImpl::self", this, false)); + mStsThread->Dispatch(NS_NewRunnableFunction( + __func__, [self] { self->mTransportHandler = nullptr; })); +} + +bool RTCRtpTransceiver::IsVideo() const { return mIsVideo; } + +bool RTCRtpTransceiver::IsSending() const { + return mCurrentDirection == Nullable(RTCRtpTransceiverDirection::Sendonly) || + mCurrentDirection == Nullable(RTCRtpTransceiverDirection::Sendrecv); +} + +bool RTCRtpTransceiver::IsReceiving() const { + return mCurrentDirection == Nullable(RTCRtpTransceiverDirection::Recvonly) || + mCurrentDirection == Nullable(RTCRtpTransceiverDirection::Sendrecv); +} + +void RTCRtpTransceiver::ChainToDomPromiseWithCodecStats( + nsTArray<RefPtr<RTCStatsPromise>> aStats, + const RefPtr<dom::Promise>& aDomPromise) { + nsTArray<RTCCodecStats> codecStats = + mPc->GetCodecStats(mPc->GetTimestampMaker().GetNow().ToDom()); + + AutoTArray< + std::tuple<RTCRtpTransceiver*, RefPtr<RTCStatsPromise::AllPromiseType>>, + 1> + statsPromises; + statsPromises.AppendElement(std::make_tuple( + this, RTCStatsPromise::All(GetMainThreadSerialEventTarget(), aStats))); + + ApplyCodecStats(std::move(codecStats), std::move(statsPromises)) + ->Then( + GetMainThreadSerialEventTarget(), __func__, + [aDomPromise, window = mWindow, + idGen = mIdGenerator](UniquePtr<RTCStatsCollection> aStats) mutable { + // Rewrite ids and merge stats collections into the final report. + AutoTArray<UniquePtr<RTCStatsCollection>, 1> stats; + stats.AppendElement(std::move(aStats)); + + RTCStatsCollection opaqueStats; + idGen->RewriteIds(std::move(stats), &opaqueStats); + + RefPtr<RTCStatsReport> report(new RTCStatsReport(window)); + report->Incorporate(opaqueStats); + + aDomPromise->MaybeResolve(std::move(report)); + }, + [aDomPromise](nsresult aError) { + aDomPromise->MaybeReject(NS_ERROR_FAILURE); + }); +} + +RefPtr<RTCStatsPromise> RTCRtpTransceiver::ApplyCodecStats( + nsTArray<RTCCodecStats> aCodecStats, + nsTArray< + std::tuple<RTCRtpTransceiver*, RefPtr<RTCStatsPromise::AllPromiseType>>> + aTransceiverStatsPromises) { + MOZ_ASSERT(NS_IsMainThread()); + // The process here is roughly: + // - Gather all inputs to the codec filtering process, including: + // - Each transceiver's transportIds + // - Each transceiver's active payload types (resolved) + // - Each transceiver's resolved stats + // + // Waiting (async) for multiple promises of different types is not supported + // by the MozPromise API (bug 1752318), so we are a bit finicky here. We + // create media::Refcountables of the types we want to resolve, and let + // these be shared across Then-functions through RefPtrs. + // + // - For each active payload type in a transceiver: + // - Register the codec stats for this payload type and transport if we + // haven't already done so + // - If it was a send payload type, assign the codec stats id for this + // payload type and transport to the transceiver's outbound-rtp and + // remote-inbound-rtp stats as codecId + // - If it was a recv payload type, assign the codec stats id for this + // payload type and transport to the transceiver's inbound-rtp and + // remote-outbound-rtp stats as codecId + // + // - Flatten all transceiver stats collections into one, and set the + // registered codec stats on it + + // Wrap codec stats in a Refcountable<> to allow sharing across promise + // handlers. + auto codecStats = MakeRefPtr<media::Refcountable<nsTArray<RTCCodecStats>>>(); + *codecStats = std::move(aCodecStats); + + struct IdComparator { + bool operator()(const RTCCodecStats& aA, const RTCCodecStats& aB) const { + return aA.mId.Value() < aB.mId.Value(); + } + }; + + // Stores distinct codec stats by id; to avoid dupes within a transport. + auto finalCodecStats = + MakeRefPtr<media::Refcountable<std::set<RTCCodecStats, IdComparator>>>(); + + // All the transceiver rtp stream stats in a single array. These stats will, + // when resolved, contain codecIds. + nsTArray<RefPtr<RTCStatsPromise>> promises( + aTransceiverStatsPromises.Length()); + + for (const auto& [transceiver, allPromise] : aTransceiverStatsPromises) { + // Per transceiver, gather up what we need