@lakutata/webrtc 0.0.0-stage → 3.1.1

package/dist/cjs/lib/peer/media/rtpSender.js

@@ -0,0 +1,826 @@
+"use strict";
+/**
+   [10 Nov 1995 11:33:25.125 UTC]       [10 Nov 1995 11:33:36.5 UTC]
+   n                 SR(n)              A=b710:8000 (46864.500 s)
+   ---------------------------------------------------------------->
+                      v                 ^
+   ntp_sec =0xb44db705 v               ^ dlsr=0x0005:4000 (    5.250s)
+   ntp_frac=0x20000000  v             ^  lsr =0xb705:2000 (46853.125s)
+     (3024992005.125 s)  v           ^
+   r                      v         ^ RR(n)
+   ---------------------------------------------------------------->
+                          |<-DLSR->|
+                           (5.250 s)
+        
+   A     0xb710:8000 (46864.500 s)
+   DLSR -0x0005:4000 (    5.250 s)
+   LSR  -0xb705:2000 (46853.125 s)
+   -------------------------------
+   delay 0x0006:2000 (    6.125 s)
+        
+Figure 2: Example for round-trip time computation
+ */
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.RTCRtpSender = void 0;
+const node_crypto_1 = require("node:crypto");
+const promises_1 = require("node:timers/promises");
+const common_js_1 = require("../imports/common.js");
+const index_js_1 = require("../index.js");
+const errors_js_1 = require("../errors.js");
+const rtp_js_1 = require("../imports/rtp.js");
+const utils_js_1 = require("../utils.js");
+const codecCompatibility_js_1 = require("./codecCompatibility.js");
+const stats_js_1 = require("./stats.js");
+const track_js_1 = require("./track.js");
+const log = (0, rtp_js_1.debug)('webrtc/peer/media/rtpSender');
+const RTP_HISTORY_SIZE = 128;
+const DEFAULT_PENDING_RTP_MAX_LENGTH = 256;
+const RTT_ALPHA = 0.85;
+function resolvePendingRtpOptions(pendingRtp) {
+    if (pendingRtp === true) {
+        return { enabled: true, maxLength: DEFAULT_PENDING_RTP_MAX_LENGTH };
+    }
+    if (pendingRtp == undefined || pendingRtp === false) {
+        return { enabled: false, maxLength: DEFAULT_PENDING_RTP_MAX_LENGTH };
+    }
+    const maxLength = typeof pendingRtp.maxLength === 'number' &&
+        Number.isFinite(pendingRtp.maxLength) &&
+        pendingRtp.maxLength >= 1
+        ? Math.floor(pendingRtp.maxLength)
+        : DEFAULT_PENDING_RTP_MAX_LENGTH;
+    return { enabled: pendingRtp.enabled ?? true, maxLength };
+}
+function freezeRtpContinuityOffsets(lastOutputSeq, lastOutputTimestamp, firstInputSeq, firstInputTimestamp, timestampStep) {
+    return {
+        seqOffset: (0, common_js_1.uint16Add)((0, common_js_1.uint16Add)(lastOutputSeq, 1), -firstInputSeq),
+        timestampOffset: (0, common_js_1.uint32Add)((0, common_js_1.uint32Add)(lastOutputTimestamp, timestampStep), -firstInputTimestamp)
+    };
+}
+class RTCRtpSender {
+    constructor(trackOrKind, options = {}) {
+        this.trackOrKind = trackOrKind;
+        this.type = 'sender';
+        this.ssrc = (0, node_crypto_1.randomBytes)(4).readUInt32BE(0);
+        this.rtxSsrc = (0, node_crypto_1.randomBytes)(4).readUInt32BE(0);
+        //The track id of the MSID (an application forwarding a track sets the one of the track it forwards, before it is
+        //negotiated)
+        this.trackId = (0, node_crypto_1.randomUUID)().toString();
+        this.onReady = new common_js_1.Event();
+        this.onRtcp = new common_js_1.Event();
+        this.onPictureLossIndication = new common_js_1.Event();
+        this.onGenericNack = new common_js_1.Event();
+        this.onStreamsChanged = new common_js_1.Event();
+        this.rtxSequenceNumber = (0, common_js_1.random16)();
+        this._redDistance = 2;
+        this.redEncoder = new rtp_js_1.RedEncoder(this._redDistance);
+        this.headerExtensions = [];
+        this.sendEncodings = [{}];
+        this.rtpTimestamp = 0;
+        this.octetCount = 0;
+        this.packetCount = 0;
+        this.headerBytesSent = 0;
+        this.totalRoundTripTime = 0;
+        this.roundTripTimeMeasurements = 0;
