diff options
Diffstat (limited to 'dom/media/webrtc/jsapi/RTCRtpSender.cpp')
| -rw-r--r-- | dom/media/webrtc/jsapi/RTCRtpSender.cpp | 1811 |
1 files changed, 1811 insertions, 0 deletions
diff --git a/dom/media/webrtc/jsapi/RTCRtpSender.cpp b/dom/media/webrtc/jsapi/RTCRtpSender.cpp new file mode 100644 index 0000000000..4ba0491521 --- /dev/null +++ b/dom/media/webrtc/jsapi/RTCRtpSender.cpp @@ -0,0 +1,1811 @@ +/* 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 "RTCRtpSender.h" + +#include <stdint.h> + +#include <vector> +#include <string> +#include <algorithm> +#include <utility> +#include <iterator> +#include <set> +#include <sstream> + +#include "system_wrappers/include/clock.h" +#include "call/call.h" +#include "api/rtp_parameters.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "api/video_codecs/video_codec.h" +#include "api/video/video_codec_constants.h" +#include "call/audio_send_stream.h" +#include "call/video_send_stream.h" +#include "modules/rtp_rtcp/include/report_block_data.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" + +#include "nsPIDOMWindow.h" +#include "nsString.h" +#include "MainThreadUtils.h" +#include "nsCOMPtr.h" +#include "nsContentUtils.h" +#include "nsCycleCollectionParticipant.h" +#include "nsDebug.h" +#include "nsISupports.h" +#include "nsLiteralString.h" +#include "nsStringFwd.h" +#include "nsTArray.h" +#include "nsThreadUtils.h" +#include "nsWrapperCache.h" +#include "mozilla/AbstractThread.h" +#include "mozilla/AlreadyAddRefed.h" +#include "mozilla/Assertions.h" +#include "mozilla/Attributes.h" +#include "mozilla/fallible.h" +#include "mozilla/Logging.h" +#include "mozilla/mozalloc_oom.h" +#include "mozilla/MozPromise.h" +#include "mozilla/OwningNonNull.h" +#include "mozilla/RefPtr.h" +#include "mozilla/StateWatching.h" +#include "mozilla/UniquePtr.h" +#include "mozilla/Unused.h" +#include "mozilla/StateMirroring.h" +#include "mozilla/Maybe.h" +#include "mozilla/dom/MediaStreamTrack.h" +#include "mozilla/dom/Promise.h" +#include "mozilla/dom/RTCRtpScriptTransform.h" +#include "mozilla/dom/VideoStreamTrack.h" +#include "mozilla/dom/RTCRtpSenderBinding.h" +#include "mozilla/dom/MediaStreamTrackBinding.h" +#include "mozilla/dom/Nullable.h" +#include "mozilla/dom/RTCRtpParametersBinding.h" +#include "mozilla/dom/RTCStatsReportBinding.h" +#include "mozilla/glean/GleanMetrics.h" +#include "js/RootingAPI.h" +#include "jsep/JsepTransceiver.h" +#include "RTCStatsReport.h" +#include "RTCRtpTransceiver.h" +#include "PeerConnectionImpl.h" +#include "libwebrtcglue/CodecConfig.h" +#include "libwebrtcglue/MediaConduitControl.h" +#include "libwebrtcglue/MediaConduitInterface.h" +#include "sdp/SdpAttribute.h" +#include "sdp/SdpEnum.h" + +namespace mozilla::dom { + +LazyLogModule gSenderLog("RTCRtpSender"); + +NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpSender) +NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(RTCRtpSender) + tmp->Unlink(); + NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER +NS_IMPL_CYCLE_COLLECTION_UNLINK_END +NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(RTCRtpSender) + NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mWindow, mPc, mSenderTrack, mTransceiver, + mStreams, mTransform, mDtmf) +NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END + +NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCRtpSender) +NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCRtpSender) +NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCRtpSender) + NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY + NS_INTERFACE_MAP_ENTRY(nsISupports) +NS_INTERFACE_MAP_END + +#define INIT_CANONICAL(name, val) \ + name(AbstractThread::MainThread(), val, "RTCRtpSender::" #name " (Canonical)") + +RTCRtpSender::RTCRtpSender(nsPIDOMWindowInner* aWindow, PeerConnectionImpl* aPc, + MediaTransportHandler* aTransportHandler, + AbstractThread* aCallThread, + nsISerialEventTarget* aStsThread, + MediaSessionConduit* aConduit, + dom::MediaStreamTrack* aTrack, + const Sequence<RTCRtpEncodingParameters>& aEncodings, + RTCRtpTransceiver* aTransceiver) + : mWatchManager(this, AbstractThread::MainThread()), + mWindow(aWindow), + mPc(aPc), + mSenderTrack(aTrack), + mTransportHandler(aTransportHandler), + mTransceiver(aTransceiver), + INIT_CANONICAL(mSsrcs, Ssrcs()), + INIT_CANONICAL(mVideoRtxSsrcs, Ssrcs()), + INIT_CANONICAL(mLocalRtpExtensions, RtpExtList()), + INIT_CANONICAL(mAudioCodec, Nothing()), + INIT_CANONICAL(mVideoCodec, Nothing()), + INIT_CANONICAL(mVideoRtpRtcpConfig, Nothing()), + INIT_CANONICAL(mVideoCodecMode, webrtc::VideoCodecMode::kRealtimeVideo), + INIT_CANONICAL(mCname, std::string()), + INIT_CANONICAL(mTransmitting, false), + INIT_CANONICAL(mFrameTransformerProxy, nullptr) { + mPipeline = MediaPipelineTransmit::Create( + mPc->GetHandle(), aTransportHandler, aCallThread, aStsThread, + aConduit->type() == MediaSessionConduit::VIDEO, aConduit); + mPipeline->InitControl(this); + + if (aConduit->type() == MediaSessionConduit::AUDIO) { + mDtmf = new RTCDTMFSender(aWindow, mTransceiver); + } + mPipeline->SetTrack(mSenderTrack); + + mozilla::glean::rtcrtpsender::count.Add(1); + + if (mPc->ShouldAllowOldSetParameters()) { + mAllowOldSetParameters = true; + mozilla::glean::rtcrtpsender::count_setparameters_compat.Add(1); + } + + if (aEncodings.Length()) { + // This sender was created by addTransceiver with sendEncodings. + mParameters.mEncodings = aEncodings; + mSimulcastEnvelopeSet = true; + mozilla::glean::rtcrtpsender::used_sendencodings.AddToNumerator(1); + } else { + // This sender was created by addTrack, sRD(offer), or addTransceiver + // without sendEncodings. + RTCRtpEncodingParameters defaultEncoding; + defaultEncoding.mActive = true; + if (aConduit->type() == MediaSessionConduit::VIDEO) { + defaultEncoding.mScaleResolutionDownBy.Construct(1.0f); + } + Unused << mParameters.mEncodings.AppendElement(defaultEncoding, fallible); + UpdateRestorableEncodings(mParameters.mEncodings); + MaybeGetJsepRids(); + } + + if (mDtmf) { + mWatchManager.Watch(mTransmitting, &RTCRtpSender::UpdateDtmfSender); + } +} + +#undef INIT_CANONICAL + +RTCRtpSender::~RTCRtpSender() = default; + +JSObject* RTCRtpSender::WrapObject(JSContext* aCx, + JS::Handle<JSObject*> aGivenProto) { + return RTCRtpSender_Binding::Wrap(aCx, this, aGivenProto); +} + +RTCDtlsTransport* RTCRtpSender::GetTransport() const { + if (!mTransceiver) { + return nullptr; + } + return mTransceiver->GetDtlsTransport(); +} + +RTCDTMFSender* RTCRtpSender::GetDtmf() const { return mDtmf; } + +already_AddRefed<Promise> RTCRtpSender::GetStats(ErrorResult& aError) { + RefPtr<Promise> promise = MakePromise(aError); + if (aError.Failed()) { + return nullptr; + } + if (NS_WARN_IF(!mPipeline)) { + // 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(); + } + + if (!mSenderTrack) { + promise->MaybeResolve(new RTCStatsReport(mWindow)); + return promise.forget(); + } + + mTransceiver->ChainToDomPromiseWithCodecStats(GetStatsInternal(), promise); + return promise.forget(); +} + +nsTArray<RefPtr<dom::RTCStatsPromise>> RTCRtpSender::GetStatsInternal( + bool aSkipIceStats) { + MOZ_ASSERT(NS_IsMainThread()); + nsTArray<RefPtr<RTCStatsPromise>> promises(2); + if (!mSenderTrack || !mPipeline) { + return promises; + } + + nsAutoString trackName; + if (auto track = mPipeline->GetTrack()) { + track->GetId(trackName); + } + + { + // Add bandwidth estimation stats + promises.AppendElement(InvokeAsync( + mPipeline->mCallThread, __func__, + [conduit = mPipeline->mConduit, trackName]() mutable { + auto report = MakeUnique<dom::RTCStatsCollection>(); + Maybe<webrtc::Call::Stats> stats = conduit->GetCallStats(); + stats.apply([&](const auto aStats) { + dom::RTCBandwidthEstimationInternal bw; + bw.mTrackIdentifier = trackName; + 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( + mPipeline->mCallThread, __func__, [pipeline = mPipeline, trackName] { + 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())); + + for (uint32_t ssrc : pipeline->mConduit->GetLocalSSRCs()) { + nsString localId = u"outbound_rtp_"_ns + idstr + u"_"_ns; + localId.AppendInt(ssrc); + nsString remoteId; + Maybe<uint16_t> base_seq = + pipeline->mConduit->RtpSendBaseSeqFor(ssrc); + + auto constructCommonRemoteInboundRtpStats = + [&](RTCRemoteInboundRtpStreamStats& aRemote, + const webrtc::ReportBlockData& aRtcpData) { + remoteId = u"outbound_rtcp_"_ns + idstr + u"_"_ns; + remoteId.AppendInt(ssrc); + aRemote.mTimestamp.Construct( + RTCStatsTimestamp::FromNtp( + pipeline->GetTimestampMaker(), + webrtc::Timestamp::Micros( + aRtcpData.report_block_timestamp_utc().us()) + + webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970)) + .ToDom()); + aRemote.mId.Construct(remoteId); + aRemote.mType.Construct(RTCStatsType::Remote_inbound_rtp); + aRemote.mSsrc = ssrc; + aRemote.mKind = kind; + aRemote.mMediaType.Construct( + kind); // mediaType is the old name for kind. + aRemote.mLocalId.Construct(localId); + if (base_seq) { + if (aRtcpData.extended_highest_sequence_number() < + *base_seq) { + aRemote.mPacketsReceived.Construct(0); + } else { + aRemote.mPacketsReceived.Construct( + aRtcpData.extended_highest_sequence_number() - + aRtcpData.cumulative_lost() - *base_seq + 1); + } + } + }; + + auto constructCommonOutboundRtpStats = + [&](RTCOutboundRtpStreamStats& aLocal) { + aLocal.mSsrc = ssrc; + aLocal.mTimestamp.Construct( + pipeline->GetTimestampMaker().GetNow().ToDom()); + aLocal.mId.Construct(localId); + aLocal.mType.Construct(RTCStatsType::Outbound_rtp); + aLocal.mKind = kind; + aLocal.mMediaType.Construct( + kind); // mediaType is the old name for kind. + if (remoteId.Length()) { + aLocal.mRemoteId.Construct(remoteId); + } + }; + + auto constructCommonMediaSourceStats = + [&](RTCMediaSourceStats& aStats) { + nsString id = u"mediasource_"_ns + idstr + trackName; + aStats.mTimestamp.Construct( + pipeline->GetTimestampMaker().GetNow().ToDom()); + aStats.mId.Construct(id); + aStats.mType.Construct(RTCStatsType::Media_source); + aStats.mTrackIdentifier = trackName; + aStats.mKind = kind; + }; + + asAudio.apply([&](auto& aConduit) { + Maybe<webrtc::AudioSendStream::Stats> audioStats = + aConduit->GetSenderStats(); + if (audioStats.isNothing()) { + return; + } + + if (audioStats->packets_sent == 0) { + // 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 receiver data, if present. + // ReceiverReports have less information than SenderReports, so fill + // in what we can. + Maybe<webrtc::ReportBlockData> reportBlockData; + { + if (const auto remoteSsrc = aConduit->GetRemoteSSRC(); + remoteSsrc) { + for (auto& data : audioStats->report_block_datas) { + if (data.source_ssrc() == ssrc && + data.sender_ssrc() == *remoteSsrc) { + reportBlockData.emplace(data); + break; + } + } + } + } + reportBlockData.apply([&](auto& aReportBlockData) { + RTCRemoteInboundRtpStreamStats remote; + constructCommonRemoteInboundRtpStats(remote, aReportBlockData); + if (audioStats->jitter_ms >= 0) { + remote.mJitter.Construct(audioStats->jitter_ms / 1000.0); + } + if (audioStats->packets_lost >= 0) { + remote.mPacketsLost.Construct(audioStats->packets_lost); + } + if (audioStats->rtt_ms >= 0) { + remote.mRoundTripTime.Construct( + static_cast<double>(audioStats->rtt_ms) / 1000.0); + } + remote.mFractionLost.Construct(audioStats->fraction_lost); + remote.mTotalRoundTripTime.Construct( + double(aReportBlockData.sum_rtts().ms()) / 1000); + remote.mRoundTripTimeMeasurements.Construct( + aReportBlockData.num_rtts()); + if (!report->mRemoteInboundRtpStreamStats.AppendElement( + std::move(remote), fallible)) { + mozalloc_handle_oom(0); + } + }); + + // Then, fill in local side (with cross-link to remote only if + // present) + RTCOutboundRtpStreamStats local; + constructCommonOutboundRtpStats(local); + local.mPacketsSent.Construct(audioStats->packets_sent); + local.mBytesSent.Construct(audioStats->payload_bytes_sent); + local.mNackCount.Construct( + audioStats->rtcp_packet_type_counts.nack_packets); + local.mHeaderBytesSent.Construct( + audioStats->header_and_padding_bytes_sent); + local.mRetransmittedPacketsSent.Construct( + audioStats->retransmitted_packets_sent); + local.mRetransmittedBytesSent.Construct( + audioStats->retransmitted_bytes_sent); + /* + * Potential new stats that are now available upstream. + * Note: when we last tried exposing this we were getting + * targetBitrate for audio was ending up as 0. We did not + * investigate why. + local.mTargetBitrate.Construct(audioStats->target_bitrate_bps); + */ + if (!report->mOutboundRtpStreamStats.AppendElement(std::move(local), + fallible)) { + mozalloc_handle_oom(0); + } + + // TODO(bug 1804678): Use RTCAudioSourceStats + RTCMediaSourceStats mediaSourceStats; + constructCommonMediaSourceStats(mediaSourceStats); + if (!report->mMediaSourceStats.AppendElement( + std::move(mediaSourceStats), fallible)) { + mozalloc_handle_oom(0); + } + }); + + asVideo.apply([&](auto& aConduit) { + Maybe<webrtc::VideoSendStream::Stats> videoStats = + aConduit->GetSenderStats(); + if (videoStats.isNothing()) { + return; + } + + Maybe<webrtc::VideoSendStream::StreamStats> streamStats; + auto kv = videoStats->substreams.find(ssrc); + if (kv != videoStats->substreams.end()) { + streamStats = Some(kv->second); + } + + if (!streamStats) { + // 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; + } + + aConduit->GetAssociatedLocalRtxSSRC(ssrc).apply( + [&](const auto rtxSsrc) { + auto kv = videoStats->substreams.find(rtxSsrc); + if (kv != videoStats->substreams.end()) { + streamStats->rtp_stats.Add(kv->second.rtp_stats); + } + }); + + if (streamStats->rtp_stats.first_packet_time == + webrtc::Timestamp::PlusInfinity()) { + return; + } + + // First, fill in remote stat with rtcp receiver data, if present. + // ReceiverReports have less information than SenderReports, so fill + // in what we can. + if (streamStats->report_block_data) { + const webrtc::ReportBlockData& rtcpReportData = + *streamStats->report_block_data; + RTCRemoteInboundRtpStreamStats remote; + remote.mJitter.Construct( + static_cast<double>(rtcpReportData.jitter()) / + webrtc::kVideoPayloadTypeFrequency); + remote.mPacketsLost.Construct(rtcpReportData.cumulative_lost()); + if (rtcpReportData.has_rtt()) { + remote.mRoundTripTime.Construct( + static_cast<double>(rtcpReportData.last_rtt().ms()) / + 1000.0); + } + constructCommonRemoteInboundRtpStats(remote, rtcpReportData); + remote.mTotalRoundTripTime.Construct( + streamStats->report_block_data->sum_rtts().ms() / 1000.0); + remote.mFractionLost.Construct( + static_cast<float>(rtcpReportData.fraction_lost_raw()) / + (1 << 8)); + remote.mRoundTripTimeMeasurements.Construct( + streamStats->report_block_data->num_rtts()); + if (!report->mRemoteInboundRtpStreamStats.AppendElement( + std::move(remote), fallible)) { + mozalloc_handle_oom(0); + } + } + + // Then, fill