diff options
Diffstat (limited to 'dom/media/webrtc/jsapi/RTCRtpReceiver.cpp')
| -rw-r--r-- | dom/media/webrtc/jsapi/RTCRtpReceiver.cpp | 942 |
1 files changed, 942 insertions, 0 deletions
diff --git a/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp new file mode 100644 index 0000000000..136aa5142f --- /dev/null +++ b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp @@ -0,0 +1,942 @@ +/* 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 "RTCRtpReceiver.h" +#include "PeerConnectionImpl.h" +#include "mozilla/dom/RTCRtpCapabilitiesBinding.h" +#include "transport/logging.h" +#include "mozilla/dom/MediaStreamTrack.h" +#include "mozilla/dom/Promise.h" +#include "nsPIDOMWindow.h" +#include "PrincipalHandle.h" +#include "nsIPrincipal.h" +#include "mozilla/dom/Document.h" +#include "mozilla/NullPrincipal.h" +#include "MediaTrackGraph.h" +#include "RemoteTrackSource.h" +#include "libwebrtcglue/RtpRtcpConfig.h" +#include "nsString.h" +#include "mozilla/dom/AudioStreamTrack.h" +#include "mozilla/dom/VideoStreamTrack.h" +#include "MediaTransportHandler.h" +#include "jsep/JsepTransceiver.h" +#include "mozilla/dom/RTCRtpReceiverBinding.h" +#include "mozilla/dom/RTCRtpSourcesBinding.h" +#include "RTCStatsReport.h" +#include "mozilla/Preferences.h" +#include "PeerConnectionCtx.h" +#include "RTCRtpTransceiver.h" +#include "libwebrtcglue/AudioConduit.h" +#include "call/call.h" + +namespace mozilla::dom { + +LazyLogModule gReceiverLog("RTCRtpReceiver"); + +NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpReceiver) +NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(RTCRtpReceiver) + // We do not do anything here, we wait for BreakCycles to be called + NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER +NS_IMPL_CYCLE_COLLECTION_UNLINK_END +NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(RTCRtpReceiver) + NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mWindow, mPc, mTransceiver, mTrack, + mTrackSource) +NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END + +NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCRtpReceiver) +NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCRtpReceiver) +NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCRtpReceiver) + NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY + NS_INTERFACE_MAP_ENTRY(nsISupports) +NS_INTERFACE_MAP_END + +static PrincipalHandle GetPrincipalHandle(nsPIDOMWindowInner* aWindow, + PrincipalPrivacy aPrivacy) { + // Set the principal used for creating the tracks. This makes the track + // data (audio/video samples) accessible to the receiving page. We're + // only certain that privacy hasn't been requested if we're connected. + nsCOMPtr<nsIScriptObjectPrincipal> winPrincipal = do_QueryInterface(aWindow); + RefPtr<nsIPrincipal> principal = winPrincipal->GetPrincipal(); + if (NS_WARN_IF(!principal)) { + principal = NullPrincipal::CreateWithoutOriginAttributes(); + } else if (aPrivacy == PrincipalPrivacy::Private) { + principal = NullPrincipal::CreateWithInheritedAttributes(principal); + } + return MakePrincipalHandle(principal); +} + +#define INIT_CANONICAL(name, val) \ + name(AbstractThread::MainThread(), val, \ + "RTCRtpReceiver::" #name " (Canonical)") + +RTCRtpReceiver::RTCRtpReceiver( + nsPIDOMWindowInner* aWindow, PrincipalPrivacy aPrivacy, + PeerConnectionImpl* aPc, MediaTransportHandler* aTransportHandler, + AbstractThread* aCallThread, nsISerialEventTarget* aStsThread, + MediaSessionConduit* aConduit, RTCRtpTransceiver* aTransceiver, + const TrackingId& aTrackingId) + : mWatchManager(this, AbstractThread::MainThread()), + mWindow(aWindow), + mPc(aPc), + mCallThread(aCallThread), + mStsThread(aStsThread), + mTransportHandler(aTransportHandler), + mTransceiver(aTransceiver), + INIT_CANONICAL(mSsrc, 0), + INIT_CANONICAL(mVideoRtxSsrc, 0), + INIT_CANONICAL(mLocalRtpExtensions, RtpExtList()), + INIT_CANONICAL(mAudioCodecs, std::vector<AudioCodecConfig>()), + INIT_CANONICAL(mVideoCodecs, std::vector<VideoCodecConfig>()), + INIT_CANONICAL(mVideoRtpRtcpConfig, Nothing()), + INIT_CANONICAL(mReceiving, false) { + PrincipalHandle principalHandle = GetPrincipalHandle(aWindow, aPrivacy); + const bool isAudio = aConduit->type() == MediaSessionConduit::AUDIO; + + MediaTrackGraph* graph = MediaTrackGraph::GetInstance( + isAudio ? MediaTrackGraph::AUDIO_THREAD_DRIVER + : MediaTrackGraph::SYSTEM_THREAD_DRIVER, + aWindow, MediaTrackGraph::REQUEST_DEFAULT_SAMPLE_RATE, + MediaTrackGraph::DEFAULT_OUTPUT_DEVICE); + + if (isAudio) { + auto* source = graph->CreateSourceTrack(MediaSegment::AUDIO); + mTrackSource = MakeAndAddRef<RemoteTrackSource>( + source, this, principalHandle, u"remote audio"_ns, aTrackingId); + mTrack = MakeAndAddRef<AudioStreamTrack>(aWindow, source, mTrackSource); + mPipeline = MakeAndAddRef<MediaPipelineReceiveAudio>( + mPc->GetHandle(), aTransportHandler, aCallThread, mStsThread.get(), + *aConduit->AsAudioSessionConduit(), mTrackSource->Stream(), aTrackingId, + principalHandle, aPrivacy); + } else { + auto* source = graph->CreateSourceTrack(MediaSegment::VIDEO); + mTrackSource = MakeAndAddRef<RemoteTrackSource>( + source, this, principalHandle, u"remote video"_ns, aTrackingId); + mTrack = MakeAndAddRef<VideoStreamTrack>(aWindow, source, mTrackSource); + mPipeline = MakeAndAddRef<MediaPipelineReceiveVideo>( + mPc->GetHandle(), aTransportHandler, aCallThread, mStsThread.get(), + *aConduit->AsVideoSessionConduit(), mTrackSource->Stream(), aTrackingId, + principalHandle, aPrivacy); + } + + mPipeline->InitControl(this); + + // Spec says remote tracks start out muted. + mTrackSource->SetMuted(true); + + // Until Bug 1232234 is fixed, we'll get extra RTCP BYES during renegotiation, + // so we'll disable muting on RTCP BYE and timeout for now. + if (Preferences::GetBool("media.peerconnection.mute_on_bye_or_timeout", + false)) { + mRtcpByeListener = aConduit->RtcpByeEvent().Connect( + GetMainThreadSerialEventTarget(), this, &RTCRtpReceiver::OnRtcpBye); + mRtcpTimeoutListener = aConduit->RtcpTimeoutEvent().Connect( + GetMainThreadSerialEventTarget(), this, &RTCRtpReceiver::OnRtcpTimeout); + } + + mWatchManager.Watch(mReceiveTrackMute, + &RTCRtpReceiver::UpdateReceiveTrackMute); +} + +#undef INIT_CANONICAL + +RTCRtpReceiver::~RTCRtpReceiver() { MOZ_ASSERT(!mPipeline); } + +JSObject* RTCRtpReceiver::WrapObject(JSContext* aCx, + JS::Handle<JSObject*> aGivenProto) { + return RTCRtpReceiver_Binding::Wrap(aCx, this, aGivenProto); +} + +RTCDtlsTransport* RTCRtpReceiver::GetTransport() const { + if (!mTransceiver) { + return nullptr; + } + return mTransceiver->GetDtlsTransport(); +} + +void RTCRtpReceiver::GetCapabilities( + const GlobalObject&, const nsAString& aKind, + Nullable<dom::RTCRtpCapabilities>& aResult) { + PeerConnectionImpl::GetCapabilities(aKind, aResult, sdp::Direction::kRecv); +} + +already_AddRefed<Promise> RTCRtpReceiver::GetStats(ErrorResult& aError) { + nsCOMPtr<nsIGlobalObject> global = do_QueryInterface(mWindow); + RefPtr<Promise> promise = Promise::Create(global, aError); + if (NS_WARN_IF(aError.Failed())) { + return nullptr; + } + + if (NS_WARN_IF(!mTransceiver)) { + // TODO(bug 1056433): When we stop nulling this out when the PC is closed + // (or when the transceiver is stopped), we can remove this code. We + // resolve instead of reject in order to make this eventual change in + // behavior a little smaller. + promise->MaybeResolve(new RTCStatsReport(mWindow)); + return promise.forget(); + } + + mTransceiver->ChainToDomPromiseWithCodecStats(GetStatsInternal(), promise); + return promise.forget(); +} + +nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal( + bool aSkipIceStats) { + MOZ_ASSERT(NS_IsMainThread()); + nsTArray<RefPtr<RTCStatsPromise>> promises(3); + + if (!mPipeline) { + return promises; + } + + if (!mHaveStartedReceiving) { + return promises; + } + + nsString recvTrackId; + MOZ_ASSERT(mTrack); + if (mTrack) { + mTrack->GetId(recvTrackId); + } + + { + // Add bandwidth estimation stats + promises.AppendElement(InvokeAsync( + mCallThread, __func__, + [conduit = mPipeline->mConduit, recvTrackId]() mutable { + auto report = MakeUnique<dom::RTCStatsCollection>(); + const Maybe<webrtc::Call::Stats> stats = conduit->GetCallStats(); + stats.apply([&](const auto& aStats) { + dom::RTCBandwidthEstimationInternal bw; + bw.mTrackIdentifier = recvTrackId; + bw.mSendBandwidthBps.Construct(aStats.send_bandwidth_bps / 8); + bw.mMaxPaddingBps.Construct(aStats.max_padding_bitrate_bps / 8); + bw.mReceiveBandwidthBps.Construct(aStats.recv_bandwidth_bps / 8); + bw.mPacerDelayMs.Construct(aStats.pacer_delay_ms); + if (aStats.rtt_ms >= 0) { + bw.mRttMs.Construct(aStats.rtt_ms); + } + if (!report->mBandwidthEstimations.AppendElement(std::move(bw), + fallible)) { + mozalloc_handle_oom(0); + } + }); + return RTCStatsPromise::CreateAndResolve(std::move(report), __func__); + })); + } + + promises.AppendElement( + InvokeAsync( + mCallThread, __func__, + [pipeline = mPipeline, recvTrackId] { + auto report = MakeUnique<dom::RTCStatsCollection>(); + auto asAudio = pipeline->mConduit->AsAudioSessionConduit(); + auto asVideo = pipeline->mConduit->AsVideoSessionConduit(); + + nsString kind = asVideo.isNothing() ? u"audio"_ns : u"video"_ns; + nsString idstr = kind + u"_"_ns; + idstr.AppendInt(static_cast<uint32_t>(pipeline->Level())); + + Maybe<uint32_t> ssrc = pipeline->mConduit->GetRemoteSSRC(); + + // Add frame history + asVideo.apply([&](const auto& conduit) { + if (conduit->AddFrameHistory(&report->mVideoFrameHistories)) { + auto& history = report->mVideoFrameHistories.LastElement(); + history.mTrackIdentifier = recvTrackId; + } + }); + + // TODO(@@NG):ssrcs handle Conduits having multiple stats at the + // same level. + // This is pending spec work. + // Gather pipeline stats. + nsString localId = u"inbound_rtp_"_ns + idstr; + nsString remoteId; + + auto constructCommonRemoteOutboundRtpStats = + [&](RTCRemoteOutboundRtpStreamStats& aRemote, + const DOMHighResTimeStamp& aTimestamp) { + remoteId = u"inbound_rtcp_"_ns + idstr; + aRemote.mTimestamp.Construct(aTimestamp); + aRemote.mId.Construct(remoteId); + aRemote.mType.Construct(RTCStatsType::Remote_outbound_rtp); + ssrc.apply([&](uint32_t aSsrc) { aRemote.mSsrc = aSsrc; }); + aRemote.mKind = kind; + aRemote.mMediaType.Construct( + kind); // mediaType is the old name for kind. + aRemote.mLocalId.Construct(localId); + }; + + auto constructCommonInboundRtpStats = + [&](RTCInboundRtpStreamStats& aLocal) { + aLocal.mTrackIdentifier = recvTrackId; + aLocal.mTimestamp.Construct( + pipeline->GetTimestampMaker().GetNow().ToDom()); + aLocal.mId.Construct(localId); + aLocal.mType.Construct(RTCStatsType::Inbound_rtp); + ssrc.apply([&](uint32_t aSsrc) { aLocal.mSsrc = aSsrc; }); + aLocal.mKind = kind; + aLocal.mMediaType.Construct( + kind); // mediaType is the old name for kind. + if (remoteId.Length()) { + aLocal.mRemoteId.Construct(remoteId); + } + }; + + asAudio.apply([&](auto& aConduit) { + Maybe<webrtc::AudioReceiveStreamInterface::Stats> audioStats = + aConduit->GetReceiverStats(); + if (audioStats.isNothing()) { + return; + } + + if (!audioStats->last_packet_received_timestamp_ms) { + // By spec: "The lifetime of all RTP monitored objects starts + // when the RTP stream is first used: When the first RTP packet + // is sent or received on the SSRC it represents" + return; + } + + // First, fill in remote stat with rtcp sender data, if present. + if (audioStats->last_sender_report_timestamp_ms) { + RTCRemoteOutboundRtpStreamStats remote; + constructCommonRemoteOutboundRtpStats( + remote, + RTCStatsTimestamp::FromNtp( + aConduit->GetTimestampMaker(), + webrtc::Timestamp::Millis( + *audioStats->last_sender_report_timestamp_ms) + + webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970)) + .ToDom()); + remote.mPacketsSent.Construct( + audioStats->sender_reports_packets_sent); + remote.mBytesSent.Construct( + audioStats->sender_reports_bytes_sent); + remote.mRemoteTimestamp.Construct( + *audioStats->last_sender_report_remote_timestamp_ms); + if (!report->mRemoteOutboundRtpStreamStats.AppendElement( + std::move(remote), fallible)) { + mozalloc_handle_oom(0); + } + } + + // Then, fill in local side (with cross-link to remote only if + // present) + RTCInboundRtpStreamStats local; + constructCommonInboundRtpStats(local); + local.mJitter.Construct(audioStats->jitter_ms / 1000.0); + local.mPacketsLost.Construct(audioStats->packets_lost); + local.mPacketsReceived.Construct(audioStats->packets_rcvd); + local.mPacketsDiscarded.Construct(audioStats->packets_discarded); + local.mBytesReceived.Construct(audioStats->payload_bytes_rcvd); + // Always missing from libwebrtc stats + // if (audioStats->estimated_playout_ntp_timestamp_ms) { + // local.mEstimatedPlayoutTimestamp.Construct( + // RTCStatsTimestamp::FromNtp( + // aConduit->GetTimestampMaker(), + // webrtc::Timestamp::Millis( + // *audioStats->estimated_playout_ntp_timestamp_ms)) + // .ToDom()); + // } + local.mJitterBufferDelay.Construct( + audioStats->jitter_buffer_delay_seconds); + local.mJitterBufferEmittedCount.Construct( + audioStats->jitter_buffer_emitted_count); + local.mTotalSamplesReceived.Construct( + audioStats->total_samples_received); + local.mConcealedSamples.Construct(audioStats->concealed_samples); + local.mSilentConcealedSamples.Construct( + audioStats->silent_concealed_samples); + if (audioStats->last_packet_received_timestamp_ms) { + local.mLastPacketReceivedTimestamp.Construct( + RTCStatsTimestamp::FromNtp( + aConduit->GetTimestampMaker(), + webrtc::Timestamp::Millis( + *audioStats->last_packet_received_timestamp_ms) + + webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970)) + .ToDom()); + } + local.mHeaderBytesReceived.Construct( + audioStats->header_and_padding_bytes_rcvd); + local.mFecPacketsReceived.Construct( + audioStats->fec_packets_received); + local.mFecPacketsDiscarded.Construct( + audioStats->fec_packets_discarded); + local.mConcealmentEvents.Construct( + audioStats->concealment_events); + + local.mInsertedSamplesForDeceleration.Construct( + audioStats->inserted_samples_for_deceleration); + local.mRemovedSamplesForAcceleration.Construct( + audioStats->removed_samples_for_acceleration); + if (audioStats->audio_level >= 0 && + audioStats->audio_level <= 32767) { + local.mAudioLevel.Construct(audioStats->audio_level / 32767.0); + } + local.mTotalAudioEnergy.Construct( + audioStats->total_output_energy); + local.mTotalSamplesDuration.Construct( + audioStats->total_output_duration); + + if (!report->mInboundRtpStreamStats.AppendElement( + std::move(local), fallible)) { + mozalloc_handle_oom(0); + } + }); + + asVideo.apply([&](auto& aConduit) { + Maybe<webrtc::VideoReceiveStreamInterface::Stats> videoStats = + aConduit->GetReceiverStats(); + if (videoStats.isNothing()) { + return; + } + + if (!videoStats->rtp_stats.last_packet_received_timestamp_ms) { + // By spec: "The lifetime of all RTP monitored objects starts + // when the RTP stream is first used: When the first RTP packet + // is sent or received on the SSRC it represents" + return; + } + + // First, fill in remote stat with rtcp sender data, if present. + if (videoStats->rtcp_sender_ntp_timestamp_ms) { + RTCRemoteOutboundRtpStreamStats remote; + constructCommonRemoteOutboundRtpStats( + remote, RTCStatsTimestamp::FromNtp( + aConduit->GetTimestampMaker(), + webrtc::Timestamp::Millis( + videoStats->rtcp_sender_ntp_timestamp_ms)) + .ToDom()); + remote.mPacketsSent.Construct( + videoStats->rtcp_sender_packets_sent); + remote.mBytesSent.Construct( + videoStats->rtcp_sender_octets_sent); + remote.mRemoteTimestamp.Construct( + (webrtc::TimeDelta::Millis( + videoStats->rtcp_sender_remote_ntp_timestamp_ms) - + webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970)) + .ms()); + if (!report->mRemoteOutboundRtpStreamStats.AppendElement( + std::move(remote), fallible)) { + mozalloc_handle_oom(0); + } + } + + // Then, fill in local side (with cross-link to remote only if + // present) + RTCInboundRtpStreamStats local; + constructCommonInboundRtpStats(local); + local.mJitter.Construct( + static_cast<double>(videoStats->rtp_stats.jitter) / + webrtc::kVideoPayloadTypeFrequency); + local.mPacketsLost.Construct(videoStats->rtp_stats.packets_lost); + local.mPacketsReceived.Construct( + videoStats->rtp_stats.packet_counter.packets); + local.mPacketsDiscarded.Construct(videoStats->packets_discarded); + local.mDiscardedPackets.Construct(videoStats->packets_discarded); + local.mBytesReceived.Construct( + videoStats->rtp_stats.packet_counter.payload_bytes); + + // Fill in packet type statistics + local.mNackCount.Construct( + videoStats->rtcp_packet_type_counts.nack_packets); + local.mFirCount.Construct( + videoStats->rtcp_packet_type_counts.fir_packets); + local.mPliCount.Construct( + videoStats->rtcp_packet_type_counts.pli_packets); + + // Lastly, fill in video decoder stats + local.mFramesDecoded.Construct(videoStats->frames_decoded); + + local.mFramesPerSecond.Construct(videoStats->decode_frame_rate); + local.mFrameWidth.Construct(videoStats->width); + local.mFrameHeight.Construct(videoStats->height); + // XXX: key_frames + delta_frames may undercount frames because + // they were dropped in FrameBuffer::InsertFrame. (bug 1766553) + local.mFramesReceived.Construct( + videoStats->frame_counts.key_frames + + videoStats->frame_counts.delta_frames); + local.mJitterBufferDelay.Construct( + videoStats->jitter_buffer_delay_seconds); + local.mJitterBufferEmittedCount.Construct( + videoStats->jitter_buffer_emitted_count); + + if (videoStats->qp_sum) { + local.mQpSum.Construct(videoStats->qp_sum.value()); + } + local.mTotalDecodeTime.Construct( + double(videoStats->total_decode_time.ms()) / 1000); + local.mTotalInterFrameDelay.Construct( + videoStats->total_inter_frame_delay); + local.mTotalSquaredInterFrameDelay.Construct( + videoStats->total_squared_inter_frame_delay); + if (videoStats->rtp_stats.last_packet_received_timestamp_ms) { + local.mLastPacketReceivedTimestamp.Construct( + RTCStatsTimestamp::FromNtp( + aConduit->GetTimestampMaker(), + webrtc::Timestamp::Millis( + *videoStats->rtp_stats + .last_packet_received_timestamp_ms) + + webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970)) + .ToDom()); + } + local.mHeaderBytesReceived.Construct( + videoStats->rtp_stats.packet_counter.header_bytes + + videoStats->rtp_stats.packet_counter.padding_bytes); + local.mTotalProcessingDelay.Construct( + videoStats->total_processing_delay.seconds<double>()); + /* + * Potential new stats that are now available upstream + .if (videoStats->estimated_playout_ntp_timestamp_ms) { + local.mEstimatedPlayoutTimestamp.Construct( + RTCStatsTimestamp::FromNtp( + aConduit->GetTimestampMaker(), + webrtc::Timestamp::Millis( + *videoStats->estimated_playout_ntp_timestamp_ms)) + .ToDom()); + } + */ + // Not including frames dropped in the rendering pipe, which + // is not of webrtc's concern anyway?! + local.mFramesDropped.Construct(videoStats->frames_dropped); + if (!report->mInboundRtpStreamStats.AppendElement( + std::move(local), fallible)) { + mozalloc_handle_oom(0); + } + }); + return RTCStatsPromise::CreateAndResolve(std::move(report), + __func__); + }) + ->Then( + mStsThread, __func__, + [pipeline = mPipeline](UniquePtr<RTCStatsCollection> aReport) { + // Fill in Contributing Source statistics + if (!aReport->mInboundRtpStreamStats.IsEmpty() && + aReport->mInboundRtpStreamStats[0].mId.WasPassed()) { + pipeline->GetContributingSourceStats( + aReport->mInboundRtpStreamStats[0].mId.Value(), + aReport->mRtpContributingSourceStats); + } + return RTCStatsPromise::CreateAndResolve(std::move(aReport), + __func__); + }, + [] { + MOZ_CRASH("Unexpected reject"); + return RTCStatsPromise::CreateAndReject(NS_ERROR_UNEXPECTED, + __func__); + })); + + if (!aSkipIceStats && GetJsepTransceiver().mTransport.mComponents) { + promises.AppendElement(mTransportHandler->GetIceStats( + GetJsepTransceiver().mTransport.mTransportId, + mPipeline->GetTimestampMaker().GetNow().ToDom())); + } + + return promises; +} + +void RTCRtpReceiver::SetJitterBufferTarget( + const Nullable<DOMHighResTimeStamp>& aTargetMs, ErrorResult& aError) { + // Spec says jitter buffer target cannot be negative or larger than 4000 + // milliseconds and to throw RangeError if it is. If an invalid value is + // received we return early to preserve the current JitterBufferTarget + // internal slot and jitter buffer values. + if (mPipeline && mPipeline->mConduit) { + if (!aTargetMs.IsNull() && + (aTargetMs.Value() < 0.0 || aTargetMs.Value() > 4000.0)) { + aError.