+        this.retransmittedPacketsSent = 0;
+        this.retransmittedBytesSent = 0;
+        this.nackCount = 0;
+        this.pliCount = 0;
+        this.firCount = 0;
+        this.receiverEstimatedMaxBitrate = 0n;
+        this.timestampOffset = 0;
+        this.seqOffset = 0;
+        this.rtpContinuityPending = false;
+        this.pendingTimestampStep = 1;
+        this.rtpCache = [];
+        this.pendingRtp = [];
+        this.drainingPendingRtp = false;
+        this.negotiatedCodecs = [];
+        this.dtlsDisposer = [];
+        this.track = null;
+        this.streamIds = [];
+        this.stopped = false;
+        this.rtcpRunning = false;
+        this.rtcpCancel = new AbortController();
+        const pendingRtp = resolvePendingRtpOptions(options.pendingRtp);
+        this.pendingRtpEnabled = pendingRtp.enabled;
+        this.pendingRtpMaxLength = pendingRtp.maxLength;
+        this.kind =
+            typeof this.trackOrKind === 'string'
+                ? this.trackOrKind
+                : this.trackOrKind.kind;
+        if (typeof trackOrKind !== 'string') {
+            this.registerTrack(trackOrKind);
+        }
+    }
+    get transport() {
+        return this.dtlsTransport ?? null;
+    }
+    get streamId() {
+        return this.streamIds[0];
+    }
+    set streamId(value) {
+        this.streamIds = value ? [value] : [];
+    }
+    setDtlsTransport(dtlsTransport) {
+        if (this.dtlsTransport) {
+            this.dtlsDisposer.forEach((dispose) => dispose());
+        }
+        this.dtlsTransport = dtlsTransport;
+        this.dtlsDisposer = [
+            this.dtlsTransport.onStateChange.subscribe((state) => {
+                if (state === 'connected') {
+                    this.onReady.execute();
+                    void this.drainPendingRtp();
+                }
+            }).unSubscribe
+        ];
+    }
+    get redDistance() {
+        return this._redDistance;
+    }
+    set redDistance(n) {
+        this._redDistance = n;
+        this.redEncoder.distance = n;
+    }
+    prepareSend(params) {
+        this.cname = params.rtcp?.cname;
+        this.mid = params.muxId;
+        this.headerExtensions = params.headerExtensions;
+        this.rtpStreamId = params.rtpStreamId ?? this.rtpStreamId;
+        this.repairedRtpStreamId = params.repairedRtpStreamId;
+        this.negotiatedCodecs = [...params.codecs];
+        this.selectCodec(this.track ? (0, track_js_1.getTrackSourceCodecs)(this.track) : undefined);
+        if (this.track) {
+            this.track.codec = this.codec;
+        }
+        void this.drainPendingRtp();
+    }
+    /**
+     * The codec sending: the first negotiated one, or the first one compatible with the codecs of the source of the
+     * track (the other side accepts any of the negotiated ones), with its retransmissions and its redundancy
+     * @param sourceCodecs
+     */
+    selectCodec(sourceCodecs) {
+        const codecs = this.negotiatedCodecs;
+        const primaryOf = (codec) => {
+            if (codec?.name.toLowerCase() !== 'red')
+                return codec;
+            const payloadType = Number((codec.parameters ?? '').split('/')[0]);
+            return codecs.find((each) => !['red', 'rtx'].includes(each.name.toLowerCase()) && each.payloadType === payloadType);
+        };
+        const compatible = (codec) => {
+            const primary = primaryOf(codec);
+            return !!primary && (!sourceCodecs || sourceCodecs.some((source) => (0, codecCompatibility_js_1.isCodecCompatible)(source, primary)));
+        };
+        const first = codecs.find((codec) => codec.name.toLowerCase() !== 'rtx') ?? codecs[0];
+        this.codec = codecs.find((codec) => codec.name.toLowerCase() !== 'rtx' && compatible(codec)) ?? first;
+        this.sendPrimaryCodec = primaryOf(this.codec);
+        this.rtxPayloadType = undefined;
+        this.redRedundantPayloadType = undefined;
+        codecs.forEach((codec) => {
+            const codecParams = (0, index_js_1.codecParametersFromString)(codec.parameters ?? '');
