@relaymessenger/livekit 0.1.0-staging.15 → 0.1.0-staging.17

package/dist/engine-werift.js

@@ -1,457 +0,0 @@
-/**
- * Relay's default call participant engine: werift (pure-TypeScript WebRTC:
- * ICE, DTLS, SRTP, RTP) plus `@evan/opus` (prebuilt libopus, WASM fallback).
- *
- * API sources read for this file, in node_modules/werift/lib/webrtc/src:
- * - peerConnection.d.ts: `new RTCPeerConnection({ codecs: { audio: [...] } })`,
- *   `addTransceiver(track, { direction })`, the W3C-style `ontrack`,
- *   `onconnectionstatechange`, `localDescription`, `close()`, and the
- *   `"icegatheringstatechange"` / `"connectionstatechange"` events emitted
- *   through `EventTarget.addEventListener` (peerConnection.js:317, :330).
- *   `RTCConfiguration.iceServers: RTCIceServer[]` ({ urls, username?,
- *   credential? }) and `iceTransportPolicy: "all" | "relay"` (peerConnection.d.ts
- *   :264-273); utils.js:110-179 parses `stun|stuns|turn|turns:` URLs with
- *   `?transport=`, and `secureTransportManager.js:121` maps `"relay"` to
- *   `forceTurn`. The W3C setters `onicecandidate`, `onicegatheringstatechange`
- *   and `oniceconnectionstatechange` (peerConnection.d.ts:67-82, fired at
- *   peerConnection.js:314-341) feed the transport's ICE diagnostics.
- * - media/track.d.ts: `MediaStreamTrack({ kind })`, `writeRtp(RtpPacket)`,
- *   `onReceiveRtp: Event<[RtpPacket, Extensions?]>` with `subscribe()`.
- * - media/rtpSender.js:544-549: the sender rewrites ssrc and payloadType and
- *   offsets sequenceNumber/timestamp, so the local track starts both at 0.
- * - ../../rtp/src/codec/opus.d.ts: `OpusRtpPayload.deSerialize(buffer)`.
- * - node_modules/@evan/opus/lib.d.ts: `Encoder.encode(pcm16)` returns one Opus
- *   packet; `Decoder.decode(packet)` returns interleaved PCM16 bytes;
- *   `new Decoder({ channels?: 1 | 2, sample_rate?: 8000 | 12000 | 16000 |
- *   24000 | 48000 })` decodes any Opus stream to that count and rate (libopus
- *   `opus_decoder_create(Fs, channels)`), which is how the sink honours
- *   `RelayInboundAudioFormat`.
- * Participant shapes copied from the headless phone that joined Relay's room
- * through Cloudflare's SFU on 2026-09-21 (_runtime/bin/headless-phone-webrtc.mjs
- * lines 93-160): `bundlePolicy: "max-bundle", iceServers: []` (Cloudflare
- * supplies its ICE candidates in the answer), the local track is id
- * "microphone" in stream "relay-call" (Relay-iOS RelayWebRTCAudioTransport),
- * the first RTP packet carries the marker bit, empty payloads are skipped and a
- * decode failure never tears the sink down.