to assign codecId to this + // transceiver's rtp stream stats. Register codec stats while we're at it. + auto payloadTypes = + MakeRefPtr<media::Refcountable<RTCRtpTransceiver::PayloadTypes>>(); + promises.AppendElement( + transceiver->GetActivePayloadTypes() + ->Then( + GetMainThreadSerialEventTarget(), __func__, + [payloadTypes, allPromise = allPromise]( + RTCRtpTransceiver::PayloadTypes aPayloadTypes) { + // Forward active payload types to the next Then-handler. + *payloadTypes = std::move(aPayloadTypes); + return allPromise; + }, + [] { + MOZ_CRASH("Unexpected reject"); + return RTCStatsPromise::AllPromiseType::CreateAndReject( + NS_ERROR_UNEXPECTED, __func__); + }) + ->Then( + GetMainThreadSerialEventTarget(), __func__, + [codecStats, finalCodecStats, payloadTypes, + transportId = + NS_ConvertASCIItoUTF16(transceiver->GetTransportId())]( + nsTArray<UniquePtr<RTCStatsCollection>> + aTransceiverStats) mutable { + // We have all the data we need to register codec stats and + // assign codecIds for this transceiver's rtp stream stats. + + auto report = MakeUnique<RTCStatsCollection>(); + FlattenStats(std::move(aTransceiverStats), report.get()); + + // Find the codec stats we are looking for, based on the + // transportId and the active payload types. + Maybe<RTCCodecStats&> sendCodec; + Maybe<RTCCodecStats&> recvCodec; + for (auto& codec : *codecStats) { + if (payloadTypes->mSendPayloadType.isSome() == + sendCodec.isSome() && + payloadTypes->mRecvPayloadType.isSome() == + recvCodec.isSome()) { + // We have found all the codec stats we were looking for. + break; + } + if (codec.mTransportId != transportId) { + continue; + } + if (payloadTypes->mSendPayloadType && + *payloadTypes->mSendPayloadType == + static_cast<int>(codec.mPayloadType) && + (!codec.mCodecType.WasPassed() || + codec.mCodecType.Value() == RTCCodecType::Encode)) { + MOZ_ASSERT(!sendCodec, + "At most one send codec stat per transceiver"); + sendCodec = SomeRef(codec); + } + if (payloadTypes->mRecvPayloadType && + *payloadTypes->mRecvPayloadType == + static_cast<int>(codec.mPayloadType) && + (!codec.mCodecType.WasPassed() || + codec.mCodecType.Value() == RTCCodecType::Decode)) { + MOZ_ASSERT(!recvCodec, + "At most one recv codec stat per transceiver"); + recvCodec = SomeRef(codec); + } + } + + // Register and assign codecIds for the found codec stats. + if (sendCodec) { + finalCodecStats->insert(*sendCodec); + for (auto& stat : report->mOutboundRtpStreamStats) { + stat.mCodecId.Construct(sendCodec->mId.Value()); + } + for (auto& stat : report->mRemoteInboundRtpStreamStats) { + stat.mCodecId.Construct(sendCodec->mId.Value()); + } + } + if (recvCodec) { + finalCodecStats->insert(*recvCodec); + for (auto& stat : report->mInboundRtpStreamStats) { + stat.mCodecId.Construct(recvCodec->mId.Value()); + } + for (auto& stat : report->mRemoteOutboundRtpStreamStats) { + stat.mCodecId.Construct(recvCodec->mId.Value()); + } + } + + return RTCStatsPromise::CreateAndResolve(std::move(report), + __func__); + }, + [] { + MOZ_CRASH("Unexpected reject"); + return RTCStatsPromise::CreateAndReject(NS_ERROR_UNEXPECTED, + __func__); + })); + } + + return RTCStatsPromise::All(GetMainThreadSerialEventTarget(), promises) + ->Then( + GetMainThreadSerialEventTarget(), __func__, + [finalCodecStats = std::move(finalCodecStats)]( + nsTArray<UniquePtr<RTCStatsCollection>> aStats) mutable { + auto finalStats = MakeUnique<RTCStatsCollection>(); + FlattenStats(std::move(aStats), finalStats.get()); + MOZ_ASSERT(finalStats->mCodecStats.IsEmpty()); + if (!finalStats->mCodecStats.SetCapacity(finalCodecStats->size(), + fallible)) { + mozalloc_handle_oom(0); + } + while (!finalCodecStats->empty()) { + auto node = finalCodecStats->extract(finalCodecStats->begin()); + if (!finalStats->mCodecStats.AppendElement( + std::move(node.value()), fallible)) { + mozalloc_handle_oom(0); + } + } + return RTCStatsPromise::CreateAndResolve(std::move(finalStats), + __func__); + }, + [] { + MOZ_CRASH("Unexpected reject"); + return RTCStatsPromise::CreateAndReject(NS_ERROR_UNEXPECTED, + __func__); + }); +} + +} // namespace mozilla |