+            if (codec.name.toLowerCase() === 'rtx' &&
+                this.sendPrimaryCodec?.payloadType === codecParams['apt']) {
+                this.rtxPayloadType = codec.payloadType;
+            }
+        });
+        if (this.codec?.name.toLowerCase() === 'red' &&
+            this.sendPrimaryCodec != undefined) {
+            this.redRedundantPayloadType = this.sendPrimaryCodec.payloadType;
+        }
+    }
+    canSendRtp() {
+        return (!this.stopped && this.dtlsTransport?.state === 'connected' && !!this.codec);
+    }
+    settlePendingRtp(item, error) {
+        if (error == undefined) {
+            item.resolve();
+            return;
+        }
+        item.reject(error);
+    }
+    discardPendingRtp() {
+        const dropped = this.pendingRtp.splice(0);
+        for (const item of dropped) {
+            this.settlePendingRtp(item);
+        }
+    }
+    enqueuePendingRtp(rtp, resolve, reject) {
+        const packet = Buffer.isBuffer(rtp)
+            ? rtp_js_1.RtpPacket.deSerialize(rtp)
+            : rtp.clone();
+        this.pendingRtp.push({ packet, resolve, reject });
+        while (this.pendingRtp.length > this.pendingRtpMaxLength) {
+            const dropped = this.pendingRtp.shift();
+            if (dropped) {
+                this.settlePendingRtp(dropped);
+            }
+        }
+    }
+    /**
+   * Send queued RTP once DTLS is connected and a codec is set. Used only when
+   * pending RTP is enabled: `writeRtp` / `sendRtp` may run before ICE/DTLS
+   * completes; dropping those packets would lose the start of a media source.
+   * `drainingPendingRtp` blocks re-entry while `dispatchRtp` is awaited;
+   * packets enqueued during that wait are drained by the recursive call after
+   * the flag is cleared.
+   */
+    async drainPendingRtp() {
+        if (this.drainingPendingRtp) {
+            return;
+        }
+        this.drainingPendingRtp = true;
+        try {
+            while (this.pendingRtp.length > 0 && this.canSendRtp()) {
+                const item = this.pendingRtp.shift();
+                if (!this.canSendRtp()) {
+                    this.settlePendingRtp(item);
+                    continue;
+                }
+                try {
+                    await this.dispatchRtp(item.packet);
+                    this.settlePendingRtp(item);
+                }
+                catch (error) {
+                    this.settlePendingRtp(item, error);
+                }
+            }
+        }
+        finally {
+            this.drainingPendingRtp = false;
+        }
+        if (this.pendingRtp.length > 0 && this.canSendRtp()) {
+            await this.drainPendingRtp();
+        }
+    }
+    registerTrack(track) {
+        if (track.stopped)
+            throw new Error('track is ended');
+        (0, track_js_1.captureTrackSourceCodecs)(track);
+        if (this.disposeTrack) {
+            this.disposeTrack();
+        }
+        track.id = this.trackId;
+        const { unSubscribe: unSubscribeRtp } = track.onReceiveRtp.subscribe((0, common_js_1.detached)((rtp, extensions) => track.remote ? this.sendRtp(this.relayedRtp(rtp, extensions), true) : this.sendRtp(rtp), (error) => log('send rtp failed', error)));
+        const { unSubscribe: unSubscribeSourceChanged } = track.onSourceChanged.subscribe((header) => {
+            this.replaceRTP(header);
+        });
+        this.track = track;
+        this.disposeTrack = () => {
+            unSubscribeRtp();
+            unSubscribeSourceChanged();
+        };
+        if (this.codec) {
+            track.codec = this.codec;
+        }
+    }
+    /**
+     * A packet received on a connection, to send on this one: its header extensions are the ones of that connection,
+     * by their ids there. The ones about the media (the audio level, the video orientation) go on with their ids here,
+     * if negotiated; the ones of the transport are this connection's own.