- */
-import { Decoder, Encoder } from "@evan/opus";
-import { MediaStreamTrack, OpusRtpPayload, RTCPeerConnection, RTCRtpCodecParameters, RTCRtpHeaderExtensionParameters, RtpHeader, RtpPacket, useFIR, useNACK, usePLI, } from "werift";
-import { createWeriftVideoFactory } from "./engine-werift-video.js";
-/** Opus on the wire is always 48 kHz; Relay sends stereo so callers may pass either channel count. */
-export const WERIFT_SAMPLE_RATE = 48_000;
-export const WERIFT_CHANNEL_COUNT = 2;
-/** One RTP packet carries 20 ms: 960 frames at 48 kHz, timestamp advances by 960. */
-export const WERIFT_PACKET_MS = 20;
-const PACKET_FRAMES = (WERIFT_SAMPLE_RATE * WERIFT_PACKET_MS) / 1000;
-const PACKET_SAMPLES = PACKET_FRAMES * WERIFT_CHANNEL_COUNT;
-/** Dynamic payload type; werift's sender replaces it with the negotiated one. */
-const LOCAL_PAYLOAD_TYPE = 111;
-/** Packets counted for `diagnostics()`; the recent window is the last 5 s. */
-export const STATS_WINDOW_MS = 5_000;
-/** Monotone packet counter with first/last epoch timestamps and a 5 s window. */
-class PacketClock {
-    count = 0;
-    firstAt;
-    lastAt;
-    #recent = [];
-    mark(now = Date.now()) {
-        this.count += 1;
-        this.firstAt ??= now;
-        this.lastAt = now;
-        this.#recent.push(now);
-        this.#prune(now);
-    }
-    recent(now = Date.now()) {
-        this.#prune(now);
-        return this.#recent.length;
-    }
-    #prune(now) {
-        const floor = now - STATS_WINDOW_MS;
-        let drop = 0;
-        while (drop < this.#recent.length && this.#recent[drop] < floor)
-            drop += 1;
-        if (drop)
-            this.#recent.splice(0, drop);
-    }
-}
-/**
- * A pacer more than this far behind its clock re-anchors instead of bursting
- * the backlog. LiveKit's native `AudioSource` (rust-sdks
- * webrtc-sys/src/audio_track.cpp:161-188) runs a 10 ms libwebrtc
- * `RepeatingTask`, which schedules against an absolute run time and makes up
- * every lost tick with zero delay, unbounded (rtc_base/task_utils/
- * repeating_task.cc:86-89, `delay -= lost_time; delay = max(delay, 0)`). The
- * bound is ours: 10 packets, so an event-loop hiccup is made up in full and a
- * real stall does not dump seconds of audio on the receiver at once.
- * Sources saved under _sources/audio-pacing-20260922.
- */
-export const WERIFT_MAX_CATCH_UP_MS = 200;
-/**
- * Wall-clock RTP audio pacing: packet `n` of a run is due at `start + n × 20 ms`,
- * so the average rate is exactly one packet per 20 ms however late the timer
- * fires. `take(now)` returns how many packets are due and counts them as sent.
- */
-export class RtpAudioPacer {
-    packetMs;
-    maxCatchUpMs;
-    #startedAt;
-    #sent = 0;
-    constructor(packetMs = WERIFT_PACKET_MS, maxCatchUpMs = WERIFT_MAX_CATCH_UP_MS) {
-        this.packetMs = packetMs;
-        this.maxCatchUpMs = maxCatchUpMs;
-    }
-    /** Packets due by `now` and not yet sent; the first call of a run sends one at once. */
-    take(now) {
-        if (this.#startedAt === undefined || now - this.nextDueAt() > this.maxCatchUpMs) {
-            this.#startedAt = now;
-            this.#sent = 0;
-        }
-        const due = Math.floor((now - this.#startedAt) / this.packetMs) + 1 - this.#sent;
-        if (due <= 0)
-            return 0;
-        this.#sent += due;
-        return due;
-    }
-    /** When the next packet is due; `-Infinity` before the first `take` of a run. */
-    nextDueAt() {
-        return this.#startedAt === undefined ? -Infinity : this.#startedAt + this.#sent * this.packetMs;
-    }
-    /** Forget the run; the next `take` starts a new clock. */
-    reset() {
-        this.#startedAt = undefined;
-        this.#sent = 0;
-    }
-}
-/** Convert any PCM16 frame to interleaved 48 kHz stereo by linear interpolation. */
-export const toWireFormat = (data) => {
-    const { samples, sampleRate, channelCount } = data;
-    const inputFrames = Math.floor(samples.length / channelCount);
-    if (sampleRate === WERIFT_SAMPLE_RATE && channelCount === WERIFT_CHANNEL_COUNT) {
