werift 0.14.4-debug0 → 0.15.0-alpha.0
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- package/lib/common/src/binary.d.ts +0 -1
- package/lib/common/src/binary.js +1 -2
- package/lib/common/src/binary.js.map +1 -1
- package/lib/dtls/src/context/cipher.js +5 -2
- package/lib/dtls/src/context/cipher.js.map +1 -1
- package/lib/dtls/src/handshake/extensions/useSrtp.js +5 -2
- package/lib/dtls/src/handshake/extensions/useSrtp.js.map +1 -1
- package/lib/ice/src/candidate.js +5 -2
- package/lib/ice/src/candidate.js.map +1 -1
- package/lib/ice/src/ice.d.ts +2 -2
- package/lib/ice/src/ice.js +6 -5
- package/lib/ice/src/ice.js.map +1 -1
- package/lib/ice/src/stun/attributes.js +5 -2
- package/lib/ice/src/stun/attributes.js.map +1 -1
- package/lib/rtp/src/rtcp/rr.js +5 -2
- package/lib/rtp/src/rtcp/rr.js.map +1 -1
- package/lib/rtp/src/rtcp/rtpfb/nack.js +6 -3
- package/lib/rtp/src/rtcp/rtpfb/nack.js.map +1 -1
- package/lib/rtp/src/rtcp/rtpfb/twcc.js +6 -6
- package/lib/rtp/src/rtcp/rtpfb/twcc.js.map +1 -1
- package/lib/rtp/src/rtcp/sr.js +5 -2
- package/lib/rtp/src/rtcp/sr.js.map +1 -1
- package/lib/rtp/src/srtp/cipher/ctr.js +5 -2
- package/lib/rtp/src/srtp/cipher/ctr.js.map +1 -1
- package/lib/rtp/src/srtp/cipher/gcm.js +6 -3
- package/lib/rtp/src/srtp/cipher/gcm.js.map +1 -1
- package/lib/sctp/src/param.js +5 -2
- package/lib/sctp/src/param.js.map +1 -1
- package/lib/sctp/src/sctp.js +5 -5
- package/lib/sctp/src/sctp.js.map +1 -1
- package/lib/webrtc/src/media/receiver/nack.js +2 -2
- package/lib/webrtc/src/media/receiver/nack.js.map +1 -1
- package/lib/webrtc/src/media/rtpReceiver.d.ts +3 -2
- package/lib/webrtc/src/media/rtpReceiver.js +4 -2
- package/lib/webrtc/src/media/rtpReceiver.js.map +1 -1
- package/lib/webrtc/src/media/rtpSender.d.ts +4 -2
- package/lib/webrtc/src/media/rtpSender.js +15 -7
- package/lib/webrtc/src/media/rtpSender.js.map +1 -1
- package/lib/webrtc/src/media/rtpTransceiver.d.ts +7 -6
- package/lib/webrtc/src/media/rtpTransceiver.js +8 -3
- package/lib/webrtc/src/media/rtpTransceiver.js.map +1 -1
- package/lib/webrtc/src/nonstandard/recorder/writer/webm.js +1 -1
- package/lib/webrtc/src/nonstandard/recorder/writer/webm.js.map +1 -1
- package/lib/webrtc/src/peerConnection.d.ts +10 -10
- package/lib/webrtc/src/peerConnection.js +324 -188
- package/lib/webrtc/src/peerConnection.js.map +1 -1
- package/lib/webrtc/src/sdp.js +5 -2
- package/lib/webrtc/src/sdp.js.map +1 -1
- package/lib/webrtc/src/transport/dtls.d.ts +3 -2
- package/lib/webrtc/src/transport/dtls.js +3 -0
- package/lib/webrtc/src/transport/dtls.js.map +1 -1
- package/lib/webrtc/src/transport/sctp.d.ts +5 -3
- package/lib/webrtc/src/transport/sctp.js +46 -34
- package/lib/webrtc/src/transport/sctp.js.map +1 -1
- package/package.json +2 -2
- package/src/media/receiver/nack.ts +1 -1
- package/src/media/rtpReceiver.ts +6 -5
- package/src/media/rtpSender.ts +18 -9
- package/src/media/rtpTransceiver.ts +13 -6
- package/src/nonstandard/recorder/writer/webm.ts +1 -1
- package/src/peerConnection.ts +386 -223
- package/src/sdp.ts +1 -1
- package/src/transport/dtls.ts +4 -1
- package/src/transport/sctp.ts +53 -33
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@@ -40,9 +40,8 @@ const log = (0, debug_1.default)("werift:packages/webrtc/src/media/rtpSender.ts"
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const RTP_HISTORY_SIZE = 128;
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const RTT_ALPHA = 0.85;
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class RTCRtpSender {
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constructor(trackOrKind
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constructor(trackOrKind) {
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this.trackOrKind = trackOrKind;
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this.dtlsTransport = dtlsTransport;
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this.type = "sender";
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this.kind = typeof this.trackOrKind === "string"
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? this.trackOrKind
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@@ -69,14 +68,10 @@ class RTCRtpSender {
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this.timestampOffset = 0;
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this.seqOffset = 0;
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this.rtpCache = [];
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this.dtlsDisposer = [];
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this.stopped = false;
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this.rtcpRunning = false;
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this.rtcpCancel = new AbortController();
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dtlsTransport.onStateChange.subscribe((state) => {
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if (state === "connected") {
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this.onReady.execute();
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}
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});
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if (trackOrKind instanceof track_1.MediaStreamTrack) {
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if (trackOrKind.streamId) {
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this.streamId = trackOrKind.streamId;
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@@ -84,6 +79,19 @@ class RTCRtpSender {
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this.registerTrack(trackOrKind);
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}
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}
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setDtlsTransport(dtlsTransport) {
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if (this.dtlsTransport) {
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this.dtlsDisposer.forEach((dispose) => dispose());
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}
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this.dtlsTransport = dtlsTransport;
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this.dtlsDisposer = [
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this.dtlsTransport.onStateChange.subscribe((state) => {
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if (state === "connected") {
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this.onReady.execute();
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}
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}).unSubscribe,
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];