+     * @param rtp
+     * @param extensions The extensions of the packet, by URI
+     */
+    relayedRtp(rtp, extensions = {}) {
+        const packet = rtp.clone();
+        packet.header.extensions = this.headerExtensions.flatMap((extension) => {
+            switch (extension.uri) {
+                case rtp_js_1.RTP_EXTENSION_URI.audioLevelIndication: {
+                    const level = extensions[extension.uri];
+                    return level
+                        ? [{ id: extension.id, payload: Buffer.from([(level.v ? 0x80 : 0) | (level.level & 0x7f)]) }]
+                        : [];
+                }
+                case rtp_js_1.RTP_EXTENSION_URI.videoOrientation: {
+                    const orientation = extensions[extension.uri];
+                    return orientation
+                        ? [{
+                                id: extension.id,
+                                payload: Buffer.from([
+                                    (orientation.c << 3) | (orientation.f << 2) | (orientation.r1 << 1) | orientation.r0
+                                ])
+                            }]
+                        : [];
+                }
+                default:
+                    return [];
+            }
+        });
+        return packet;
+    }
+    setStreams(streams = []) {
+        const streamIds = [...new Set(streams.map((stream) => stream.id))];
+        const changed = streamIds.join('\n') !== (this.streamIds ?? []).join('\n');
+        this.streamIds = streamIds;
+        //The streams are negotiated (W3C: the negotiation-needed flag)
+        if (changed)
+            this.onStreamsChanged.execute();
+    }
+    setSendEncodings(encodings = []) {
+        this.sendEncodings =
+            encodings.length > 0
+                ? encodings.map((encoding) => ({ ...encoding }))
+                : [{}];
+        const rid = this.sendEncodings.find((encoding) => typeof encoding.rid === 'string')?.rid;
+        if (typeof rid === 'string') {
+            this.rtpStreamId = rid;
+        }
+    }
+    async replaceTrack(track) {
+        if (track === null) {
+            this.detachTrack();
+            return;
+        }
+        if (track.stopped)
+            throw new Error('track is ended');
+        (0, track_js_1.captureTrackSourceCodecs)(track);
+        const sourceCodecs = (0, track_js_1.getTrackSourceCodecs)(track);
+        //Any codec negotiated may send it: the other side accepts each of them
+        if (this.sendPrimaryCodec != undefined &&
+            sourceCodecs != undefined &&
+            !this.negotiatedCodecs.some((codec) => !['red', 'rtx'].includes(codec.name.toLowerCase()) &&
+                sourceCodecs.some((source) => (0, codecCompatibility_js_1.isCodecCompatible)(source, codec)))) {
+            throw (0, errors_js_1.createWebRtcDomException)('InvalidModificationError', `Track codec ${sourceCodecs.map((codec) => codec.mimeType).join(', ')} is incompatible with negotiated codecs ${this.negotiatedCodecs.map((codec) => codec.mimeType).join(', ')}`);
+        }
+        if (this.negotiatedCodecs.length)
+            this.selectCodec(sourceCodecs);
+        if (this.sequenceNumber != undefined) {
+            this.scheduleRtpContinuity();
+        }
+        this.registerTrack(track);
+        log('replaceTrack', 'ssrc', track.ssrc, 'rid', track.rid);
+    }
+    /**
+   * Detach the current track without stopping the sender (`removeTrack`).
+   * The same sender can resume sending after `replaceTrack(track)`.
+   */
+    detachTrack() {
+        this.rtpContinuityPending = false;
+        this.discardPendingRtp();
+        if (this.disposeTrack) {
+            this.disposeTrack();
+            this.disposeTrack = undefined;
+        }
+        this.track = null;
+    }
+    stop({ keepTrack = false } = {}) {
+        this.stopped = true;
+        this.rtpContinuityPending = false;
+        this.discardPendingRtp();
+        this.rtcpRunning = false;
+        this.rtcpCancel.abort();
+        if (this.disposeTrack) {
+            this.disposeTrack();
+            this.disposeTrack = undefined;
+        }
+        if (!keepTrack) {
+            this.track = null;
+        }
+    }
+    async runRtcp() {
+        if (this.rtcpRunning || this.stopped)
+            return;
+        this.rtcpRunning = true;
+        try {
+            while (this.rtcpRunning) {
+                await (0, promises_1.setTimeout)(500 + Math.random() * 1000, undefined, {
+                    signal: this.rtcpCancel.signal
+                });
+                const ntpTimestamp = this.sentAtEpochMs === undefined ? 0n : (0, utils_js_1.ntpTime)(this.sentAtEpochMs);
+                const packets = [
+                    new rtp_js_1.RtcpSrPacket({
+                        ssrc: this.ssrc,
+                        senderInfo: new rtp_js_1.RtcpSenderInfo({
+                            ntpTimestamp: ntpTimestamp,
+                            rtpTimestamp: this.rtpTimestamp,
+                            packetCount: this.packetCount,
+                            octetCount: this.octetCount
+                        })
+                    })
+                ];
+                this.lastSRtimestamp = (0, utils_js_1.compactNtp)(ntpTimestamp);
+                this.lastSentSRTimestamp = (0, utils_js_1.timestampSeconds)();
+                if (this.cname) {
+                    packets.push(new rtp_js_1.RtcpSourceDescriptionPacket({
+                        chunks: [
+                            new rtp_js_1.SourceDescriptionChunk({
+                                source: this.ssrc,
+                                items: [
+                                    new rtp_js_1.SourceDescriptionItem({ type: 1, text: this.cname })
+                                ]
+                            })
+                        ]
+                    }));
+                }
+                try {
+                    await this.dtlsTransport.sendRtcp(packets);
+                }
+                catch (error) {
+                    log('sendRtcp failed', error);
+                    await (0, promises_1.setTimeout)(500 + Math.random() * 1000);
+                }
+            }
+        }
+        catch (error) {
+            //The loop of the RTCP reports ends with the transport
+            log('rtcp loop ended', error);
+        }
+    }
+    /**
+   * Send an RTP packet. Pending RTP is disabled by default: the packet is
+   * written immediately if DTLS is connected and a codec is set, otherwise
+   * dropped.