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("jitterBufferTarget"); + return; + } + + mJitterBufferTarget.reset(); + + if (!aTargetMs.IsNull()) { + mJitterBufferTarget = Some(aTargetMs.Value()); + } + // If aJitterBufferTarget is null then we are resetting the jitter buffer so + // pass the default target of 0.0. + mPipeline->mConduit->SetJitterBufferTarget( + mJitterBufferTarget.valueOr(0.0)); + } +} + +void RTCRtpReceiver::GetContributingSources( + nsTArray<RTCRtpContributingSource>& aSources) { + // Duplicate code... + if (mPipeline && mPipeline->mConduit) { + nsTArray<dom::RTCRtpSourceEntry> sources; + mPipeline->mConduit->GetRtpSources(sources); + sources.RemoveElementsBy([](const dom::RTCRtpSourceEntry& aEntry) { + return aEntry.mSourceType != dom::RTCRtpSourceEntryType::Contributing; + }); + aSources.ReplaceElementsAt(0, aSources.Length(), sources.Elements(), + sources.Length()); + } +} + +void RTCRtpReceiver::GetSynchronizationSources( + nsTArray<dom::RTCRtpSynchronizationSource>& aSources) { + // Duplicate code... + if (mPipeline && mPipeline->mConduit) { + nsTArray<dom::RTCRtpSourceEntry> sources; + mPipeline->mConduit->GetRtpSources(sources); + sources.RemoveElementsBy([](const dom::RTCRtpSourceEntry& aEntry) { + return aEntry.mSourceType != dom::RTCRtpSourceEntryType::Synchronization; + }); + aSources.ReplaceElementsAt(0, aSources.Length(), sources.Elements(), + sources.Length()); + } +} + +nsPIDOMWindowInner* RTCRtpReceiver::GetParentObject() const { return mWindow; } + +void RTCRtpReceiver::Shutdown() { + MOZ_ASSERT(NS_IsMainThread()); + mWatchManager.Shutdown(); + if (mPipeline) { + mPipeline->Shutdown(); + mPipeline = nullptr; + } + if (mTrackSource) { + mTrackSource->Destroy(); + } + mCallThread = nullptr; + mRtcpByeListener.DisconnectIfExists(); + mRtcpTimeoutListener.DisconnectIfExists(); + mUnmuteListener.DisconnectIfExists(); +} + +void RTCRtpReceiver::BreakCycles() { + mWindow = nullptr; + mPc = nullptr; + mTrack = nullptr; + mTrackSource = nullptr; +} + +void RTCRtpReceiver::UpdateTransport() { + MOZ_ASSERT(NS_IsMainThread()); + if (!mHaveSetupTransport) { + mPipeline->SetLevel(GetJsepTransceiver().GetLevel()); + mHaveSetupTransport = true; + } + + UniquePtr<MediaPipelineFilter> filter; + + auto const& details = GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails(); + if (GetJsepTransceiver().HasBundleLevel() && details) { + std::vector<webrtc::RtpExtension> extmaps; + details->ForEachRTPHeaderExtension( + [&extmaps](const SdpExtmapAttributeList::Extmap& extmap) { + extmaps.emplace_back(extmap.extensionname, extmap.entry); + }); + filter = MakeUnique<MediaPipelineFilter>(extmaps); + + // Add remote SSRCs so we can distinguish which RTP packets actually + // belong to this pipeline (also RTCP sender reports). + for (uint32_t ssrc : GetJsepTransceiver().mRecvTrack.GetSsrcs()) { + filter->AddRemoteSSRC(ssrc); + } + for (uint32_t ssrc : GetJsepTransceiver().mRecvTrack.GetRtxSsrcs()) { + filter->AddRemoteSSRC(ssrc); + } + auto mid = Maybe<std::string>(); + if (GetMid() != "") { + mid = Some(GetMid()); + } + filter->SetRemoteMediaStreamId(mid); + + // Add unique payload types as a last-ditch fallback + auto uniquePts = GetJsepTransceiver() + .mRecvTrack.GetNegotiatedDetails() + ->GetUniquePayloadTypes(); + for (unsigned char& uniquePt : uniquePts) { + filter->AddUniquePT(uniquePt); + } + } + + mPipeline->UpdateTransport_m(GetJsepTransceiver().mTransport.mTransportId, + std::move(filter)); +} + +void RTCRtpReceiver::UpdateConduit() { + if (mPipeline->mConduit->type() == MediaSessionConduit::VIDEO) { + UpdateVideoConduit(); + } else { + UpdateAudioConduit(); + } + + if ((mReceiving = mTransceiver->IsReceiving())) { + mHaveStartedReceiving = true; + } +} + +void RTCRtpReceiver::UpdateVideoConduit() { + RefPtr<VideoSessionConduit> conduit = + *mPipeline->mConduit->AsVideoSessionConduit(); + + // NOTE(pkerr) - this is new behavior. Needed because the + // CreateVideoReceiveStream method of the Call API will assert (in debug) + // and fail if a value is not provided for the remote_ssrc that will be used + // by the far-end sender. + if (!GetJsepTransceiver().mRecvTrack.GetSsrcs().empty()) { + MOZ_LOG(gReceiverLog, LogLevel::Debug, + ("%s[%s]: %s Setting remote SSRC %u", mPc->GetHandle().c_str(), + GetMid().c_str(), __FUNCTION__, + GetJsepTransceiver().mRecvTrack.GetSsrcs().front())); + uint32_t rtxSsrc = + GetJsepTransceiver().mRecvTrack.GetRtxSsrcs().empty() + ? 