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}
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get redDistance() {
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return this._redDistance;
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}
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@@ -1 +1 @@
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1
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-
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{ randomBytes } from \"crypto\";\nimport debug from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport * as uuid from \"uuid\";\n\nimport {\n bufferWriter,\n random16,\n uint16Add,\n uint32Add,\n} from \"../../../common/src\";\nimport {\n Extension,\n GenericNack,\n PictureLossIndication,\n ReceiverEstimatedMaxBitrate,\n RedEncoder,\n RtcpPacket,\n RtcpPayloadSpecificFeedback,\n RtcpRrPacket,\n RtcpSenderInfo,\n RtcpSourceDescriptionPacket,\n RtcpSrPacket,\n RtcpTransportLayerFeedback,\n RtpHeader,\n RtpPacket,\n SourceDescriptionChunk,\n SourceDescriptionItem,\n TransportWideCC,\n} from \"../../../rtp/src\";\nimport { codecParametersFromString } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, milliTime, ntpTime } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport {\n RTCRtpCodecParameters,\n RTCRtpHeaderExtensionParameters,\n RTCRtpSendParameters,\n} from \"./parameters\";\nimport { SenderBandwidthEstimator, SentInfo } from \"./sender/senderBWE\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpSender.ts\");\n\nconst RTP_HISTORY_SIZE = 128;\nconst RTT_ALPHA = 0.85;\n\nexport class RTCRtpSender {\n readonly type = \"sender\";\n readonly kind =\n typeof this.trackOrKind === \"string\"\n ? this.trackOrKind\n : this.trackOrKind.kind;\n readonly ssrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n readonly rtxSsrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n streamId = uuid.v4();\n readonly trackId = uuid.v4();\n readonly onReady = new Event();\n readonly onRtcp = new Event<[RtcpPacket]>();\n readonly onPictureLossIndication = new Event<[]>();\n readonly onGenericNack = new Event<[GenericNack]>();\n readonly senderBWE = new SenderBandwidthEstimator();\n\n private cname?: string;\n private mid?: string;\n private rtpStreamId?: string;\n private repairedRtpStreamId?: string;\n private rtxPayloadType?: number;\n private rtxSequenceNumber = random16();\n redRedundantPayloadType?: number;\n private _redDistance = 2;\n redEncoder = new RedEncoder(this._redDistance);\n private headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n private disposeTrack?: () => void;\n\n // # stats\n private lsr?: number;\n private lsrTime: number = Date.now() / 1000;\n private ntpTimestamp = 0n;\n private rtpTimestamp = 0;\n private octetCount = 0;\n private packetCount = 0;\n private rtt?: number;\n receiverEstimatedMaxBitrate: bigint = 0n;\n\n // rtp\n private sequenceNumber?: number;\n private timestamp?: number;\n private timestampOffset = 0;\n private seqOffset = 0;\n private rtpCache: RtpPacket[] = [];\n codec?: RTCRtpCodecParameters;\n\n track?: MediaStreamTrack;\n stopped = false;\n rtcpRunning = false;\n private rtcpCancel = new AbortController();\n\n constructor(\n public trackOrKind: Kind | MediaStreamTrack,\n public dtlsTransport: RTCDtlsTransport\n ) {\n dtlsTransport.onStateChange.subscribe((state) => {\n if (state === \"connected\") {\n this.onReady.execute();\n }\n });\n if (trackOrKind instanceof MediaStreamTrack) {\n if (trackOrKind.streamId) {\n this.streamId = trackOrKind.streamId;\n }\n this.registerTrack(trackOrKind);\n }\n }\n\n get redDistance() {\n return this._redDistance;\n }\n set redDistance(n: number) {\n this._redDistance = n;\n this.redEncoder.distance = n;\n }\n\n prepareSend(params: RTCRtpSendParameters) {\n this.cname = params.rtcp?.cname;\n this.mid = params.muxId;\n this.headerExtensions = params.headerExtensions;\n this.rtpStreamId = params.rtpStreamId;\n this.repairedRtpStreamId = params.repairedRtpStreamId;\n\n this.codec = params.codecs[0];\n if (this.track) {\n this.track.codec = this.codec;\n }\n\n params.codecs.forEach((codec) => {\n const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n if (\n codec.name.toLowerCase() === \"rtx\" &&\n codecParams[\"apt\"] === this.codec?.payloadType\n ) {\n this.rtxPayloadType = codec.payloadType;\n }\n if (codec.name.toLowerCase() === \"red\") {\n this.redRedundantPayloadType = Number(\n (codec.parameters ?? \"\").split(\"/\")[0]\n );\n }\n });\n }\n\n registerTrack(track: MediaStreamTrack) {\n if (track.stopped) throw new Error(\"track is ended\");\n\n if (this.disposeTrack) {\n this.disposeTrack();\n }\n\n track.id = this.trackId;\n\n const { unSubscribe } = track.onReceiveRtp.subscribe(async (rtp) => {\n await this.sendRtp(rtp);\n });\n this.track = track;\n this.disposeTrack = unSubscribe;\n\n if (this.codec) {\n track.codec = this.codec;\n }\n\n track.onSourceChanged.subscribe((header) => {\n this.replaceRTP(header);\n });\n }\n\n async replaceTrack(track: MediaStreamTrack | null) {\n if (track === null) {\n // todo impl\n return;\n }\n\n if (track.stopped) throw new Error(\"track is ended\");\n\n if (this.sequenceNumber != undefined) {\n const header =\n track.header || (await track.onReceiveRtp.asPromise())[0].header;\n\n this.replaceRTP(header);\n }\n\n this.registerTrack(track);\n log(\"replaceTrack\", \"ssrc\", track.ssrc, \"rid\", track.rid);\n }\n\n stop() {\n this.stopped = true;\n this.rtcpRunning = false;\n this.rtcpCancel.abort();\n\n this.track = undefined;\n }\n\n async runRtcp() {\n if (this.rtcpRunning || this.stopped) return;\n this.rtcpRunning = true;\n\n try {\n while (this.rtcpRunning) {\n await setTimeout(500 + Math.random() * 1000, undefined, {\n signal: this.rtcpCancel.signal,\n });\n\n const packets: RtcpPacket[] = [\n new RtcpSrPacket({\n ssrc: this.ssrc,\n senderInfo: new RtcpSenderInfo({\n ntpTimestamp: this.ntpTimestamp,\n rtpTimestamp: this.rtpTimestamp,\n packetCount: this.packetCount,\n octetCount: this.octetCount,\n }),\n }),\n ];\n this.lsr = compactNtp(this.ntpTimestamp);\n this.lsrTime = Date.now() / 1000;\n\n if (this.cname) {\n packets.push(\n new RtcpSourceDescriptionPacket({\n chunks: [\n new SourceDescriptionChunk({\n source: this.ssrc,\n items: [\n new SourceDescriptionItem({ type: 1, text: this.cname }),\n ],\n }),\n ],\n })\n );\n }\n\n try {\n await this.dtlsTransport.sendRtcp(packets);\n } catch (error) {\n log(\"sendRtcp failed\", error);\n await setTimeout(500 + Math.random() * 1000);\n }\n }\n } catch (error) {}\n }\n\n replaceRTP({\n sequenceNumber,\n timestamp,\n }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">) {\n if (this.sequenceNumber != undefined) {\n this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n }\n if (this.timestamp != undefined) {\n this.timestampOffset = uint32Add(this.timestamp, -timestamp);\n }\n this.rtpCache = [];\n log(\"replaceRTP\", this.sequenceNumber, sequenceNumber, this.seqOffset);\n }\n\n async sendRtp(rtp: Buffer | RtpPacket) {\n if (this.dtlsTransport.state !