+   *
+   * When pending RTP is enabled, the packet is queued until it can be sent.
+   * The returned promise then settles for this packet only:
+   * - resolve: DTLS send completed, or the packet was dropped by `stop()`,
+   *   `replaceTrack(null)`, or pending-queue overflow
+   * - reject: DTLS send threw while writing this packet
+   *
+   * Later `sendRtp()` calls stay pending until their own packet is sent or
+   * dropped, even if a drain is already in progress.
+   * @param rtp
+   * @param owned The packet is a copy of the sender's own (a packet relayed), changed in place instead of copied again
+   */
+    async sendRtp(rtp, owned = false) {
+        if (this.stopped) {
+            return;
+        }
+        if (!this.pendingRtpEnabled) {
+            if (!this.canSendRtp()) {
+                return;
+            }
+            await this.dispatchRtp(rtp, owned);
+            return;
+        }
+        await new Promise((resolve, reject) => {
+            this.enqueuePendingRtp(rtp, resolve, reject);
+            void this.drainPendingRtp();
+        });
+    }
+    /**
+   * Schedule RTP continuity rewrite for the next packet that is actually sent.
+   * The header argument is kept for API compatibility and is not used to compute
+   * offsets. `discontinuity` does not change sequence or timestamp mapping.
+   * `timestampStep` (default 1) is the only way to choose the timestamp increment
+   * at the source-switch boundary.
+   */
+    replaceRTP(header, discontinuity = false, timestampStep = 1) {
+        this.scheduleRtpContinuity(timestampStep);
+        log('replaceRTP', this.sequenceNumber, header.sequenceNumber, discontinuity, timestampStep);
+    }
+    scheduleRtpContinuity(timestampStep = 1) {
+        this.rtpContinuityPending = true;
+        this.pendingTimestampStep = timestampStep;
+        this.rtpCache = [];
+    }
+    async dispatchRtp(rtp, owned = false) {
+        if (!this.canSendRtp()) {
+            return;
+        }
+        const codec = this.codec;
+        if (!codec) {
+            return;
+        }
+        rtp = Buffer.isBuffer(rtp) ? rtp_js_1.RtpPacket.deSerialize(rtp) : owned ? rtp : rtp.clone();
+        const { header, payload } = rtp;
+        const inputSequenceNumber = header.sequenceNumber;
+        const inputTimestamp = header.timestamp;
+        if (this.rtpContinuityPending) {
+            if (this.sequenceNumber != undefined && this.timestamp != undefined) {
+                const offsets = freezeRtpContinuityOffsets(this.sequenceNumber, this.timestamp, inputSequenceNumber, inputTimestamp, this.pendingTimestampStep);
+                this.seqOffset = offsets.seqOffset;
+                this.timestampOffset = offsets.timestampOffset;
+            }
+            this.rtpContinuityPending = false;
+            this.pendingTimestampStep = 1;
+        }
+        header.ssrc = this.ssrc;
+        header.payloadType = codec.payloadType;
+        header.timestamp = (0, common_js_1.uint32Add)(header.timestamp, this.timestampOffset);
+        header.sequenceNumber = (0, common_js_1.uint16Add)(header.sequenceNumber, this.seqOffset);
+        this.timestamp = header.timestamp;
+        this.sequenceNumber = header.sequenceNumber;
+        const sentAtEpochMs = performance.timeOrigin + performance.now();
+        let transportSequenceNumber;
+        //The extensions of this sender, then the ones of the packet (which win over them), in the order of their ids:
+        //plain loops, as it is done for each packet
+        const extensions = [];
+        for (const extension of this.headerExtensions) {
+            let payload;
+            switch (extension.uri) {
+                case rtp_js_1.RTP_EXTENSION_URI.sdesMid:
+                    if (this.mid)
+                        payload = (0, rtp_js_1.serializeSdesMid)(this.mid);
+                    break;
+                // todo : sender simulcast unsupported now
+                case rtp_js_1.RTP_EXTENSION_URI.sdesRTPStreamID:
+                    if (this.rtpStreamId)
+                        payload = (0, rtp_js_1.serializeSdesRTPStreamID)(this.rtpStreamId);
+                    break;
+                // todo : sender simulcast unsupported now
+                case rtp_js_1.RTP_EXTENSION_URI.repairedRtpStreamId:
+                    if (this.repairedRtpStreamId)
+                        payload = (0, rtp_js_1.serializeRepairedRtpStreamId)(this.repairedRtpStreamId);