in local side (with cross-link to remote only if + // present) + RTCOutboundRtpStreamStats local; + constructCommonOutboundRtpStats(local); + local.mPacketsSent.Construct( + streamStats->rtp_stats.transmitted.packets); + local.mBytesSent.Construct( + streamStats->rtp_stats.transmitted.payload_bytes); + local.mNackCount.Construct( + streamStats->rtcp_packet_type_counts.nack_packets); + local.mFirCount.Construct( + streamStats->rtcp_packet_type_counts.fir_packets); + local.mPliCount.Construct( + streamStats->rtcp_packet_type_counts.pli_packets); + local.mFramesEncoded.Construct(streamStats->frames_encoded); + if (streamStats->qp_sum) { + local.mQpSum.Construct(*streamStats->qp_sum); + } + local.mHeaderBytesSent.Construct( + streamStats->rtp_stats.transmitted.header_bytes + + streamStats->rtp_stats.transmitted.padding_bytes); + local.mRetransmittedPacketsSent.Construct( + streamStats->rtp_stats.retransmitted.packets); + local.mRetransmittedBytesSent.Construct( + streamStats->rtp_stats.retransmitted.payload_bytes); + local.mTotalEncodedBytesTarget.Construct( + videoStats->total_encoded_bytes_target); + local.mFrameWidth.Construct(streamStats->width); + local.mFrameHeight.Construct(streamStats->height); + local.mFramesPerSecond.Construct(streamStats->encode_frame_rate); + local.mFramesSent.Construct(streamStats->frames_encoded); + local.mHugeFramesSent.Construct(streamStats->huge_frames_sent); + local.mTotalEncodeTime.Construct( + double(streamStats->total_encode_time_ms) / 1000.); + /* + * Potential new stats that are now available upstream. + local.mTargetBitrate.Construct(videoStats->target_media_bitrate_bps); + */ + if (!report->mOutboundRtpStreamStats.AppendElement(std::move(local), + fallible)) { + mozalloc_handle_oom(0); + } + + RTCVideoSourceStats videoSourceStats; + constructCommonMediaSourceStats(videoSourceStats); + // webrtc::VideoSendStream::Stats does not have width/height. We + // might be able to get this somewhere else? + // videoStats->frames is not documented, but looking at the + // implementation it appears to be the number of frames inputted to + // the encoder, which ought to work. + videoSourceStats.mFrames.Construct(videoStats->frames); + videoSourceStats.mFramesPerSecond.Construct( + videoStats->input_frame_rate); + auto resolution = aConduit->GetLastResolution(); + resolution.apply( + [&](const VideoSessionConduit::Resolution& aResolution) { + videoSourceStats.mWidth.Construct(aResolution.width); + videoSourceStats.mHeight.Construct(aResolution.height); + }); + if (!report->mVideoSourceStats.AppendElement( + std::move(videoSourceStats), fallible)) { + mozalloc_handle_oom(0); + } + }); + } + + return RTCStatsPromise::CreateAndResolve(std::move(report), __func__); + })); + + if (!aSkipIceStats && GetJsepTransceiver().mTransport.mComponents) { + promises.AppendElement(mTransportHandler->GetIceStats( + GetJsepTransceiver().mTransport.mTransportId, + mPipeline->GetTimestampMaker().GetNow().ToDom())); + } + + return promises; +} + +void RTCRtpSender::GetCapabilities(const GlobalObject&, const nsAString& aKind, + Nullable<dom::RTCRtpCapabilities>& aResult) { + PeerConnectionImpl::GetCapabilities(aKind, aResult, sdp::Direction::kSend); +} + +void RTCRtpSender::WarnAboutBadSetParameters(const nsCString& aError) { + nsCString warning( + "WARNING! Invalid setParameters call detected! The good news? Firefox " + "supports sendEncodings in addTransceiver now, so we ask that you switch " + "over to using the parameters code you use for other browsers. Thank you " + "for your patience and support. The specific error was: "); + warning += aError; + mPc->SendWarningToConsole(warning); +} + +nsCString RTCRtpSender::GetEffectiveTLDPlus1() const { + return mPc->GetEffectiveTLDPlus1(); +} + +already_AddRefed<Promise> RTCRtpSender::SetParameters( + const dom::RTCRtpSendParameters& aParameters, ErrorResult& aError) { + dom::RTCRtpSendParameters paramsCopy(aParameters); + // When the setParameters method is called, the user agent MUST run the + // following steps: + // Let parameters be the method's first argument. + // Let sender be the RTCRtpSender object on which setParameters is invoked. + // Let transceiver be the RTCRtpTransceiver object associated with sender + // (i.e.sender is transceiver.[[Sender]]). + + RefPtr<dom::Promise> p = MakePromise(aError); + if (aError.Failed()) { + return nullptr; + } + + if (mPc->IsClosed()) { + p->MaybeRejectWithInvalidStateError("Peer connection is closed"); + return p.forget(); + } + + // If transceiver.[[Stopping]] is true, return a promise rejected with a newly + // created InvalidStateError. + if (mTransceiver->Stopping()) { + p->MaybeRejectWithInvalidStateError( + "This sender's transceiver is stopping/stopped"); + return p.forget(); + } + + // If sender.[[LastReturnedParameters]] is null, return a promise rejected + // with a newly created InvalidStateError. + if (!mLastReturnedParameters.isSome()) { + nsCString error( + "Cannot call setParameters without first calling getParameters"); + if (mAllowOldSetParameters) { + if (!mHaveWarnedBecauseNoGetParameters) { + mHaveWarnedBecauseNoGetParameters = true; + mozilla::glean::rtcrtpsender_setparameters::warn_no_getparameters + .AddToNumerator(1); +#ifdef EARLY_BETA_OR_EARLIER + mozilla::glean::rtcrtpsender_setparameters::blame_no_getparameters + .Get(GetEffectiveTLDPlus1()) + .Add(1); +#endif + } + WarnAboutBadSetParameters(error); + } else { + if (!mHaveFailedBecauseNoGetParameters) { + mHaveFailedBecauseNoGetParameters = true; + mozilla::glean::rtcrtpsender_setparameters::fail_no_getparameters + .AddToNumerator(1); + } + p->MaybeRejectWithInvalidStateError(error); + return p.forget(); + } + } + + // According to the spec, our consistency checking is based on + // [[LastReturnedParameters]], but if we're letting + // [[LastReturnedParameters]]==null slide, we still want to do + // consistency checking on _something_ so we can warn implementers if they + // are messing that up also. Just find something, _anything_, to do that + // checking with. + // TODO(bug 1803388): Remove this stuff once it is no longer needed. + // TODO(bug 1803389): Remove the glean errors once they are no longer needed. + Maybe<RTCRtpSendParameters> oldParams; + if (mAllowOldSetParameters) { + if (mLastReturnedParameters.isSome()) { + oldParams = mLastReturnedParameters; + } else if (mPendingParameters.isSome()) { + oldParams = mPendingParameters; + } else { + oldParams = Some(mParameters); + } + MOZ_ASSERT(oldParams.isSome()); + } else { + oldParams = mLastReturnedParameters; + } + MOZ_ASSERT(oldParams.isSome()); + + // Validate parameters by running the following steps: + // Let encodings be parameters.encodings. + // Let codecs be parameters.codecs. + // Let N be the number of RTCRtpEncodingParameters stored in + // sender.