== \"connected\" || !this.codec) {\n return;\n }\n\n rtp = Buffer.isBuffer(rtp) ? RtpPacket.deSerialize(rtp) : rtp;\n\n const header = rtp.header;\n header.ssrc = this.ssrc;\n header.payloadType = this.codec.payloadType;\n header.timestamp = uint32Add(header.timestamp, this.timestampOffset);\n header.sequenceNumber = uint16Add(header.sequenceNumber, this.seqOffset);\n this.timestamp = header.timestamp;\n this.sequenceNumber = header.sequenceNumber;\n\n header.extensions = this.headerExtensions\n .map((extension) => {\n const payload = (() => {\n switch (extension.uri) {\n case RTP_EXTENSION_URI.sdesMid:\n if (this.mid) {\n return Buffer.from(this.mid);\n }\n return;\n // todo : sender simulcast unsupported now\n case RTP_EXTENSION_URI.sdesRTPStreamID:\n if (this.rtpStreamId) {\n return Buffer.from(this.rtpStreamId);\n }\n return;\n // todo : sender simulcast unsupported now\n case RTP_EXTENSION_URI.repairedRtpStreamId:\n if (this.repairedRtpStreamId) {\n return Buffer.from(this.repairedRtpStreamId);\n }\n return;\n case RTP_EXTENSION_URI.transportWideCC:\n this.dtlsTransport.transportSequenceNumber = uint16Add(\n this.dtlsTransport.transportSequenceNumber,\n 1\n );\n return bufferWriter(\n [2],\n [this.dtlsTransport.transportSequenceNumber]\n );\n case RTP_EXTENSION_URI.absSendTime:\n const buf = Buffer.alloc(3);\n const time = (ntpTime() >> 14n) & 0x00ffffffn;\n buf.writeUIntBE(Number(time), 0, 3);\n return buf;\n }\n })();\n\n if (payload) return { id: extension.id, payload };\n })\n .filter((v) => v) as Extension[];\n\n this.ntpTimestamp = ntpTime();\n this.rtpTimestamp = rtp.header.timestamp;\n this.octetCount += rtp.payload.length;\n this.packetCount = uint32Add(this.packetCount, 1);\n\n rtp.header = header;\n\n this.rtpCache.push(rtp);\n this.rtpCache = this.rtpCache.slice(-RTP_HISTORY_SIZE);\n\n let rtpPayload = rtp.payload;\n\n if (this.redRedundantPayloadType) {\n this.redEncoder.push({\n block: rtpPayload,\n timestamp: header.timestamp,\n blockPT: this.redRedundantPayloadType,\n });\n const red = this.redEncoder.build();\n rtpPayload = red.serialize();\n }\n\n const size = await this.dtlsTransport.sendRtp(rtpPayload, header);\n\n this.runRtcp();\n const sentInfo: SentInfo = {\n wideSeq: this.dtlsTransport.transportSequenceNumber,\n size,\n sendingAtMs: milliTime(),\n sentAtMs: milliTime(),\n };\n this.senderBWE.rtpPacketSent(sentInfo);\n }\n\n handleRtcpPacket(rtcpPacket: RtcpPacket) {\n switch (rtcpPacket.type) {\n case RtcpSrPacket.type:\n case RtcpRrPacket.type:\n {\n const packet = rtcpPacket as RtcpSrPacket | RtcpRrPacket;\n packet.reports\n .filter((report) => report.ssrc === this.ssrc)\n .forEach((report) => {\n if (this.lsr === report.lsr && report.dlsr) {\n const rtt =\n Date.now() / 1000 - this.lsrTime - report.dlsr / 65536;\n if (this.rtt === undefined) {\n this.rtt = rtt;\n } else {\n this.rtt = RTT_ALPHA * this.rtt + (1 - RTT_ALPHA) * rtt;\n }\n }\n });\n }\n break;\n case RtcpTransportLayerFeedback.type:\n {\n const packet = rtcpPacket as RtcpTransportLayerFeedback;\n switch (packet.feedback.count) {\n case TransportWideCC.count:\n {\n const feedback = packet.feedback as TransportWideCC;\n this.senderBWE.receiveTWCC(feedback);\n }\n break;\n case GenericNack.count:\n {\n const feedback = packet.feedback as GenericNack;\n feedback.lost.forEach(async (seqNum) => {\n let packet = this.rtpCache.find(\n (rtp) => rtp.header.sequenceNumber === seqNum\n );\n if (packet) {\n if (this.rtxPayloadType != undefined) {\n packet = wrapRtx(\n packet,\n this.rtxPayloadType,\n this.rtxSequenceNumber,\n this.rtxSsrc\n );\n this.rtxSequenceNumber = uint16Add(\n this.rtxSequenceNumber,\n 1\n );\n }\n await this.dtlsTransport.sendRtp(\n packet.payload,\n packet.header\n );\n }\n });\n this.onGenericNack.execute(feedback);\n }\n break;\n }\n }\n break;\n case RtcpPayloadSpecificFeedback.type:\n {\n const packet = rtcpPacket as RtcpPayloadSpecificFeedback;\n switch (packet.feedback.count) {\n case ReceiverEstimatedMaxBitrate.count:\n {\n const feedback = packet.feedback as ReceiverEstimatedMaxBitrate;\n this.receiverEstimatedMaxBitrate = feedback.bitrate;\n }\n break;\n case PictureLossIndication.count:\n {\n this.onPictureLossIndication.execute();\n }\n break;\n }\n }\n break;\n }\n this.onRtcp.execute(rtcpPacket);\n }\n}\n\nexport function wrapRtx(\n packet: RtpPacket,\n payloadType: number,\n sequenceNumber: number,\n ssrc: number\n) {\n const rtx = new RtpPacket(\n new RtpHeader({\n payloadType,\n marker: packet.header.marker,\n sequenceNumber,\n timestamp: packet.header.timestamp,\n ssrc,\n csrc: packet.header.csrc,\n extensions: packet.header.extensions,\n }),\n Buffer.concat([\n Buffer.from(jspack.Pack(\"!H\", [packet.header.sequenceNumber])),\n packet.payload,\n ])\n );\n return rtx;\n}\n"]}
|
|
1
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+
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{ randomBytes } from \"crypto\";\nimport debug from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport * as uuid from \"uuid\";\n\nimport {\n bufferWriter,\n random16,\n uint16Add,\n uint32Add,\n} from \"../../../common/src\";\nimport {\n Extension,\n GenericNack,\n PictureLossIndication,\n ReceiverEstimatedMaxBitrate,\n RedEncoder,\n RtcpPacket,\n RtcpPayloadSpecificFeedback,\n RtcpRrPacket,\n RtcpSenderInfo,\n RtcpSourceDescriptionPacket,\n RtcpSrPacket,\n RtcpTransportLayerFeedback,\n RtpHeader,\n RtpPacket,\n SourceDescriptionChunk,\n SourceDescriptionItem,\n TransportWideCC,\n} from \"../../../rtp/src\";\nimport { codecParametersFromString } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, milliTime, ntpTime } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport {\n RTCRtpCodecParameters,\n RTCRtpHeaderExtensionParameters,\n RTCRtpSendParameters,\n} from \"./parameters\";\nimport { SenderBandwidthEstimator, SentInfo } from \"./sender/senderBWE\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpSender.ts\");\n\nconst RTP_HISTORY_SIZE = 128;\nconst RTT_ALPHA = 0.85;\n\nexport class RTCRtpSender {\n readonly type = \"sender\";\n readonly kind =\n typeof this.trackOrKind === \"string\"\n ? this.trackOrKind\n : this.trackOrKind.kind;\n readonly ssrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n readonly rtxSsrc = jspack.Unpack(\"!L\", randomBytes(4))[0];\n streamId = uuid.v4();\n readonly trackId = uuid.v4();\n readonly onReady = new Event();\n readonly onRtcp = new Event<[RtcpPacket]>();\n readonly onPictureLossIndication = new Event<[]>();\n readonly onGenericNack = new Event<[GenericNack]>();\n readonly senderBWE = new SenderBandwidthEstimator();\n\n private cname?: string;\n private mid?: string;\n private rtpStreamId?: string;\n private repairedRtpStreamId?: string;\n private rtxPayloadType?: number;\n private rtxSequenceNumber = random16();\n redRedundantPayloadType?: number;\n private _redDistance = 2;\n redEncoder = new RedEncoder(this._redDistance);\n private headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n private disposeTrack?: () => void;\n\n // # stats\n private lsr?: number;\n private lsrTime: number = Date.now() / 1000;\n private ntpTimestamp = 0n;\n private rtpTimestamp = 0;\n private octetCount = 0;\n private packetCount = 0;\n private rtt?: number;\n receiverEstimatedMaxBitrate: bigint = 0n;\n\n // rtp\n private sequenceNumber?: number;\n private timestamp?: number;\n private timestampOffset = 0;\n private seqOffset = 0;\n private rtpCache: RtpPacket[] = [];\n codec?: RTCRtpCodecParameters;\n public dtlsTransport!: RTCDtlsTransport;\n private dtlsDisposer: (() => void)[] = [];\n\n track?: MediaStreamTrack;\n stopped = false;\n rtcpRunning = false;\n private rtcpCancel = new AbortController();\n\n constructor(public trackOrKind: Kind | MediaStreamTrack) {\n if (trackOrKind instanceof MediaStreamTrack) {\n if (trackOrKind.streamId) {\n this.streamId = trackOrKind.streamId;\n }\n this.registerTrack(trackOrKind);\n }\n }\n\n setDtlsTransport(dtlsTransport: RTCDtlsTransport) {\n if (this.dtlsTransport) {\n this.dtlsDisposer.forEach((dispose) => dispose());\n }\n\n this.dtlsTransport = dtlsTransport;\n this.dtlsDisposer = [\n this.dtlsTransport.onStateChange.subscribe((state) => {\n if (state === \"connected\") {\n this.onReady.execute();\n }\n }).unSubscribe,\n ];\n }\n\n get redDistance() {\n return this._redDistance;\n }\n set redDistance(n: number) {\n this._redDistance = n;\n this.redEncoder.distance = n;\n }\n\n prepareSend(params: RTCRtpSendParameters) {\n this.cname = params.rtcp?.cname;\n this.mid = params.muxId;\n this.headerExtensions = params.headerExtensions;\n this.rtpStreamId = params.rtpStreamId;\n this.repairedRtpStreamId = params.repairedRtpStreamId;\n\n this.codec = params.codecs[0];\n if (this.track) {\n this.track.codec = this.codec;\n }\n\n params.codecs.forEach((codec) => {\n const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n if (\n codec.name.toLowerCase() === \"rtx\" &&\n codecParams[\"apt\"] === this.codec?.payloadType\n ) {\n this.rtxPayloadType = codec.payloadType;\n }\n if (codec.name.toLowerCase() === \"red\") {\n this.redRedundantPayloadType = Number(\n (codec.parameters ?? \"\").split(\"/\")[0]\n );\n }\n });\n }\n\n registerTrack(track: MediaStreamTrack) {\n if (track.stopped) throw new Error(\"track is ended\");\n\n if (this.disposeTrack) {\n this.disposeTrack();\n }\n\n track.id = this.trackId;\n\n const { unSubscribe } = track.onReceiveRtp.subscribe(async (rtp) => {\n await this.sendRtp(rtp);\n });\n this.track = track;\n this.disposeTrack = unSubscribe;\n\n if (this.codec) {\n track.codec = this.codec;\n }\n\n track.onSourceChanged.subscribe((header) => {\n this.replaceRTP(header);\n });\n }\n\n async replaceTrack(track: MediaStreamTrack | null) {\n if (track === null) {\n // todo impl\n return;\n }\n\n if (track.stopped) throw new Error(\"track is ended\");\n\n if (this.sequenceNumber != undefined) {\n const header =\n track.header || (await track.onReceiveRtp.asPromise())[0].header;\n\n this.replaceRTP(header);\n }\n\n this.registerTrack(track);\n log(\"replaceTrack\", \"ssrc\", track.ssrc, \"rid\", track.rid);\n }\n\n stop() {\n this.stopped = true;\n this.rtcpRunning = false;\n this.rtcpCancel.abort();\n\n this.track = undefined;\n }\n\n async runRtcp() {\n if (this.rtcpRunning || this.stopped) return;\n this.rtcpRunning = true;\n\n try {\n while (this.rtcpRunning) {\n await setTimeout(500 + Math.random() * 1000, undefined, {\n signal: this.rtcpCancel.signal,\n });\n\n const packets: RtcpPacket[] = [\n new RtcpSrPacket({\n ssrc: this.ssrc,\n senderInfo: new RtcpSenderInfo({\n ntpTimestamp: this.ntpTimestamp,\n rtpTimestamp: this.rtpTimestamp,\n packetCount: this.packetCount,\n octetCount: this.octetCount,\n }),\n }),\n ];\n this.lsr = compactNtp(this.ntpTimestamp);\n this.lsrTime = Date.now() / 1000;\n\n if (this.cname) {\n packets.push(\n new RtcpSourceDescriptionPacket({\n chunks: [\n new SourceDescriptionChunk({\n source: this.ssrc,\n items: [\n new SourceDescriptionItem({ type: 1, text: this.cname }),\n ],\n }),\n ],\n })\n );\n }\n\n try {\n await this.dtlsTransport.sendRtcp(packets);\n } catch (error) {\n log(\"sendRtcp failed\", error);\n await setTimeout(500 + Math.random() * 1000);\n }\n }\n } catch (error) {}\n }\n\n replaceRTP({\n sequenceNumber,\n timestamp,\n }: Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">) {\n if (this.sequenceNumber != undefined) {\n this.seqOffset = uint16Add(this.sequenceNumber, -sequenceNumber);\n }\n if (this.timestamp != undefined) {\n this.timestampOffset = uint32Add(this.timestamp, -timestamp);\n }\n this.rtpCache = [];\n log(\"replaceRTP\", this.sequenceNumber, sequenceNumber, this.seqOffset);\n }\n\n async sendRtp(rtp: Buffer | RtpPacket) {\n if (this.dtlsTransport.state !== \"connected\" || !this.codec) {\n return;\n }\n\n rtp = Buffer.isBuffer(rtp) ? RtpPacket.deSerialize(rtp) : rtp;\n\n const header = rtp.header;\n header.ssrc = this.ssrc;\n header.payloadType = this.codec.payloadType;\n header.timestamp = uint32Add(header.timestamp, this.timestampOffset);\n header.sequenceNumber = uint16Add(header.sequenceNumber, this.seqOffset);\n this.timestamp = header.timestamp;\n this.sequenceNumber = header.sequenceNumber;\n\n header.extensions = this.headerExtensions\n .map((extension) => {\n const payload = (() => {\n switch (extension.uri) {\n case RTP_EXTENSION_URI.sdesMid:\n if (this.mid) {\n return Buffer.from(this.mid);\n }\n return;\n // todo : sender simulcast unsupported now\n case RTP_EXTENSION_URI.sdesRTPStreamID:\n if (this.rtpStreamId) {\n return Buffer.from(this.rtpStreamId);\n }\n return;\n // todo : sender simulcast unsupported now\n case RTP_EXTENSION_URI.repairedRtpStreamId:\n if (this.repairedRtpStreamId) {\n return Buffer.from(this.repairedRtpStreamId);\n }\n return;\n case RTP_EXTENSION_URI.transportWideCC:\n this.dtlsTransport.transportSequenceNumber = uint16Add(\n this.dtlsTransport.transportSequenceNumber,\n 1\n );\n return bufferWriter(\n [2],\n [this.dtlsTransport.transportSequenceNumber]\n );\n case RTP_EXTENSION_URI.absSendTime:\n const buf = Buffer.alloc(3);\n const time = (ntpTime() >> 14n) & 0x00ffffffn;\n buf.writeUIntBE(Number(time), 0, 3);\n return buf;\n }\n })();\n\n if (payload) return { id: extension.id, payload };\n })\n .filter((v) => v) as Extension[];\n\n this.ntpTimestamp = ntpTime();\n this.rtpTimestamp = rtp.header.timestamp;\n this.octetCount += rtp.payload.length;\n this.packetCount = uint32Add(this.packetCount, 1);\n\n rtp.header = header;\n\n this.rtpCache.push(rtp);\n this.rtpCache = this.rtpCache.slice(-RTP_HISTORY_SIZE);\n\n let rtpPayload = rtp.payload;\n\n if (this.redRedundantPayloadType) {\n this.redEncoder.push({\n block: rtpPayload,\n timestamp: header.timestamp,\n blockPT: this.redRedundantPayloadType,\n });\n const red = this.redEncoder.build();\n rtpPayload = red.serialize();\n }\n\n const size = await this.dtlsTransport.sendRtp(rtpPayload, header);\n\n this.runRtcp();\n const sentInfo: SentInfo = {\n wideSeq: this.dtlsTransport.transportSequenceNumber,\n size,\n sendingAtMs: milliTime(),\n sentAtMs: milliTime(),\n };\n this.senderBWE.rtpPacketSent(sentInfo);\n }\n\n handleRtcpPacket(rtcpPacket: RtcpPacket) {\n switch (rtcpPacket.type) {\n case RtcpSrPacket.type:\n case RtcpRrPacket.type:\n {\n const packet = rtcpPacket as RtcpSrPacket | RtcpRrPacket;\n packet.reports\n .filter((report) => report.ssrc === this.ssrc)\n .forEach((report) => {\n if (this.lsr === report.lsr && report.dlsr) {\n const rtt =\n Date.now() / 1000 - this.lsrTime - report.dlsr / 65536;\n if (this.rtt === undefined) {\n this.rtt = rtt;\n } else {\n this.rtt = RTT_ALPHA * this.rtt + (1 - RTT_ALPHA) * rtt;\n }\n }\n });\n }\n break;\n case RtcpTransportLayerFeedback.type:\n {\n const packet = rtcpPacket as RtcpTransportLayerFeedback;\n switch (packet.feedback.count) {\n case TransportWideCC.count:\n {\n const feedback = packet.feedback as TransportWideCC;\n this.senderBWE.receiveTWCC(feedback);\n }\n break;\n case GenericNack.count:\n {\n const feedback = packet.feedback as GenericNack;\n feedback.lost.forEach(async (seqNum) => {\n let packet = this.rtpCache.find(\n (rtp) => rtp.header.sequenceNumber === seqNum\n );\n if (packet) {\n if (this.rtxPayloadType != undefined) {\n packet = wrapRtx(\n packet,\n this.rtxPayloadType,\n this.rtxSequenceNumber,\n this.rtxSsrc\n );\n this.rtxSequenceNumber = uint16Add(\n this.rtxSequenceNumber,\n 1\n );\n }\n await this.dtlsTransport.sendRtp(\n packet.payload,\n packet.header\n );\n }\n });\n this.onGenericNack.execute(feedback);\n }\n break;\n }\n }\n break;\n case RtcpPayloadSpecificFeedback.type:\n {\n const packet = rtcpPacket as RtcpPayloadSpecificFeedback;\n switch (packet.feedback.count) {\n case ReceiverEstimatedMaxBitrate.count:\n {\n const feedback = packet.feedback as ReceiverEstimatedMaxBitrate;\n this.receiverEstimatedMaxBitrate = feedback.bitrate;\n }\n break;\n case PictureLossIndication.count:\n {\n this.onPictureLossIndication.execute();\n }\n break;\n }\n }\n break;\n }\n this.onRtcp.execute(rtcpPacket);\n }\n}\n\nexport function wrapRtx(\n packet: RtpPacket,\n payloadType: number,\n sequenceNumber: number,\n ssrc: number\n) {\n const rtx = new RtpPacket(\n new RtpHeader({\n payloadType,\n marker: packet.header.marker,\n sequenceNumber,\n timestamp: packet.header.timestamp,\n ssrc,\n csrc: packet.header.csrc,\n extensions: packet.header.extensions,\n }),\n Buffer.concat([\n Buffer.from(jspack.Pack(\"!H\", [packet.header.sequenceNumber])),\n packet.payload,\n ])\n );\n return rtx;\n}\n"]}
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@@ -1,5 +1,5 @@
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1
1
|
import Event from "rx.mini";
|
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2
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-
import { RTCDtlsTransport } from "
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2
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+
import { RTCDtlsTransport } from "..";
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3
3
|
import { Kind } from "../types/domain";
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4
4
|
import { RTCRtpCodecParameters, RTCRtpHeaderExtensionParameters } from "./parameters";
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5
5
|
import { RTCRtpReceiver } from "./rtpReceiver";
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@@ -7,11 +7,11 @@ import { RTCRtpSender } from "./rtpSender";
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7
7
|
import { MediaStreamTrack } from "./track";
|
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8
8
|
export declare class RTCRtpTransceiver {
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9
9
|
readonly kind: Kind;
|
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10
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-
|
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11
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-
|
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10
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+
dtlsTransport: RTCDtlsTransport;
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11
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+
receiver: RTCRtpReceiver;
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12
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+
sender: RTCRtpSender;
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12
13
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/**RFC 8829 4.2.4. direction the transceiver was initialized with */
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13
14
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direction: Direction;
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14
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-
dtlsTransport: RTCDtlsTransport;
|
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15
15
|
readonly uuid: string;
|
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16
16
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readonly onTrack: Event<[MediaStreamTrack]>;
|
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17
17
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mid?: string;
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@@ -27,9 +27,10 @@ export declare class RTCRtpTransceiver {
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27
27
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options: Partial<TransceiverOptions>;
|
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28
28
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stopping: boolean;