+                    break;
+                case rtp_js_1.RTP_EXTENSION_URI.transportWideCC:
+                    transportSequenceNumber = this.nextTransportSequenceNumber();
+                    payload = (0, rtp_js_1.serializeTransportWideCC)(transportSequenceNumber);
+                    break;
+                case rtp_js_1.RTP_EXTENSION_URI.absSendTime:
+                    payload = (0, utils_js_1.absSendTimeOf)(sentAtEpochMs);
+                    break;
+            }
+            if (payload)
+                extensions.push({ id: extension.id, payload });
+        }
+        for (const ext of header.extensions) {
+            let exist;
+            for (const each of extensions) {
+                if (each.id === ext.id) {
+                    exist = each;
+                    break;
+                }
+            }
+            if (exist) {
+                exist.payload = ext.payload;
+            }
+            else {
+                extensions.push(ext);
+            }
+        }
+        for (let index = 1; index < extensions.length; index++) {
+            if (extensions[index - 1].id > extensions[index].id) {
+                extensions.sort((a, b) => a.id - b.id);
+                break;
+            }
+        }
+        header.extensions = extensions;
+        this.sentAtEpochMs = sentAtEpochMs;
+        this.rtpTimestamp = header.timestamp;
+        this.octetCount += payload.length;
+        this.headerBytesSent += header.serializeSize;
+        this.packetCount = (0, common_js_1.uint32Add)(this.packetCount, 1);
+        this.rtpCache[header.sequenceNumber % RTP_HISTORY_SIZE] = rtp;
+        let rtpPayload = payload;
+        if (this.redRedundantPayloadType) {
+            this.redEncoder.push({
+                block: rtpPayload,
+                timestamp: header.timestamp,
+                blockPT: this.redRedundantPayloadType
+            });
+            const red = this.redEncoder.build();
+            rtpPayload = red.serialize();
+        }
+        const sendingAtMs = (0, utils_js_1.milliTime)();
+        const size = await this.dtlsTransport.sendRtp(rtpPayload, header);
+        //Started once: the call of each packet would only make a promise
+        if (!this.rtcpRunning && !this.stopped)
+            this.runRtcp().catch((error) => log('rtcp failed', error));
+        if (transportSequenceNumber !== undefined) {
+            this.packetSent(transportSequenceNumber, size, sendingAtMs);
+        }
+    }
+    /**
+     * The next transport-wide sequence number, shared by the senders of the transport
+     */
+    nextTransportSequenceNumber() {
+        this.dtlsTransport.transportSequenceNumber = (0, common_js_1.uint16Add)(this.dtlsTransport.transportSequenceNumber, 1);
+        return this.dtlsTransport.transportSequenceNumber;
+    }
+    /**
+     * A packet sent with a transport-wide sequence number, for the bandwidth estimation of the transport
+     * @param wideSeq
+     * @param size
+     * @param sendingAtMs
+     */
+    packetSent(wideSeq, size, sendingAtMs) {
+        const sentInfo = {
+            wideSeq,
+            size,
+            sendingAtMs,
+            sentAtMs: (0, utils_js_1.milliTime)()
+        };
+        this.dtlsTransport.senderBWE.rtpPacketSent(sentInfo);
+    }
+    /**
+     * Send a packet again, for a negative acknowledgement: with a new transport-wide sequence number and send time, in
+     * a retransmission packet when RTX is negotiated
+     * @param cached
+     */
+    async resendRtp(cached) {
+        let transportSequenceNumber;
+        const extensions = cached.header.extensions.map((extension) => {
+            const uri = this.headerExtensions.find((negotiated) => negotiated.id === extension.id)?.uri;
+            switch (uri) {
+                case rtp_js_1.RTP_EXTENSION_URI.transportWideCC:
+                    transportSequenceNumber = this.nextTransportSequenceNumber();
+                    return { id: extension.id, payload: (0, rtp_js_1.serializeTransportWideCC)(transportSequenceNumber) };
+                case rtp_js_1.RTP_EXTENSION_URI.absSendTime:
+                    return { id: extension.id, payload: (0, utils_js_1.absSendTimeOf)(performance.timeOrigin + performance.now()) };
+                default:
+                    return extension;
+            }
+        });