[[SendEncodings]]. + // If any of the following conditions are met, + // return a promise rejected with a newly created InvalidModificationError: + + bool pendingRidChangeFromCompatMode = false; + // encodings.length is different from N. + if (paramsCopy.mEncodings.Length() != oldParams->mEncodings.Length()) { + nsCString error("Cannot change the number of encodings with setParameters"); + if (!mAllowOldSetParameters) { + if (!mHaveFailedBecauseEncodingCountChange) { + mHaveFailedBecauseEncodingCountChange = true; + mozilla::glean::rtcrtpsender_setparameters::fail_length_changed + .AddToNumerator(1); + } + p->MaybeRejectWithInvalidModificationError(error); + return p.forget(); + } + // Make sure we don't use the old rids in SyncToJsep while we wait for the + // queued task below to update mParameters. + pendingRidChangeFromCompatMode = true; + mSimulcastEnvelopeSet = true; + if (!mHaveWarnedBecauseEncodingCountChange) { + mHaveWarnedBecauseEncodingCountChange = true; + mozilla::glean::rtcrtpsender_setparameters::warn_length_changed + .AddToNumerator(1); +#ifdef EARLY_BETA_OR_EARLIER + mozilla::glean::rtcrtpsender_setparameters::blame_length_changed + .Get(GetEffectiveTLDPlus1()) + .Add(1); +#endif + } + WarnAboutBadSetParameters(error); + } else { + // encodings has been re-ordered. + for (size_t i = 0; i < paramsCopy.mEncodings.Length(); ++i) { + const auto& oldEncoding = oldParams->mEncodings[i]; + const auto& newEncoding = paramsCopy.mEncodings[i]; + if (oldEncoding.mRid != newEncoding.mRid) { + nsCString error("Cannot change rid, or reorder encodings"); + if (!mHaveFailedBecauseRidChange) { + mHaveFailedBecauseRidChange = true; + mozilla::glean::rtcrtpsender_setparameters::fail_rid_changed + .AddToNumerator(1); + } + p->MaybeRejectWithInvalidModificationError(error); + return p.forget(); + } + } + } + + // TODO(bug 1803388): Handle this in webidl, once we stop allowing the old + // setParameters style. + if (!paramsCopy.mTransactionId.WasPassed()) { + nsCString error("transactionId is not set!"); + if (!mAllowOldSetParameters) { + if (!mHaveFailedBecauseNoTransactionId) { + mHaveFailedBecauseNoTransactionId = true; + mozilla::glean::rtcrtpsender_setparameters::fail_no_transactionid + .AddToNumerator(1); + } + p->MaybeRejectWithTypeError(error); + return p.forget(); + } + if (!mHaveWarnedBecauseNoTransactionId) { + mHaveWarnedBecauseNoTransactionId = true; + mozilla::glean::rtcrtpsender_setparameters::warn_no_transactionid + .AddToNumerator(1); +#ifdef EARLY_BETA_OR_EARLIER + mozilla::glean::rtcrtpsender_setparameters::blame_no_transactionid + .Get(GetEffectiveTLDPlus1()) + .Add(1); +#endif + } + WarnAboutBadSetParameters(error); + } else if (oldParams->mTransactionId != paramsCopy.mTransactionId) { + // Any parameter in parameters is marked as a Read-only parameter (such as + // RID) and has a value that is different from the corresponding parameter + // value in sender.[[LastReturnedParameters]]. Note that this also applies + // to transactionId. + nsCString error( + "Cannot change transaction id: call getParameters, modify the result, " + "and then call setParameters"); + if (!mHaveFailedBecauseStaleTransactionId) { + mHaveFailedBecauseStaleTransactionId = true; + mozilla::glean::rtcrtpsender_setparameters::fail_stale_transactionid + .AddToNumerator(1); + } + p->MaybeRejectWithInvalidModificationError(error); + return p.forget(); + } + + // This could conceivably happen if we are allowing the old setParameters + // behavior. + if (!paramsCopy.mEncodings.Length()) { + nsCString error("Cannot set an empty encodings array"); + if (!mAllowOldSetParameters) { + if (!mHaveFailedBecauseNoEncodings) { + mHaveFailedBecauseNoEncodings = true; + mozilla::glean::rtcrtpsender_setparameters::fail_no_encodings + .AddToNumerator(1); + } + + p->MaybeRejectWithInvalidModificationError(error); + return p.forget(); + } + // TODO: Add some warning telemetry here + WarnAboutBadSetParameters(error); + // Just don't do this; it's stupid. + paramsCopy.mEncodings = oldParams->mEncodings; + } + + // TODO: Verify remaining read-only parameters + // headerExtensions (bug 1765851) + // rtcp (bug 1765852) + // codecs (bug 1534687) + + // CheckAndRectifyEncodings handles the following steps: + // If transceiver kind is "audio", remove the scaleResolutionDownBy member + // from all encodings that contain one. + // + // If transceiver kind is "video", and any encoding in encodings contains a + // scaleResolutionDownBy member whose value is less than 1.0, return a + // promise rejected with a newly created RangeError. + // + // Verify that each encoding in encodings has a maxFramerate member whose + // value is greater than or equal to 0.0. If one of the maxFramerate values + // does not meet this requirement, return a promise rejected with a newly + // created RangeError. + ErrorResult rv; + CheckAndRectifyEncodings(paramsCopy.mEncodings, mTransceiver->IsVideo(), rv); + if (rv.Failed()) { + if (!mHaveFailedBecauseOtherError) { + mHaveFailedBecauseOtherError = true; + mozilla::glean::rtcrtpsender_setparameters::fail_other.AddToNumerator(1); + } + p->MaybeReject(std::move(rv)); + return p.forget(); + } + + // If transceiver kind is "video", then for each encoding in encodings that + // doesn't contain a scaleResolutionDownBy member, add a + // scaleResolutionDownBy member with the value 1.0. + if (mTransceiver->IsVideo()) { + for (auto& encoding : paramsCopy.mEncodings) { + if (!encoding.mScaleResolutionDownBy.WasPassed()) { + encoding.mScaleResolutionDownBy.Construct(1.0); + } + } + } + + // Let p be a new promise. (see above) + + // In parallel, configure the media stack to use parameters to transmit + // sender.[[SenderTrack]]. + // Right now this is infallible. That may change someday. + + // We need to put this in a member variable, since MaybeUpdateConduit needs it + // This also allows PeerConnectionImpl to detect when there is a pending + // setParameters, which has implcations for the handling of + // setRemoteDescription. + mPendingRidChangeFromCompatMode = pendingRidChangeFromCompatMode; + mPendingParameters = Some(paramsCopy); + uint32_t serialNumber = ++mNumSetParametersCalls; + MaybeUpdateConduit(); + + // If the media stack is successfully configured with parameters, + // queue a task to run the following steps: + GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction( + __func__, + [this, self = RefPtr<RTCRtpSender>(this), p, paramsCopy, serialNumber] { + // Set sender.[[LastReturnedParameters]] to null. + mLastReturnedParameters = Nothing(); + // Set sender.[[SendEncodings]] to parameters.encodings. + mParameters = paramsCopy; + UpdateRestorableEncodings(mParameters.mEncodings); + // Only clear mPendingParameters if it matches; there could have been + // back-to-back calls to setParameters, and we only want to clear this + // if no subsequent setParameters is pending. + if (serialNumber == mNumSetParametersCalls) { + mPendingParameters = Nothing(); + // Ok, nothing has called SyncToJsep while this async task was + // pending. No need for special handling anymore. + mPendingRidChangeFromCompatMode = false; + } + MOZ_ASSERT(mParameters.mEncodings.Length()); + // Resolve p with undefined. + p->MaybeResolveWithUndefined(); + })); + + // Return p. + return p.forget(); +} + +// static +void RTCRtpSender::CheckAndRectifyEncodings( + Sequence<RTCRtpEncodingParameters>& aEncodings, bool aVideo, + ErrorResult& aRv) { + // If any encoding contains a rid member whose value does not conform to the + // grammar requirements specified in Section 10 of [RFC8851], throw a + // TypeError. + for (const auto& encoding : aEncodings) { + if (encoding.mRid.WasPassed()) { + std::string utf8Rid = NS_ConvertUTF16toUTF8(encoding.mRid.Value()).get(); + std::string error; + if (!SdpRidAttributeList::CheckRidValidity(utf8Rid, &error)) { + aRv.ThrowTypeError(nsCString(error)); + return; + } + if (utf8Rid.size() > SdpRidAttributeList::kMaxRidLength) { + std::ostringstream ss; + ss << "Rid