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29
29
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stopped: boolean;
|
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30
|
-
constructor(kind: Kind, receiver: RTCRtpReceiver, sender: RTCRtpSender,
|
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30
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+
constructor(kind: Kind, dtlsTransport: RTCDtlsTransport, receiver: RTCRtpReceiver, sender: RTCRtpSender,
|
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31
31
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/**RFC 8829 4.2.4. direction the transceiver was initialized with */
|
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32
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-
direction: Direction
|
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32
|
+
direction: Direction);
|
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33
|
+
setDtlsTransport(dtls: RTCDtlsTransport): void;
|
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33
34
|
get msid(): string;
|
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34
35
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addTrack(track: MediaStreamTrack): void;
|
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35
36
|
stop(): void;
|
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@@ -27,14 +27,14 @@ const rx_mini_1 = __importDefault(require("rx.mini"));
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27
27
|
const uuid = __importStar(require("uuid"));
|
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28
28
|
const const_1 = require("../const");
|
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29
29
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class RTCRtpTransceiver {
|
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30
|
-
constructor(kind, receiver, sender,
|
|
30
|
+
constructor(kind, dtlsTransport, receiver, sender,
|
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31
31
|
/**RFC 8829 4.2.4. direction the transceiver was initialized with */
|
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32
|
-
direction
|
|
32
|
+
direction) {
|
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33
33
|
this.kind = kind;
|
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34
|
+
this.dtlsTransport = dtlsTransport;
|
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34
35
|
this.receiver = receiver;
|
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35
36
|
this.sender = sender;
|
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36
37
|
this.direction = direction;
|
|
37
|
-
this.dtlsTransport = dtlsTransport;
|
|
38
38
|
this.uuid = uuid.v4();
|
|
39
39
|
this.onTrack = new rx_mini_1.default();
|
|
40
40
|
this.usedForSender = false;
|
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@@ -43,6 +43,7 @@ class RTCRtpTransceiver {
|
|
|
43
43
|
this.options = {};
|
|
44
44
|
this.stopping = false;
|
|
45
45
|
this.stopped = false;
|
|
46
|
+
this.setDtlsTransport(dtlsTransport);
|
|
46
47
|
}
|
|
47
48
|
set currentDirection(direction) {
|
|
48
49
|
this._currentDirection = direction;
|
|
@@ -54,6 +55,10 @@ class RTCRtpTransceiver {
|
|
|
54
55
|
get currentDirection() {
|
|
55
56
|
return this._currentDirection;
|
|
56
57
|
}
|
|
58
|
+
setDtlsTransport(dtls) {
|
|
59
|
+
this.receiver.setDtlsTransport(dtls);
|
|
60
|
+
this.sender.setDtlsTransport(dtls);
|
|
61
|
+
}
|
|
57
62
|
get msid() {
|
|
58
63
|
return `${this.sender.streamId} ${this.sender.trackId}`;
|
|
59
64
|
}
|
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@@ -1 +1 @@
|
|
|
1
|
-
{"version":3,"file":"rtpTransceiver.js","sourceRoot":"","sources":["../../../../src/media/rtpTransceiver.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;;;;;;;;;;AAAA,sDAA4B;AAC5B,2CAA6B;
|
|
1
|
+
{"version":3,"file":"rtpTransceiver.js","sourceRoot":"","sources":["../../../../src/media/rtpTransceiver.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;;;;;;;;;;AAAA,sDAA4B;AAC5B,2CAA6B;AAG7B,oCAA4C;AAU5C,MAAa,iBAAiB;IAyB5B,YACkB,IAAU,EACnB,aAA+B,EAC/B,QAAwB,EACxB,MAAoB;IAC3B,qEAAqE;IAC9D,SAAoB;QALX,SAAI,GAAJ,IAAI,CAAM;QACnB,kBAAa,GAAb,aAAa,CAAkB;QAC/B,aAAQ,GAAR,QAAQ,CAAgB;QACxB,WAAM,GAAN,MAAM,CAAc;QAEpB,cAAS,GAAT,SAAS,CAAW;QA9BpB,SAAI,GAAG,IAAI,CAAC,EAAE,EAAE,CAAC;QACjB,YAAO,GAAG,IAAI,iBAAK,EAAsB,CAAC;QAGnD,kBAAa,GAAG,KAAK,CAAC;QActB,WAAM,GAA4B,EAAE,CAAC;QACrC,qBAAgB,GAAsC,EAAE,CAAC;QACzD,YAAO,GAAgC,EAAE,CAAC;QAC1C,aAAQ,GAAG,KAAK,CAAC;QACjB,YAAO,GAAG,KAAK,CAAC;QAUd,IAAI,CAAC,gBAAgB,CAAC,aAAa,CAAC,CAAC;IACvC,CAAC;IA3BD,IAAI,gBAAgB,CAAC,SAA4C;QAC/D,IAAI,CAAC,iBAAiB,GAAG,SAAS,CAAC;QACnC,IAAI,wBAAgB,CAAC,QAAQ,CAAC,IAAI,CAAC,iBAAiB,IAAI,EAAE,CAAC,EAAE;YAC3D,IAAI,CAAC,aAAa,GAAG,IAAI,CAAC;SAC3B;IACH,CAAC;IACD,+CAA+C;IAC/C,IAAI,gBAAgB;QAClB,OAAO,IAAI,CAAC,iBAAiB,CAAC;IAChC,CAAC;IAoBD,gBAAgB,CAAC,IAAsB;QACrC,IAAI,CAAC,QAAQ,CAAC,gBAAgB,CAAC,IAAI,CAAC,CAAC;QACrC,IAAI,CAAC,MAAM,CAAC,gBAAgB,CAAC,IAAI,CAAC,CAAC;IACrC,CAAC;IAED,IAAI,IAAI;QACN,OAAO,GAAG,IAAI,CAAC,MAAM,CAAC,QAAQ,IAAI,IAAI,CAAC,MAAM,CAAC,OAAO,EAAE,CAAC;IAC1D,CAAC;IAED,QAAQ,CAAC,KAAuB;QAC9B,MAAM,GAAG,GAAG,IAAI,CAAC,QAAQ,CAAC,QAAQ,CAAC,KAAK,CAAC,CAAC;QAC1C,IAAI,GAAG;YAAE,IAAI,CAAC,OAAO,CAAC,OAAO,CAAC,KAAK,CAAC,CAAC;IACvC,CAAC;IAED,YAAY;IACZ,0CAA0C;IAC1C,IAAI;QACF,IAAI,IAAI,CAAC,QAAQ;YAAE,OAAO;QAE1B,oDAAoD;QAEpD,IAAI,CAAC,QAAQ,GAAG,IAAI,CAAC;IACvB,CAAC;IAED,cAAc,CAAC,QAAgB;QAC7B,OAAO,IAAI,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC,KAAK,EAAE,EAAE,CAChC,KAAK,CAAC,QAAQ,CAAC,WAAW,EAAE,CAAC,QAAQ,CAAC,QAAQ,CAAC,WAAW,EAAE,CAAC,CAC9D,EAAE,WAAW,CAAC;IACjB,CAAC;CACF;AAjED,8CAiEC;AAEY,QAAA,UAAU,GAAG;IACxB,UAAU;IACV,UAAU;IACV,UAAU;IACV,UAAU;CACF,CAAC","sourcesContent":["import Event from \"rx.mini\";\nimport * as uuid from \"uuid\";\n\nimport { RTCDtlsTransport } from \"..