+        let packet = new rtp_js_1.RtpPacket(new rtp_js_1.RtpHeader({ ...cached.header, extensions }), cached.payload);
+        if (this.rtxPayloadType != undefined) {
+            packet = (0, rtp_js_1.wrapRtx)(packet, this.rtxPayloadType, this.rtxSequenceNumber, this.rtxSsrc);
+            this.rtxSequenceNumber = (0, common_js_1.uint16Add)(this.rtxSequenceNumber, 1);
+        }
+        this.retransmittedPacketsSent++;
+        this.retransmittedBytesSent += packet.payload.length;
+        this.headerBytesSent += packet.header.serializeSize;
+        const sendingAtMs = (0, utils_js_1.milliTime)();
+        const size = await this.dtlsTransport.sendRtp(packet.payload, packet.header);
+        if (transportSequenceNumber !== undefined) {
+            this.packetSent(transportSequenceNumber, size, sendingAtMs);
+        }
+    }
+    handleRtcpPacket(rtcpPacket) {
+        switch (rtcpPacket.type) {
+            case rtp_js_1.RtcpSrPacket.type:
+            case rtp_js_1.RtcpRrPacket.type:
+                {
+                    const packet = rtcpPacket;
+                    packet.reports
+                        .filter((report) => report.ssrc === this.ssrc)
+                        .forEach((report) => {
+                        this.remotePacketsLost = report.packetsLost;
+                        this.remoteFractionLost = report.fractionLost / 256;
+                        if (this.lastSRtimestamp === report.lsr && report.dlsr) {
+                            if (this.lastSentSRTimestamp) {
+                                const rtt = (0, utils_js_1.timestampSeconds)() -
+                                    this.lastSentSRTimestamp -
+                                    report.dlsr / 65536;
+                                this.totalRoundTripTime += rtt;
+                                this.roundTripTimeMeasurements++;
+                                if (this.rtt === undefined) {
+                                    this.rtt = rtt;
+                                }
+                                else {
+                                    this.rtt = RTT_ALPHA * this.rtt + (1 - RTT_ALPHA) * rtt;
+                                }
+                            }
+                        }
+                    });
+                }
+                break;
+            case rtp_js_1.RtcpTransportLayerFeedback.type:
+                {
+                    const packet = rtcpPacket;
+                    switch (packet.feedback.count) {
+                        case rtp_js_1.GenericNack.count:
+                            {
+                                const feedback = packet.feedback;
+                                this.nackCount++;
+                                feedback.lost.forEach((0, common_js_1.detached)(async (seqNum) => {
+                                    const packet = this.rtpCache[seqNum % RTP_HISTORY_SIZE];
+                                    if (packet && packet.header.sequenceNumber === seqNum) {
+                                        await this.resendRtp(packet);
+                                    }
+                                }, (error) => log('retransmission failed', error)));
+                                this.onGenericNack.execute(feedback);
+                            }
+                            break;
+                    }
+                }
+                break;
+            case rtp_js_1.RtcpPayloadSpecificFeedback.type:
+                {
+                    const packet = rtcpPacket;
+                    switch (packet.feedback.count) {
+                        case rtp_js_1.ReceiverEstimatedMaxBitrate.count:
+                            {
+                                const feedback = packet.feedback;
+                                this.receiverEstimatedMaxBitrate = feedback.bitrate;
+                            }
+                            break;
+                        case rtp_js_1.PictureLossIndication.count:
+                            {
+                                this.pliCount++;
+                                this.onPictureLossIndication.execute();
+                            }
+                            break;
+                    }
+                }
+                break;
+        }
+        this.onRtcp.execute(rtcpPacket);
+    }
+    // todo impl
+    getParameters() {
+        return {
+            encodings: this.sendEncodings.map((encoding) => ({ ...encoding }))
+        };
+    }