0 + : GetJsepTransceiver().mRecvTrack.GetRtxSsrcs().front(); + mSsrc = GetJsepTransceiver().mRecvTrack.GetSsrcs().front(); + mVideoRtxSsrc = rtxSsrc; + + // TODO (bug 1423041) once we pay attention to receiving MID's in RTP + // packets (see bug 1405495) we could make this depending on the presence of + // MID in the RTP packets instead of relying on the signaling. + // In any case, do not disable SSRC changes if no SSRCs were negotiated + if (GetJsepTransceiver().HasBundleLevel() && + (!GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() || + !GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails()->GetExt( + webrtc::RtpExtension::kMidUri))) { + mCallThread->Dispatch( + NewRunnableMethod("VideoSessionConduit::DisableSsrcChanges", conduit, + &VideoSessionConduit::DisableSsrcChanges)); + } + } + + if (GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() && + GetJsepTransceiver().mRecvTrack.GetActive()) { + const auto& details( + *GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails()); + + { + std::vector<webrtc::RtpExtension> extmaps; + // @@NG read extmap from track + details.ForEachRTPHeaderExtension( + [&extmaps](const SdpExtmapAttributeList::Extmap& extmap) { + extmaps.emplace_back(extmap.extensionname, extmap.entry); + }); + mLocalRtpExtensions = extmaps; + } + + std::vector<VideoCodecConfig> configs; + RTCRtpTransceiver::NegotiatedDetailsToVideoCodecConfigs(details, &configs); + if (configs.empty()) { + // TODO: Are we supposed to plumb this error back to JS? This does not + // seem like a failure to set an answer, it just means that codec + // negotiation failed. For now, we're just doing the same thing we do + // if negotiation as a whole failed. + MOZ_LOG(gReceiverLog, LogLevel::Error, + ("%s[%s]: %s No video codecs were negotiated (recv).", + mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__)); + return; + } + + mVideoCodecs = configs; + mVideoRtpRtcpConfig = Some(details.GetRtpRtcpConfig()); + } +} + +void RTCRtpReceiver::UpdateAudioConduit() { + RefPtr<AudioSessionConduit> conduit = + *mPipeline->mConduit->AsAudioSessionConduit(); + + if (!GetJsepTransceiver().mRecvTrack.GetSsrcs().empty()) { + MOZ_LOG(gReceiverLog, LogLevel::Debug, + ("%s[%s]: %s Setting remote SSRC %u", mPc->GetHandle().c_str(), + GetMid().c_str(), __FUNCTION__, + GetJsepTransceiver().mRecvTrack.GetSsrcs().front())); + mSsrc = GetJsepTransceiver().mRecvTrack.GetSsrcs().front(); + + // TODO (bug 1423041) once we pay attention to receiving MID's in RTP + // packets (see bug 1405495) we could make this depending on the presence of + // MID in the RTP packets instead of relying on the signaling. + // In any case, do not disable SSRC changes if no SSRCs were negotiated + if (GetJsepTransceiver().HasBundleLevel() && + (!GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() || + !GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails()->GetExt( + webrtc::RtpExtension::kMidUri))) { + mCallThread->Dispatch( + NewRunnableMethod("AudioSessionConduit::DisableSsrcChanges", conduit, + &AudioSessionConduit::DisableSsrcChanges)); + } + } + + if (GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() && + GetJsepTransceiver().mRecvTrack.GetActive()) { + const auto& details( + *GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails()); + std::vector<AudioCodecConfig> configs; + RTCRtpTransceiver::NegotiatedDetailsToAudioCodecConfigs(details, &configs); + if (configs.empty()) { + // TODO: Are we supposed to plumb this error back to JS? This does not + // seem like a failure to set an answer, it just means that codec + // negotiation failed. For now, we're just doing the same thing we do + // if negotiation as a whole failed. + MOZ_LOG(gReceiverLog, LogLevel::Error, + ("%s[%s]: %s No audio codecs were negotiated (recv)", + mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__)); + return; + } + + // Ensure conduit knows about extensions prior to creating streams + { + std::vector<webrtc::RtpExtension> extmaps; + // @@NG read extmap from track + details.ForEachRTPHeaderExtension( + [&extmaps](const SdpExtmapAttributeList::Extmap& extmap) { + extmaps.emplace_back(extmap.extensionname, extmap.entry); + }); + mLocalRtpExtensions = extmaps; + } + + mAudioCodecs = configs; + } +} + +void RTCRtpReceiver::Stop() { + MOZ_ASSERT(mTransceiver->Stopped()); + mReceiving = false; +} + +bool RTCRtpReceiver::HasTrack(const dom::MediaStreamTrack* aTrack) const { + return !aTrack || (mTrack == aTrack); +} + +void RTCRtpReceiver::SyncFromJsep(const JsepTransceiver& aJsepTransceiver) { + if (!mPipeline) { + return; + } + + // Spec says we set [[Receptive]] to true on sLD(sendrecv/recvonly), and to + // false on sRD(recvonly/inactive), sLD(sendonly/inactive), or when stop() + // is called. + bool wasReceptive = mReceptive; + mReceptive = aJsepTransceiver.mRecvTrack.GetReceptive(); + if (!wasReceptive && mReceptive) { + mUnmuteListener = mPipeline->mConduit->RtpPacketEvent().Connect( + GetMainThreadSerialEventTarget(), this, &RTCRtpReceiver::OnRtpPacket); + } else if (wasReceptive && !mReceptive) { + mUnmuteListener.DisconnectIfExists(); + } +} + +void RTCRtpReceiver::SyncToJsep(JsepTransceiver& aJsepTransceiver) const {} + +void RTCRtpReceiver::UpdateStreams(StreamAssociationChanges* aChanges) { + // We don't sort and use set_difference, because we need to report the + // added/removed streams in the order that they appear in the SDP. + std::set<std::string> newIds( + GetJsepTransceiver().mRecvTrack.GetStreamIds().begin(), + GetJsepTransceiver().mRecvTrack.GetStreamIds().end()); + MOZ_ASSERT(GetJsepTransceiver().mRecvTrack.GetRemoteSetSendBit() || + newIds.empty()); + bool needsTrackEvent = false; + for (const auto& id : mStreamIds) { + if (!newIds.count(id)) { + aChanges->mStreamAssociationsRemoved.push_back({mTrack, id}); + } + } + + std::set<std::string> oldIds(mStreamIds.begin(), mStreamIds.end()); + for (const auto& id : GetJsepTransceiver().mRecvTrack.GetStreamIds()) { + if (!oldIds.count(id)) { + needsTrackEvent = true; + aChanges->mStreamAssociationsAdded.push_back({mTrack, id}); + } + } + + mStreamIds = GetJsepTransceiver().mRecvTrack.GetStreamIds(); + + if (mRemoteSetSendBit != + GetJsepTransceiver().mRecvTrack.GetRemoteSetSendBit()) { + mRemoteSetSendBit = GetJsepTransceiver().mRecvTrack.GetRemoteSetSendBit(); + if (mRemoteSetSendBit) { + needsTrackEvent = true; + } else { + aChanges->mReceiversToMute.push_back(this); + } + } + + if (needsTrackEvent) { + aChanges->mTrackEvents.push_back({this, mStreamIds}); + } +} + +void RTCRtpReceiver::UpdatePrincipalPrivacy(PrincipalPrivacy aPrivacy) { + if (!mPipeline) { + return; + } + + if (aPrivacy != PrincipalPrivacy::Private) { + return; + } + + mPipeline->SetPrivatePrincipal(GetPrincipalHandle(mWindow, aPrivacy)); +} + +// test-only: adds fake CSRCs and audio data +void RTCRtpReceiver::MozInsertAudioLevelForContributingSource( + const uint32_t aSource, const DOMHighResTimeStamp aTimestamp, + const uint32_t aRtpTimestamp, const bool aHasLevel, const uint8_t aLevel) { + if (!mPipeline || mPipeline->IsVideo() || !mPipeline->mConduit) { + return; + } + mPipeline->mConduit->InsertAudioLevelForContributingSource( + aSource, aTimestamp, aRtpTimestamp, aHasLevel, aLevel); +} + +void RTCRtpReceiver::OnRtcpBye() { mReceiveTrackMute = true; } + +void RTCRtpReceiver::OnRtcpTimeout() { mReceiveTrackMute = true; } + +void RTCRtpReceiver::SetTrackMuteFromRemoteSdp() { + MOZ_ASSERT(!mReceptive, + "PeerConnectionImpl should have blocked unmute events prior to " + "firing mute"); + mReceiveTrackMute = true; + // Set the mute state (and fire the mute event) synchronously. Unmute is + // handled asynchronously after receiving RTP packets. + UpdateReceiveTrackMute(); + MOZ_ASSERT(mTrack->Muted(), "Muted state was indeed set synchronously"); +} + +void RTCRtpReceiver::OnRtpPacket() { + MOZ_ASSERT(mReceptive, "We should not be registered unless this is set!"); + // We should be registered since we're currently getting a callback. + mUnmuteListener.Disconnect(); + if (mReceptive) { + mReceiveTrackMute = false; + } +} + +void RTCRtpReceiver::UpdateReceiveTrackMute() { + if (!mTrack) { + return; + } + if (!mTrackSource) { + return; + } + // This sets the muted state for mTrack and all its clones. + // Idempotent -- only reacts to changes. + mTrackSource->SetMuted(mReceiveTrackMute); +} + +std::string RTCRtpReceiver::GetMid() const { + return mTransceiver->GetMidAscii(); +} + +JsepTransceiver& RTCRtpReceiver::GetJsepTransceiver() { + MOZ_ASSERT(mTransceiver); + return mTransceiver->GetJsepTransceiver(); +} + +const JsepTransceiver& RTCRtpReceiver::GetJsepTransceiver() const { + MOZ_ASSERT(mTransceiver); + return mTransceiver->GetJsepTransceiver(); +} + +} // namespace mozilla::dom + +#undef LOGTAG |