\";\nimport { SenderDirections } from \"../const\";\nimport { Kind } from \"../types/domain\";\nimport {\n RTCRtpCodecParameters,\n RTCRtpHeaderExtensionParameters,\n} from \"./parameters\";\nimport { RTCRtpReceiver } from \"./rtpReceiver\";\nimport { RTCRtpSender } from \"./rtpSender\";\nimport { MediaStreamTrack } from \"./track\";\n\nexport class RTCRtpTransceiver {\n readonly uuid = uuid.v4();\n readonly onTrack = new Event<[MediaStreamTrack]>();\n mid?: string;\n mLineIndex?: number;\n usedForSender = false;\n private _currentDirection?: Direction | \"stopped\";\n set currentDirection(direction: Direction | \"stopped\" | undefined) {\n this._currentDirection = direction;\n if (SenderDirections.includes(this._currentDirection || \"\")) {\n this.usedForSender = true;\n }\n }\n /**RFC 8829 4.2.5. last negotiated direction */\n get currentDirection(): Direction | \"stopped\" | undefined {\n return this._currentDirection;\n }\n\n offerDirection!: Direction;\n codecs: RTCRtpCodecParameters[] = [];\n headerExtensions: RTCRtpHeaderExtensionParameters[] = [];\n options: Partial<TransceiverOptions> = {};\n stopping = false;\n stopped = false;\n\n constructor(\n public readonly kind: Kind,\n public dtlsTransport: RTCDtlsTransport,\n public receiver: RTCRtpReceiver,\n public sender: RTCRtpSender,\n /**RFC 8829 4.2.4. direction the transceiver was initialized with */\n public direction: Direction\n ) {\n this.setDtlsTransport(dtlsTransport);\n }\n\n setDtlsTransport(dtls: RTCDtlsTransport) {\n this.receiver.setDtlsTransport(dtls);\n this.sender.setDtlsTransport(dtls);\n }\n\n get msid() {\n return `${this.sender.streamId} ${this.sender.trackId}`;\n }\n\n addTrack(track: MediaStreamTrack) {\n const res = this.receiver.addTrack(track);\n if (res) this.onTrack.execute(track);\n }\n\n // todo impl\n // https://www.w3.org/TR/webrtc/#methods-8\n stop() {\n if (this.stopping) return;\n\n // todo Stop sending and receiving with transceiver.\n\n this.stopping = true;\n }\n\n getPayloadType(mimeType: string) {\n return this.codecs.find((codec) =>\n codec.mimeType.toLowerCase().includes(mimeType.toLowerCase())\n )?.payloadType;\n }\n}\n\nexport const Directions = [\n \"inactive\",\n \"sendonly\",\n \"recvonly\",\n \"sendrecv\",\n] as const;\n\nexport type Direction = typeof Directions[number];\n\ntype SimulcastDirection = \"send\" | \"recv\";\n\nexport interface TransceiverOptions {\n direction: Direction;\n simulcast: { direction: SimulcastDirection; rid: string }[];\n}\n"]}
|
|
@@ -25,7 +25,7 @@ const __1 = require("../../..");
|
|
|
25
25
|
const _1 = require(".");
|
|
26
26
|
class WebmFactory extends _1.MediaWriter {
|
|
27
27
|
start(tracks) {
|
|
28
|
-
this.webm = new __1.WebmOutput(fs,
|
|
28
|
+
this.webm = new __1.WebmOutput(fs, this.path, tracks.map((track, i) => {
|
|
29
29
|
const trackNumber = i + 1;
|
|
30
30
|
const payloadType = track.codec.payloadType;
|
|
31
31
|
if (track.kind === "video") {
|
|
@@ -1 +1 @@
|
|
|
1
|
-
{"version":3,"file":"webm.js","sourceRoot":"","sources":["../../../../../../src/nonstandard/recorder/writer/webm.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;;;;;;;AAAA,gDAAkC;AAGlC,gCAKkB;AAClB,wBAAgC;AAEhC,MAAa,WAAY,SAAQ,cAAW;IAG1C,KAAK,CAAC,MAA0B;QAC9B,IAAI,CAAC,IAAI,GAAG,IAAI,cAAU,CACxB,EAAE,EACF,
|
|
1
|
+
{"version":3,"file":"webm.js","sourceRoot":"","sources":["../../../../../../src/nonstandard/recorder/writer/webm.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;;;;;;;AAAA,gDAAkC;AAGlC,gCAKkB;AAClB,wBAAgC;AAEhC,MAAa,WAAY,SAAQ,cAAW;IAG1C,KAAK,CAAC,MAA0B;QAC9B,IAAI,CAAC,IAAI,GAAG,IAAI,cAAU,CACxB,EAAE,EACF,IAAI,CAAC,IAAI,EACT,MAAM,CAAC,GAAG,CAAC,CAAC,KAAK,EAAE,CAAC,EAAE,EAAE;YACtB,MAAM,WAAW,GAAG,CAAC,GAAG,CAAC,CAAC;YAC1B,MAAM,WAAW,GAAG,KAAK,CAAC,KAAM,CAAC,WAAW,CAAC;YAE7C,IAAI,KAAK,CAAC,IAAI,KAAK,OAAO,EAAE;gBAC1B,MAAM,KAAK,GAAG,CAAC,GAAmB,EAAE;oBAClC,QAAQ,KAAK,CAAC,KAAK,EAAE,IAAI,CAAC,WAAW,EAAyB,EAAE;wBAC9D,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,KAAK;4BACR,OAAO,KAAK,CAAC;wBACf,KAAK,MAAM;4BACT,OAAO,eAAe,CAAC;wBACzB,KAAK,MAAM;4BACT,OAAO,KAAK,CAAC;wBACf;4BACE,MAAM,IAAI,KAAK,EAAE,CAAC;qBACrB;gBACH,CAAC,CAAC,EAAE,CAAC;gBACL,OAAO;oBACL,IAAI,EAAE,OAAO;oBACb,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK;oBACL,KAAK,EAAE,IAAI,CAAC,OAAO,CAAC,KAAK;oBACzB,MAAM,EAAE,IAAI,CAAC,OAAO,CAAC,MAAM;iBAC5B,CAAC;aACH;iBAAM;gBACL,OAAO;oBACL,IAAI,EAAE,OAAO;oBACb,SAAS,EAAE,KAAK;oBAChB,WAAW;oBACX,WAAW;oBACX,KAAK,EAAE,MAAM;iBACd,CAAC;aACH;QACH,CAAC,CAAC,CACH,CAAC;QAEF,MAAM,CAAC,OAAO,CAAC,CAAC,KAAK,EAAE,EAAE;YACvB,MAAM,aAAa,GACjB,KAAK,CAAC,IAAI,KAAK,OAAO;gBACpB,CAAC,CAAC,IAAI,iBAAa,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,MAAM,CAAC,CAAC,IAAI,CAAC,IAAI,CAAC,IAAK,CAAC;gBACvD,CAAC,CAAC,IAAI,iBAAa,CAAC,GAAG,EAAE,CAAC,IAAI,CAAC,CAAC,IAAI,CAAC,IAAI,CAAC,IAAK,CAAC,CAAC;YACrD,IAAI,gBAAY,CAAC;gBACf,SAAS,EAAE,KAAK,CAAC,YAAY;gBAC7B,UAAU,EAAE,KAAK,CAAC,aAAa;aAChC,CAAC,CAAC,IAAI,CAAC,aAAa,CAAC,CAAC;QACzB,CAAC,CAAC,CAAC;IACL,CAAC;IAED,KAAK,CAAC,IAAI;QACR,MAAM,IAAI,CAAC,IAAK,CAAC,IAAI,EAAE,CAAC;IAC1B,CAAC;CACF;AA9DD,kCA8DC;AAED,MAAM,oBAAoB,GAAG,CAAC,MAAM,EAAE,KAAK,EAAE,KAAK,EAAE,MAAM,CAAU,CAAC","sourcesContent":["import * as fs from \"fs/promises\";\n\nimport { SupportedCodec } from \"../../../../../rtp/src/container/webm\";\nimport {\n JitterBuffer,\n MediaStreamTrack,\n SampleBuilder,\n WebmOutput,\n} from \"../../..\";\nimport { MediaWriter } from \".\";\n\nexport class WebmFactory extends MediaWriter {\n webm?: WebmOutput;\n\n start(tracks: MediaStreamTrack[]) {\n this.webm = new WebmOutput(\n fs,\n this.path,\n tracks.map((track, i) => {\n const trackNumber = i + 1;\n const payloadType = track.codec!.payloadType;\n\n if (track.kind === \"video\") {\n const codec = ((): SupportedCodec => {\n switch (track.codec?.name.toLowerCase() as SupportedVideoCodec) {\n case \"vp8\":\n return \"VP8\";\n case \"vp9\":\n return \"VP9\";\n case \"h264\":\n return \"MPEG4/ISO/AVC\";\n case \"av1x\":\n return \"AV1\";\n default:\n throw new Error();\n }\n })();\n return {\n kind: \"video\",\n clockRate: 90000,\n payloadType,\n trackNumber,\n codec,\n width: this.options.width,\n height: this.options.height,\n };\n } else {\n return {\n kind: \"audio\",\n clockRate: 48000,\n payloadType,\n trackNumber,\n codec: \"OPUS\",\n };\n }\n })\n );\n\n tracks.forEach((track) => {\n const sampleBuilder =\n track.kind === \"video\"\n ? new SampleBuilder((h) => !!h.marker).pipe(this.webm!)\n : new SampleBuilder(() => true).pipe(this.webm!);\n new JitterBuffer({\n rtpStream: track.onReceiveRtp,\n rtcpStream: track.onReceiveRtcp,\n }).pipe(sampleBuilder);\n });\n }\n\n async stop() {\n await this.webm!.stop();\n }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = typeof supportedVideoCodecs[number];\n"]}
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@@ -15,8 +15,6 @@ import { ConnectionState, Kind, RTCSignalingState } from "./types/domain";
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15
15
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import { Callback, CallbackWithValue } from "./types/util";
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16
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export declare class RTCPeerConnection extends EventTarget {
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readonly cname: string;
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18
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-
iceTransport: RTCIceTransport;
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19