+    // todo impl
+    setParameters(params) {
+        if (params.encodings) {
+            this.setSendEncodings(params.encodings);
+        }
+    }
+    get outboundRtpStatsId() {
+        return (0, stats_js_1.generateStatsId)('outbound-rtp', this.trackId);
+    }
+    get mediaSourceStatsId() {
+        return (0, stats_js_1.generateStatsId)('media-source', this.trackId);
+    }
+    get remoteInboundRtpStatsId() {
+        return (0, stats_js_1.generateStatsId)('remote-inbound-rtp', this.trackId);
+    }
+    getStatsRootIds() {
+        return [this.outboundRtpStatsId];
+    }
+    collectStats(timestamp) {
+        const stats = [];
+        const transportId = this.dtlsTransport
+            ? (0, stats_js_1.generateStatsId)('transport', this.dtlsTransport.id)
+            : undefined;
+        const codecId = this.codec && transportId
+            ? (0, stats_js_1.generateCodecStatsId)(transportId, this.codec.payloadType, this.trackId)
+            : undefined;
+        // Outbound RTP stats
+        const outboundRtpStats = {
+            type: 'outbound-rtp',
+            id: this.outboundRtpStatsId,
+            timestamp,
+            ssrc: this.ssrc,
+            kind: this.kind,
+            transportId,
+            codecId,
+            mid: this.mid,
+            packetsSent: this.packetCount,
+            bytesSent: this.octetCount,
+            headerBytesSent: this.headerBytesSent,
+            retransmittedPacketsSent: this.retransmittedPacketsSent || undefined,
+            retransmittedBytesSent: this.retransmittedBytesSent || undefined,
+            rtxSsrc: this.rtxPayloadType ? this.rtxSsrc : undefined,
+            mediaSourceId: this.track ? this.mediaSourceStatsId : undefined,
+            remoteId: this.rtt !== undefined ||
+                this.remotePacketsLost !== undefined ||
+                this.remoteFractionLost !== undefined
+                ? this.remoteInboundRtpStatsId
+                : undefined,
+            nackCount: this.nackCount || undefined,
+            pliCount: this.pliCount || undefined,
+            firCount: this.firCount || undefined
+        };
+        stats.push(outboundRtpStats);
+        // Media source stats
+        if (this.track) {
+            const mediaSourceStats = {
+                type: 'media-source',
+                id: this.mediaSourceStatsId,
+                timestamp,
+                trackIdentifier: this.track.id ?? this.trackId,
+                kind: this.kind
+            };
+            stats.push(mediaSourceStats);
+        }
+        if (this.codec && transportId) {
+            const codecStats = {
+                type: 'codec',
+                id: codecId,
+                timestamp,
+                payloadType: this.codec.payloadType,
+                transportId,
+                mimeType: this.codec.mimeType,
+                clockRate: this.codec.clockRate,
+                channels: this.codec.channels,
+                sdpFmtpLine: this.codec.parameters
+            };
+            stats.push(codecStats);
+        }
+        // Remote inbound RTP stats (if we have RTT)
+        if (this.rtt !== undefined ||
+            this.remotePacketsLost !== undefined ||
+            this.remoteFractionLost !== undefined) {
+            const remoteInboundStats = {
+                type: 'remote-inbound-rtp',
+                id: this.remoteInboundRtpStatsId,
+                timestamp,
+                ssrc: this.ssrc,
+                kind: this.kind,
+                transportId,
+                codecId: outboundRtpStats.codecId,
+                localId: outboundRtpStats.id,
+                roundTripTime: this.rtt,
+                totalRoundTripTime: this.totalRoundTripTime,
+                roundTripTimeMeasurements: this.roundTripTimeMeasurements,
+                packetsLost: this.remotePacketsLost,
+                fractionLost: this.remoteFractionLost
+            };
+            stats.push(remoteInboundStats);
+        }
+        return stats;
+    }
+    async getStats() {
+        const timestamp = (0, stats_js_1.getStatsTimestamp)();
+        const stats = this.collectStats(timestamp);
+        if (this.dtlsTransport) {
+            stats.push(...(await this.dtlsTransport.getStats(timestamp)));
+        }
+        return (0, stats_js_1.buildStatsReport)(stats, this.getStatsRootIds());
+    }
+}
+exports.RTCRtpSender = RTCRtpSender;