can be at most " << SdpRidAttributeList::kMaxRidLength + << " characters long (due to internal limitations)"; + aRv.ThrowTypeError(nsCString(ss.str())); + return; + } + } + } + + if (aEncodings.Length() > 1) { + // If some but not all encodings contain a rid member, throw a TypeError. + // rid must be set if there is more than one encoding + // NOTE: Since rid is read-only, and the number of encodings cannot grow, + // this should never happen in setParameters. + for (const auto& encoding : aEncodings) { + if (!encoding.mRid.WasPassed()) { + aRv.ThrowTypeError("Missing rid"); + return; + } + } + + // If any encoding contains a rid member whose value is the same as that of + // a rid contained in another encoding in sendEncodings, throw a TypeError. + // NOTE: Since rid is read-only, and the number of encodings cannot grow, + // this should never happen in setParameters. + std::set<nsString> uniqueRids; + for (const auto& encoding : aEncodings) { + if (uniqueRids.count(encoding.mRid.Value())) { + aRv.ThrowTypeError("Duplicate rid"); + return; + } + uniqueRids.insert(encoding.mRid.Value()); + } + } + // TODO: ptime/adaptivePtime validation (bug 1733647) + + // If kind is "audio", remove the scaleResolutionDownBy member from all + // encodings that contain one. + if (!aVideo) { + for (auto& encoding : aEncodings) { + if (encoding.mScaleResolutionDownBy.WasPassed()) { + encoding.mScaleResolutionDownBy.Reset(); + } + if (encoding.mMaxFramerate.WasPassed()) { + encoding.mMaxFramerate.Reset(); + } + } + } + + // If any encoding contains a scaleResolutionDownBy member whose value is + // less than 1.0, throw a RangeError. + for (const auto& encoding : aEncodings) { + if (encoding.mScaleResolutionDownBy.WasPassed()) { + if (encoding.mScaleResolutionDownBy.Value() < 1.0f) { + aRv.ThrowRangeError("scaleResolutionDownBy must be >= 1.0"); + return; + } + } + } + + // Verify that the value of each maxFramerate member in sendEncodings that is + // defined is greater than 0.0. If one of the maxFramerate values does not + // meet this requirement, throw a RangeError. + for (const auto& encoding : aEncodings) { + if (encoding.mMaxFramerate.WasPassed()) { + if (encoding.mMaxFramerate.Value() < 0.0f) { + aRv.ThrowRangeError("maxFramerate must be non-negative"); + return; + } + } + } +} + +void RTCRtpSender::GetParameters(RTCRtpSendParameters& aParameters) { + MOZ_ASSERT(mParameters.mEncodings.Length()); + // If sender.[[LastReturnedParameters]] is not null, return + // sender.[[LastReturnedParameters]], and abort these steps. + if (mLastReturnedParameters.isSome()) { + aParameters = *mLastReturnedParameters; + return; + } + + // Let result be a new RTCRtpSendParameters dictionary constructed as follows: + + // transactionId is set to a new unique identifier + aParameters.mTransactionId.Construct(mPc->GenerateUUID()); + + // encodings is set to the value of the [[SendEncodings]] internal slot. + aParameters.mEncodings = mParameters.mEncodings; + + // The headerExtensions sequence is populated based on the header extensions + // that have been negotiated for sending + // TODO(bug 1765851): We do not support this yet + // aParameters.mHeaderExtensions.Construct(); + + // codecs is set to the value of the [[SendCodecs]] internal slot + // TODO(bug 1534687): We do not support this yet + + // rtcp.cname is set to the CNAME of the associated RTCPeerConnection. + // rtcp.reducedSize is set to true if reduced-size RTCP has been negotiated + // for sending, and false otherwise. + // TODO(bug 1765852): We do not support this yet + aParameters.mRtcp.Construct(); + aParameters.mRtcp.Value().mCname.Construct(); + aParameters.mRtcp.Value().mReducedSize.Construct(false); + aParameters.mHeaderExtensions.Construct(); + aParameters.mCodecs.Construct(); + + // Set sender.[[LastReturnedParameters]] to result. + mLastReturnedParameters = Some(aParameters); + + // Queue a task that sets sender.[[LastReturnedParameters]] to null. + GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction( + __func__, [this, self = RefPtr<RTCRtpSender>(this)] { + mLastReturnedParameters = Nothing(); + })); +} + +bool operator==(const RTCRtpEncodingParameters& a1, + const RTCRtpEncodingParameters& a2) { + // webidl does not generate types that are equality comparable + return a1.mActive == a2.mActive && a1.mMaxBitrate == a2.mMaxBitrate && + a1.mMaxFramerate == a2.mMaxFramerate && a1.mPriority == a2.mPriority && + a1.mRid == a2.mRid && + a1.mScaleResolutionDownBy == a2.mScaleResolutionDownBy; +} + +// static +void RTCRtpSender::ApplyJsEncodingToConduitEncoding( + const RTCRtpEncodingParameters& aJsEncoding, + VideoCodecConfig::Encoding* aConduitEncoding) { + aConduitEncoding->active = aJsEncoding.mActive; + if (aJsEncoding.mMaxBitrate.WasPassed()) { + aConduitEncoding->constraints.maxBr = aJsEncoding.mMaxBitrate.Value(); + } + if (aJsEncoding.mMaxFramerate.WasPassed()) { + aConduitEncoding->constraints.maxFps = + Some(aJsEncoding.mMaxFramerate.Value()); + } + if (aJsEncoding.mScaleResolutionDownBy.WasPassed()) { + // Optional does not have a valueOr, despite being based on Maybe + // :( + aConduitEncoding->constraints.scaleDownBy = + aJsEncoding.mScaleResolutionDownBy.Value(); + } else { + aConduitEncoding->constraints.scaleDownBy = 1.0f; + } +} + +void RTCRtpSender::UpdateRestorableEncodings( + const Sequence<RTCRtpEncodingParameters>& aEncodings) { + MOZ_ASSERT(aEncodings.Length()); + + if (GetJsepTransceiver().mSendTrack.GetNegotiatedDetails()) { + // Once initial negotiation completes, we are no longer allowed to restore + // the unicast encoding. + mUnicastEncoding.reset(); + } else if (mParameters.mEncodings.Length() == 1 && + !mParameters.mEncodings[0].mRid.WasPassed()) { + // If we have not completed the initial negotiation, and we currently are + // ridless unicast, we need to save our unicast encoding in case a + // rollback occurs. + mUnicastEncoding = Some(mParameters.mEncodings[0]); + } +} + +Sequence<RTCRtpEncodingParameters> RTCRtpSender::ToSendEncodings( + const std::vector<std::string>& aRids) const { + MOZ_ASSERT(!aRids.empty()); + + Sequence<RTCRtpEncodingParameters> result; + // If sendEncodings is given as input to this algorithm, and is non-empty, + // set the [[SendEncodings]] slot to sendEncodings. + for (const auto& rid : aRids) { + MOZ_ASSERT(!rid.empty()); + RTCRtpEncodingParameters encoding; + encoding.mActive = true; + encoding.mRid.Construct(NS_ConvertUTF8toUTF16(rid.c_str())); + Unused << result.AppendElement(encoding, fallible); + } + + // If sendEncodings is non-empty, set each encoding's scaleResolutionDownBy + // to 2^(length of sendEncodings - encoding index - 1). + if (mTransceiver->IsVideo()) { + double scale = 1.0f; + for (auto it = result.rbegin(); it != result.rend(); ++it) { + it->mScaleResolutionDownBy.Construct(scale); + scale *= 2; + } + } + + return result; +} + +void RTCRtpSender::MaybeGetJsepRids() { + MOZ_ASSERT(!mSimulcastEnvelopeSet); + MOZ_ASSERT(mParameters.mEncodings.Length()); + + auto jsepRids = GetJsepTransceiver().mSendTrack.GetRids(); + if (!jsepRids.empty()) { + UpdateRestorableEncodings(mParameters.mEncodings); + if (jsepRids.size() != 1 || !jsepRids[0].empty()) { + // JSEP is using at least one rid. Stomp our single ridless encoding + mParameters.mEncodings = ToSendEncodings(jsepRids); + } + mSimulcastEnvelopeSet = true; + mSimulcastEnvelopeSetByJSEP = true; + } +} + +Sequence<RTCRtpEncodingParameters> RTCRtpSender::GetMatchingEncodings( + const std::vector<std::string>& aRids) const { + Sequence<RTCRtpEncodingParameters> result; + + if (!aRids.empty() && !aRids[0].empty()) { + // Simulcast, or unicast with rid + for (const auto& encoding : mParameters.mEncodings) { + for (const auto& rid : aRids) { + auto utf16Rid = NS_ConvertUTF8toUTF16(rid.c_str()); + if (!encoding.mRid.WasPassed() || (utf16Rid == encoding.mRid.Value())) { + auto encodingCopy(encoding); + if (!encodingCopy.mRid.WasPassed()) { + encodingCopy.mRid.Construct(NS_ConvertUTF8toUTF16(rid.c_str())); + } + Unused << result.AppendElement(encodingCopy, fallible); + break; + } + } + } + } + + // If we're allowing the old setParameters behavior, we _might_ be able to + // get into this situation even if there were rids above. Be extra careful. + // Under normal circumstances, this just handles the ridless case. + if (!result.Length()) { + // Unicast with no specified rid. Restore mUnicastEncoding, if + // it exists, otherwise pick the first encoding. + if (mUnicastEncoding.isSome()) { + Unused << result.AppendElement(*mUnicastEncoding, fallible); + } else { + Unused << result.AppendElement(mParameters.mEncodings[0], fallible); + } + } + + return result; +} + +void RTCRtpSender::SetStreams( + const Sequence<OwningNonNull<DOMMediaStream>>& aStreams, ErrorResult& aRv) { + if (mPc->IsClosed()) { + aRv.ThrowInvalidStateError( + "Cannot call setStreams if the peer connection is closed"); + return; + } + + SetStreamsImpl(aStreams); + mPc->UpdateNegotiationNeeded(); +} + +void RTCRtpSender::SetStreamsImpl( + const Sequence<OwningNonNull<DOMMediaStream>>& aStreams) { + mStreams.Clear(); + std::set<nsString> ids; + for (const auto& stream : aStreams) { + nsString id; + stream->GetId(id); + if (!ids.count(id)) { + ids.insert(id); + mStreams.AppendElement(stream); + } + } +} + +void RTCRtpSender::GetStreams(nsTArray<RefPtr<DOMMediaStream>>& aStreams) { + aStreams = mStreams.Clone(); +} + +class ReplaceTrackOperation final : public PeerConnectionImpl::Operation { + public: + ReplaceTrackOperation(PeerConnectionImpl* aPc, + const RefPtr<RTCRtpTransceiver>& aTransceiver, + const RefPtr<MediaStreamTrack>& aTrack, + ErrorResult& aError); + NS_DECL_ISUPPORTS_INHERITED + NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(ReplaceTrackOperation, + PeerConnectionImpl::Operation) + + private: + MOZ_CAN_RUN_SCRIPT + RefPtr<dom::Promise> CallImpl(ErrorResult& aError) override; + ~ReplaceTrackOperation() = default; + RefPtr<RTCRtpTransceiver> mTransceiver; + RefPtr<MediaStreamTrack> mNewTrack; +}; + +NS_IMPL_CYCLE_COLLECTION_INHERITED(ReplaceTrackOperation, + PeerConnectionImpl::Operation, mTransceiver, + mNewTrack) + +NS_IMPL_ADDREF_INHERITED(ReplaceTrackOperation, PeerConnectionImpl::Operation) +NS_IMPL_RELEASE_INHERITED(ReplaceTrackOperation, PeerConnectionImpl::Operation) + +NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(ReplaceTrackOperation) +NS_INTERFACE_MAP_END_INHERITING(PeerConnectionImpl::Operation) + +ReplaceTrackOperation::ReplaceTrackOperation( + PeerConnectionImpl* aPc, const RefPtr<RTCRtpTransceiver>& aTransceiver, + const RefPtr<MediaStreamTrack>& aTrack, ErrorResult& aError) + : PeerConnectionImpl::Operation(aPc, aError), + mTransceiver(aTransceiver), + mNewTrack(aTrack) {} + +RefPtr<dom::Promise> ReplaceTrackOperation::CallImpl(ErrorResult& aError) { + RefPtr<RTCRtpSender> sender = mTransceiver->Sender(); + // If transceiver.[[Stopping]] is true, return a promise rejected with a + // newly created InvalidStateError. + if (mTransceiver->Stopped() || mTransceiver->Stopping()) { + RefPtr<dom::Promise> error = sender->MakePromise(aError); + if (aError.Failed()) { + return nullptr; + } + MOZ_LOG(gSenderLog, LogLevel::Debug, + ("%s Cannot call replaceTrack when transceiver is stopping", + __FUNCTION__)); + error->MaybeRejectWithInvalidStateError( + "Cannot call replaceTrack when transceiver is stopping"); + return error; + } + + // Let p be a new promise. + RefPtr<dom::Promise> p = sender->MakePromise(aError); + if (aError.Failed()) { + return nullptr; + } + + if (!sender->SeamlessTrackSwitch(mNewTrack)) { + MOZ_LOG(gSenderLog, LogLevel::Info, + ("%s Could not seamlessly replace track", __FUNCTION__)); + p->MaybeRejectWithInvalidModificationError( + "Could not seamlessly replace track"); + return p; + } + + // Queue a task that runs the following steps: + GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction( + __func__, [p, sender, track = mNewTrack]() MOZ_CAN_RUN_SCRIPT_BOUNDARY { + // If connection.[[IsClosed]] is true, abort these steps. + // Set sender.[[SenderTrack]] to withTrack. + if (sender->SetSenderTrackWithClosedCheck(track)) { + // Resolve p with undefined. + p->MaybeResolveWithUndefined(); + } + })); + + // Return p. + return p; +} + +already_AddRefed<dom::Promise> RTCRtpSender::ReplaceTrack( + dom::MediaStreamTrack* aWithTrack, ErrorResult& aError) { + // If withTrack is non-null and withTrack.kind differs from the transceiver + // kind of transceiver, return a promise rejected with a newly created + // TypeError. + if (aWithTrack) { + nsString newKind; + aWithTrack->GetKind(newKind); + nsString oldKind; + mTransceiver->GetKind(oldKind); + if (newKind != oldKind) { + RefPtr<dom::Promise> error = MakePromise(aError); + if (aError.Failed()) { + return nullptr; + } + error->MaybeRejectWithTypeError( + "Cannot replaceTrack with a different kind!"); + return error.forget(); + } + } + + MOZ_LOG(gSenderLog, LogLevel::Debug, + ("%s[%s]: %s (%p to %p)", mPc->GetHandle().c_str(), GetMid().c_str(), + __FUNCTION__, mSenderTrack.get(), aWithTrack)); + + // Return the result of chaining the following steps to connection's + // operations chain: + RefPtr<PeerConnectionImpl::Operation> op = + new ReplaceTrackOperation(mPc, mTransceiver, aWithTrack, aError); + if (aError.Failed()) { + return nullptr; + } + // Static analysis forces us to use a temporary. + auto pc = mPc; + return pc->Chain(op, aError); +} + +nsPIDOMWindowInner* RTCRtpSender::GetParentObject() const { return mWindow; } + +already_AddRefed<dom::Promise> RTCRtpSender::MakePromise( + ErrorResult& aError) const { + return mPc->MakePromise(aError); +} + +bool RTCRtpSender::SeamlessTrackSwitch( + const RefPtr<MediaStreamTrack>& aWithTrack) { + // We do not actually update mSenderTrack here! Spec says that happens in a + // queued task after this is done (this happens in + // SetSenderTrackWithClosedCheck). + + mPipeline->SetTrack(aWithTrack); + + MaybeUpdateConduit(); + + // There may eventually be cases where a renegotiation is necessary to switch. + return true; +} + +void RTCRtpSender::SetTrack(const RefPtr<MediaStreamTrack>& aTrack) { + // Used for RTCPeerConnection.removeTrack and RTCPeerConnection.addTrack + if (mTransceiver->Stopping()) { + return; + } + mSenderTrack = aTrack; + SeamlessTrackSwitch(aTrack); + if (aTrack) { + // RFC says (in the section on remote rollback): + // However, an RtpTransceiver MUST NOT be removed if a track was attached + // to the RtpTransceiver via the addTrack method. + mSenderTrackSetByAddTrack = true; + } +} + +bool RTCRtpSender::SetSenderTrackWithClosedCheck( + const RefPtr<MediaStreamTrack>& aTrack) { + if (!mPc->IsClosed()) { + mSenderTrack = aTrack; + return true; + } + + return false; +} + +void RTCRtpSender::Shutdown() { + MOZ_ASSERT(NS_IsMainThread()); + mWatchManager.Shutdown(); + mPipeline->Shutdown(); + mPipeline = nullptr; + if (mTransform) { + mTransform->GetProxy().SetSender(nullptr); + } +} + +void RTCRtpSender::BreakCycles() { + mWindow = nullptr; + mPc = nullptr; + mSenderTrack = nullptr; + mTransceiver = nullptr; + mStreams.Clear(); + mDtmf = nullptr; +} + +void RTCRtpSender::Unlink() { + if (mTransceiver) { + mTransceiver->Unlink(); + } +} + +void RTCRtpSender::UpdateTransport() { + MOZ_ASSERT(NS_IsMainThread()); + if (!mHaveSetupTransport) { + mPipeline->SetLevel(GetJsepTransceiver().GetLevel()); + mHaveSetupTransport = true; + } + + mPipeline->UpdateTransport_m(GetJsepTransceiver().mTransport.mTransportId, + nullptr); +} + +void RTCRtpSender::MaybeUpdateConduit() { + // NOTE(pkerr) - the Call API requires the both local_ssrc and remote_ssrc be + // set to a non-zero value or the CreateVideo...Stream call will fail. + if (NS_WARN_IF(GetJsepTransceiver().mSendTrack.GetSsrcs().empty())) { + MOZ_ASSERT( + false, + "No local ssrcs! This is a bug in the jsep engine, and should never " + "happen!"); + return; + } + + if (!mPipeline) { + return; + } + + bool wasTransmitting = mTransmitting; + + if (mPipeline->mConduit->type() == MediaSessionConduit::VIDEO) { + Maybe<VideoConfig> newConfig = GetNewVideoConfig(); + if (newConfig.isSome()) { + ApplyVideoConfig(*newConfig); + } + } else { + Maybe<AudioConfig> newConfig = GetNewAudioConfig(); + if (newConfig.isSome()) { + ApplyAudioConfig(*newConfig); + } + } + + if (!mSenderTrack && !wasTransmitting && mTransmitting) { + MOZ_LOG(gSenderLog, LogLevel::Debug, + ("%s[%s]: %s Starting transmit conduit without send track!", + mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__)); + } +} + +void RTCRtpSender::SyncFromJsep(const JsepTransceiver& aJsepTransceiver) { + if (!mSimulcastEnvelopeSet) { + // JSEP is establishing the simulcast envelope for the first time, right now + // This is the addTrack (or addTransceiver without sendEncodings) case. + MaybeGetJsepRids(); + } else if (!aJsepTransceiver.mSendTrack.GetNegotiatedDetails() || + !aJsepTransceiver.mSendTrack.IsInHaveRemote()) { + // Spec says that we do not update our encodings until we're in stable, + // _unless_ this is the first negotiation. + std::vector<std::string> rids = aJsepTransceiver.mSendTrack.GetRids(); + if (mSimulcastEnvelopeSetByJSEP && rids.empty()) { + // JSEP previously set the simulcast envelope, but now it has no opinion + // regarding unicast/simulcast. This can only happen on rollback of the + // initial remote offer. + mParameters.mEncodings = GetMatchingEncodings(rids); + MOZ_ASSERT(mParameters.mEncodings.Length()); + mSimulcastEnvelopeSetByJSEP = false; + mSimulcastEnvelopeSet = false; + } else if (!rids.empty()) { + // JSEP has an opinion on the simulcast envelope, which trumps anything + // we have already. + mParameters.mEncodings = GetMatchingEncodings(rids); + MOZ_ASSERT(mParameters.mEncodings.Length()); + } + } + + MaybeUpdateConduit(); +} + +void RTCRtpSender::SyncToJsep(JsepTransceiver& aJsepTransceiver) const { + std::vector<std::string> streamIds; + for (const auto& stream : mStreams) { + nsString wideStreamId; + stream->GetId(wideStreamId); + std::string streamId = NS_ConvertUTF16toUTF8(wideStreamId).get(); + MOZ_ASSERT(!streamId.empty()); + streamIds.push_back(streamId); + } + + aJsepTransceiver.mSendTrack.UpdateStreamIds(streamIds); + + if (mSimulcastEnvelopeSet) { + std::vector<std::string> rids; + Maybe<RTCRtpSendParameters> parameters; + if (mPendingRidChangeFromCompatMode) { + // *sigh* If we have just let a setParameters change our rids, but we have + // not yet updated mParameters because the queued task hasn't run yet, + // we want to set the _new_ rids on the JsepTrack. So, we are forced to + // grab them from mPendingParameters. + parameters = mPendingParameters; + } else { + parameters = Some(mParameters); + } + for (const auto& encoding : parameters->mEncodings) { + if (encoding.mRid.WasPassed()) { + rids.push_back(NS_ConvertUTF16toUTF8(encoding.mRid.Value()).get()); + } else { + rids.push_back(""); + } + } + + aJsepTransceiver.mSendTrack.SetRids(rids); + } + + if (mTransceiver->IsVideo()) { + aJsepTransceiver.mSendTrack.SetMaxEncodings(webrtc::kMaxSimulcastStreams); + } else { + aJsepTransceiver.mSendTrack.SetMaxEncodings(1); + } + + if (mSenderTrackSetByAddTrack) { + aJsepTransceiver.SetOnlyExistsBecauseOfSetRemote(false); + } +} + +Maybe<RTCRtpSender::VideoConfig> RTCRtpSender::GetNewVideoConfig() { + // It is possible for SDP to signal that there is a send track, but there not + // actually be a send track, according to the specification; all that needs to + // happen is for the transceiver to be configured to send... + if (!GetJsepTransceiver().mSendTrack.GetNegotiatedDetails()) { + return Nothing(); + } + + VideoConfig oldConfig; + oldConfig.mSsrcs = mSsrcs; + oldConfig.mLocalRtpExtensions = mLocalRtpExtensions; + oldConfig.mCname = mCname; + oldConfig.mTransmitting = mTransmitting; + oldConfig.mVideoRtxSsrcs = mVideoRtxSsrcs; + oldConfig.mVideoCodec = mVideoCodec; + oldConfig.mVideoRtpRtcpConfig = mVideoRtpRtcpConfig; + oldConfig.mVideoCodecMode = mVideoCodecMode; + + VideoConfig newConfig(oldConfig); + + UpdateBaseConfig(&newConfig); + + newConfig.mVideoRtxSsrcs = GetJsepTransceiver().mSendTrack.GetRtxSsrcs(); + + const JsepTrackNegotiatedDetails details( + *GetJsepTransceiver().mSendTrack.GetNegotiatedDetails()); + + if (mSenderTrack) { + RefPtr<mozilla::dom::VideoStreamTrack> videotrack = + mSenderTrack->AsVideoStreamTrack(); + + if (!videotrack) { + MOZ_CRASH( + "In ConfigureVideoCodecMode, mSenderTrack is not video! This should " + "never happen!"); + } + + dom::MediaSourceEnum source = videotrack->GetSource().GetMediaSource(); + switch (source) { + case dom::MediaSourceEnum::Browser: + case dom::MediaSourceEnum::Screen: + case dom::MediaSourceEnum::Window: + case dom::MediaSourceEnum::Application: + newConfig.mVideoCodecMode = webrtc::VideoCodecMode::kScreensharing; + break; + + case dom::MediaSourceEnum::Camera: + case dom::MediaSourceEnum::Other: + // Other is used by canvas capture, which we treat as realtime video. + // This seems debatable, but we've been doing it this way for a long + // time, so this is likely fine. + newConfig.mVideoCodecMode = webrtc::VideoCodecMode::kRealtimeVideo; + break; + + case dom::MediaSourceEnum::Microphone: + case dom::MediaSourceEnum::AudioCapture: + case dom::MediaSourceEnum::EndGuard_: + MOZ_ASSERT(false); + break; + } + } + + 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(gSenderLog, LogLevel::Error, + ("%s[%s]: %s No video codecs were negotiated (send).", + mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__)); + return Nothing(); + } + + newConfig.mVideoCodec = Some(configs[0]); + // Spec says that we start using new parameters right away, _before_ we + // update the parameters that are visible to JS (ie; mParameters). + const RTCRtpSendParameters& parameters = + mPendingParameters.isSome() ? *mPendingParameters : mParameters; + for (VideoCodecConfig::Encoding& conduitEncoding : + newConfig.mVideoCodec->mEncodings) { + for (const RTCRtpEncodingParameters& jsEncoding : parameters.mEncodings) { + std::string rid; + if (jsEncoding.mRid.WasPassed()) { + rid = NS_ConvertUTF16toUTF8(jsEncoding.mRid.Value()).get(); + } + if (conduitEncoding.rid == rid) { + ApplyJsEncodingToConduitEncoding(jsEncoding, &conduitEncoding); + break; + } + } + } + + if (!mHaveLoggedUlpfecInfo) { + bool ulpfecNegotiated = false; + for (const auto& codec : configs) { + if (nsCRT::strcasestr(codec.mName.c_str(), "ulpfec")) { + ulpfecNegotiated = true; + } + } + mozilla::glean::codec_stats::ulpfec_negotiated + .Get(ulpfecNegotiated ? "negotiated"_ns : "not_negotiated"_ns) + .Add(1); + mHaveLoggedUlpfecInfo = true; + } + + // Log codec information we are tracking + if (!mHaveLoggedOtherFec && + !GetJsepTransceiver().mSendTrack.GetFecCodecName().empty()) { + mozilla::glean::codec_stats::other_fec_signaled + .Get(nsDependentCString( + GetJsepTransceiver().mSendTrack.GetFecCodecName().c_str())) + .Add(1); + mHaveLoggedOtherFec = true; + } + if (!mHaveLoggedVideoPreferredCodec && + !GetJsepTransceiver().mSendTrack.GetVideoPreferredCodec().empty()) { + mozilla::glean::codec_stats::video_preferred_codec + .Get(nsDependentCString( + GetJsepTransceiver().mSendTrack.GetVideoPreferredCodec().c_str())) + .Add(1); + mHaveLoggedVideoPreferredCodec = true; + } + + newConfig.mVideoRtpRtcpConfig = Some(details.GetRtpRtcpConfig()); + + if (newConfig == oldConfig) { + MOZ_LOG(gSenderLog, LogLevel::Debug, + ("%s[%s]: %s No change in video config", mPc->GetHandle().c_str(), + GetMid().c_str(), __FUNCTION__)); + return Nothing(); + } + + if (newConfig.mVideoCodec.isSome()) { + MOZ_ASSERT(newConfig.mSsrcs.size() == + newConfig.mVideoCodec->mEncodings.size()); + } + return Some(newConfig); +} + +Maybe<RTCRtpSender::AudioConfig> RTCRtpSender::GetNewAudioConfig() { + AudioConfig oldConfig; + oldConfig.mSsrcs = mSsrcs; + oldConfig.mLocalRtpExtensions = mLocalRtpExtensions; + oldConfig.mCname = mCname; + oldConfig.mTransmitting = mTransmitting; + oldConfig.mAudioCodec = mAudioCodec; + + AudioConfig newConfig(oldConfig); + + UpdateBaseConfig(&newConfig); + + if (GetJsepTransceiver().mSendTrack.GetNegotiatedDetails() && + GetJsepTransceiver().mSendTrack.GetActive()) { + const auto& details( + *GetJsepTransceiver().mSendTrack.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(gSenderLog, LogLevel::Error, + ("%s[%s]: %s No audio codecs were negotiated (send)", + mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__)); + return Nothing(); + } + + std::vector<AudioCodecConfig> dtmfConfigs; + std::copy_if( + configs.begin(), configs.end(), std::back_inserter(dtmfConfigs), + [](const auto& value) { return value.mName == "telephone-event"; }); + + const AudioCodecConfig& sendCodec = configs[0]; + + if (!dtmfConfigs.empty()) { + // There is at least one telephone-event codec. + // We primarily choose the codec whose frequency matches the send codec. + // Secondarily we choose the one with the lowest frequency. + auto dtmfIterator = + std::find_if(dtmfConfigs.begin(), dtmfConfigs.end(), + [&sendCodec](const auto& dtmfCodec) { + return dtmfCodec.mFreq == sendCodec.mFreq; + }); + if (dtmfIterator == dtmfConfigs.end()) { + dtmfIterator = std::min_element( + dtmfConfigs.begin(), dtmfConfigs.end(), + [](const auto& a, const auto& b) { return a.mFreq < b.mFreq; }); + } + MOZ_ASSERT(dtmfIterator != dtmfConfigs.end()); + newConfig.mDtmfPt = dtmfIterator->mType; + newConfig.mDtmfFreq = dtmfIterator->mFreq; + } + + newConfig.mAudioCodec = Some(sendCodec); + } + + if (!mHaveLoggedAudioPreferredCodec && + !GetJsepTransceiver().mSendTrack.GetAudioPreferredCodec().empty()) { + mozilla::glean::codec_stats::audio_preferred_codec + .Get(nsDependentCString( + GetJsepTransceiver().mSendTrack.GetAudioPreferredCodec().c_str())) + .Add(1); + mHaveLoggedAudioPreferredCodec = true; + } + + if (newConfig == oldConfig) { + MOZ_LOG(gSenderLog, LogLevel::Debug, + ("%s[%s]: %s No change in audio config", mPc->GetHandle().c_str(), + GetMid().c_str(), __FUNCTION__)); + return Nothing(); + } + + return Some(newConfig); +} + +void RTCRtpSender::UpdateBaseConfig(BaseConfig* aConfig) { + aConfig->mSsrcs = GetJsepTransceiver().mSendTrack.GetSsrcs(); + aConfig->mCname = GetJsepTransceiver().mSendTrack.GetCNAME(); + + if (GetJsepTransceiver().mSendTrack.GetNegotiatedDetails() && + GetJsepTransceiver().mSendTrack.GetActive()) { + const auto& details( + *GetJsepTransceiver().mSendTrack.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); + }); + aConfig->mLocalRtpExtensions = extmaps; + } + } + // RTCRtpTransceiver::IsSending is updated after negotiation completes, in a + // queued task (which we may be in right now). Don't use + // JsepTrack::GetActive, because that updates before the queued task, which + // is too early for some of the things we interact with here (eg; + // RTCDTMFSender). + aConfig->mTransmitting = mTransceiver->IsSending(); +} + +void RTCRtpSender::ApplyVideoConfig(const VideoConfig& aConfig) { + if (aConfig.mVideoCodec.isSome()) { + MOZ_ASSERT(aConfig.mSsrcs.size() == aConfig.mVideoCodec->mEncodings.size()); + } + + mSsrcs = aConfig.mSsrcs; + mCname = aConfig.mCname; + mLocalRtpExtensions = aConfig.mLocalRtpExtensions; + + mVideoRtxSsrcs = aConfig.mVideoRtxSsrcs; + mVideoCodec = aConfig.mVideoCodec; + mVideoRtpRtcpConfig = aConfig.mVideoRtpRtcpConfig; + mVideoCodecMode = aConfig.mVideoCodecMode; + + mTransmitting = aConfig.mTransmitting; +} + +void RTCRtpSender::ApplyAudioConfig(const AudioConfig& aConfig) { + mSsrcs = aConfig.mSsrcs; + mCname = aConfig.mCname; + mLocalRtpExtensions = aConfig.mLocalRtpExtensions; + + mAudioCodec = aConfig.mAudioCodec; + + if (aConfig.mDtmfPt >= 0) { + mDtmf->SetPayloadType(aConfig.mDtmfPt, aConfig.mDtmfFreq); + } + + mTransmitting = aConfig.mTransmitting; +} + +void RTCRtpSender::Stop() { + MOZ_ASSERT(mTransceiver->Stopping()); + mTransmitting = false; +} + +bool RTCRtpSender::HasTrack(const dom::MediaStreamTrack* aTrack) const { + if (!mSenderTrack) { + return false; + } + + if (!aTrack) { + return true; + } + + return mSenderTrack.get() == aTrack; +} + +RefPtr<MediaPipelineTransmit> RTCRtpSender::GetPipeline() const { + return mPipeline; +} + +std::string RTCRtpSender::GetMid() const { return mTransceiver->GetMidAscii(); } + +JsepTransceiver& RTCRtpSender::GetJsepTransceiver() { + return mTransceiver->GetJsepTransceiver(); +} + +void RTCRtpSender::UpdateDtmfSender() { + if (!mDtmf) { + return; + } + + if (mTransmitting) { + return; + } + + mDtmf->StopPlayout(); +} + +void RTCRtpSender::SetTransform(RTCRtpScriptTransform* aTransform, + ErrorResult& aError) { + if (aTransform == mTransform.get()) { + // Ok... smile and nod + // TODO: Depending on spec, this might throw + // https://github.com/w3c/webrtc-encoded-transform/issues/189 + return; + } + + if (aTransform && aTransform->IsClaimed()) { + aError.ThrowInvalidStateError("transform has already been used elsewhere"); + return; + } + + // Seamless switch for frames + if (aTransform) { + mFrameTransformerProxy = &aTransform->GetProxy(); + } else { + mFrameTransformerProxy = nullptr; + } + + if (mTransform) { + mTransform->GetProxy().SetSender(nullptr); + } + + mTransform = const_cast<RTCRtpScriptTransform*>(aTransform); + + if (mTransform) { + mTransform->GetProxy().SetSender(this); + mTransform->SetClaimed(); + } +} + +bool RTCRtpSender::GenerateKeyFrame(const Maybe<std::string>& aRid) { + if (!mTransform || !mPipeline || !mSenderTrack) { + return false; + } + + mPipeline->mConduit->AsVideoSessionConduit().apply([&](const auto& conduit) { + conduit->GenerateKeyFrame(aRid, &mTransform->GetProxy()); + }); + return true; +} + +} // namespace mozilla::dom + +#undef LOGTAG |