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-
dtlsTransport: RTCDtlsTransport;
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sctpTransport?: RTCSctpTransport;
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masterTransportEstablished: boolean;
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configuration: Required<PeerConfig>;
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@@ -32,11 +30,6 @@ export declare class RTCPeerConnection extends EventTarget {
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readonly connectionStateChange: Event<["disconnected" | "closed" | "new" | "connected" | "connecting" | "failed"]>;
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readonly onDataChannel: Event<[RTCDataChannel]>;
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readonly onRemoteTransceiverAdded: Event<[RTCRtpTransceiver]>;
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-
/**
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* should use onRemoteTransceiverAdded
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-
* @deprecated
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-
*/
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-
readonly onTransceiver: Event<[RTCRtpTransceiver]>;
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readonly onTransceiverAdded: Event<[RTCRtpTransceiver]>;
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readonly onIceCandidate: Event<[RTCIceCandidate]>;
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readonly onNegotiationneeded: Event<[]>;
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@@ -56,7 +49,9 @@ export declare class RTCPeerConnection extends EventTarget {
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private pendingRemoteDescription?;
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private isClosed;
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private shouldNegotiationneeded;
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-
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+
get dtlsTransports(): RTCDtlsTransport[];
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+
get iceTransports(): RTCIceTransport[];
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+
constructor({ codecs, headerExtensions, iceServers, iceTransportPolicy, icePortRange, dtls, bundlePolicy, }?: Partial<PeerConfig>);
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get localDescription(): import("./sdp").RTCSessionDescription | undefined;
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get remoteDescription(): import("./sdp").RTCSessionDescription | undefined;
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private get _localDescription();
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@@ -83,12 +78,14 @@ export declare class RTCPeerConnection extends EventTarget {
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private setLocal;
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addIceCandidate(candidateMessage: RTCIceCandidate): Promise<void>;
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private connect;
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-
private
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-
private
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+
private getLocalRtpParams;
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+
private getRemoteRtpParams;
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setRemoteDescription(sessionDescription: {
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type: "offer" | "answer";
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sdp: string;
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}): Promise<void>;
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+
private setRemoteRTP;
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+
private setRemoteSCTP;
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private validateDescription;
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private fireOnTrack;
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addTransceiver(trackOrKind: Kind | MediaStreamTrack, options?: Partial<TransceiverOptions>): RTCRtpTransceiver;
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@@ -98,6 +95,7 @@ export declare class RTCPeerConnection extends EventTarget {
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addTrack(track: MediaStreamTrack,
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/**todo impl */
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ms?: MediaStream): RTCRtpSender;
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+
private ensureCerts;
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createAnswer(): Promise<import("./sdp").RTCSessionDescription>;
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close(): Promise<void>;
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private assertNotClosed;
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@@ -111,6 +109,7 @@ export declare function createMediaDescriptionForTransceiver(transceiver: RTCRtp
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export declare function createMediaDescriptionForSctp(sctp: RTCSctpTransport, mid: string): MediaDescription;
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export declare function addTransportDescription(media: MediaDescription, dtlsTransport: RTCDtlsTransport): void;
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export declare function allocateMid(mids: Set<string>): string;
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+
export declare type BundlePolicy = "max-compat" | "max-bundle";
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export interface PeerConfig {
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codecs: Partial<{
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audio: RTCRtpCodecParameters[];
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@@ -127,6 +126,7 @@ export interface PeerConfig {
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dtls: Partial<{
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keys: DtlsKeys;
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}>;
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+
bundlePolicy: BundlePolicy;
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}
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export declare const findCodecByMimeType: (codecs: RTCRtpCodecParameters[], target: RTCRtpCodecParameters) => RTCRtpCodecParameters | undefined;
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export declare type RTCIceServer = {
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