-        return samples;
-    }
-    const outputFrames = Math.round((inputFrames * WERIFT_SAMPLE_RATE) / sampleRate);
-    const out = new Int16Array(outputFrames * WERIFT_CHANNEL_COUNT);
-    const ratio = sampleRate / WERIFT_SAMPLE_RATE;
-    const leftIndex = 0;
-    const rightIndex = channelCount > 1 ? 1 : 0;
-    for (let frame = 0; frame < outputFrames; frame += 1) {
-        const position = frame * ratio;
-        const base = Math.min(Math.floor(position), inputFrames - 1);
-        const next = Math.min(base + 1, inputFrames - 1);
-        const fraction = position - Math.floor(position);
-        const left = samples[base * channelCount + leftIndex] * (1 - fraction)
-            + samples[next * channelCount + leftIndex] * fraction;
-        const right = samples[base * channelCount + rightIndex] * (1 - fraction)
-            + samples[next * channelCount + rightIndex] * fraction;
-        out[frame * 2] = Math.round(left);
-        out[frame * 2 + 1] = Math.round(right);
-    }
-    return out;
-};
-/**
- * PCM16 in, paced 20 ms Opus RTP out. Frames are accumulated into whole
- * packets; an `RtpAudioPacer` on the monotonic clock decides how many are
- * due, so the wire sees exactly 50 packets/s regardless of how the adapter
- * chunks its writes or how late the timer fires. A timer that fired once per
- * packet (the old `setInterval`) lost every late tick while the RTP timestamp
- * still advanced 960 per packet: @relay's calls on 2026-09-22 sent 48.66-48.98
- * packets/s and the phone's jitter buffer ran dry. Adapters may push
- * far ahead of real time: `queuedMs()` is what has not left yet and
- * `waitForDrain()` resolves once the queued application audio has been
- * written (LiveKit's `AudioSource.queuedDuration` / `waitForPlayout` shape).
- *
- * From `start()` (the transport calls it when the peer first connects) until
- * the track stops, the pump never pauses: a tick with no application audio
- * queued writes an Opus silence frame instead, as a live microphone track
- * does. Cloudflare's SFU refuses to pull a published track that has carried
- * no RTP: `tracks/new` for the remote track returned `empty_track_error` "No
- * track data from remote peer" after about 8.2 s on staging 2026-09-22, and
- * the same pull returned in about 470 ms with silence flowing. Silence and
- * application packets share one sequence and timestamp line, so queued audio
- * replaces silence on the next tick with no gap and no jump.
- */
-class WeriftAudioSource {
-    #encoder = new Encoder({
-        channels: WERIFT_CHANNEL_COUNT,
-        sample_rate: WERIFT_SAMPLE_RATE,
-        application: "voip",
-    });
-    /** 20 ms of digital silence, encoded once by the same encoder and reused. */
-    #silence = Buffer.from(this.#encoder.encode(new Int16Array(PACKET_SAMPLES)));
-    #track = new MediaStreamTrack({ kind: "audio", id: "microphone", streamId: "relay-call" });
-    #packets = [];
-    /** Every RTP packet written, silence included. */
-    #rtp = new PacketClock();
-    #opusPackets = 0;
-    #applicationRtpPackets = 0;
-    #silencePackets = 0;
-    #pending = new Int16Array(0);
-    #sequenceNumber = 1;
-    #timestamp = 0;
-    #first = true;
-    #pump;
-    #pacer = new RtpAudioPacer();
-    #started = false;
-    #stopped = false;
-    #drainWaiters = new Set();
-    stats() {
-        return {
-            opusPackets: this.#opusPackets,
-            rtpPackets: this.#applicationRtpPackets,
-            silencePackets: this.#silencePackets,
-            firstRtpAt: this.#rtp.firstAt,
-            lastRtpAt: this.#rtp.lastAt,
-            recentRtpPackets: this.#rtp.recent(),
-            queued: this.#packets.length,
-            pacerAlive: this.#pump !== undefined,
-        };
-    }
-    createTrack() {
-        const track = this.#track;
-        const stop = track.stop;
-        track.stop = () => {
-            this.#stopped = true;
-            this.clear();
-            this.#stopPump();
-            stop();
-        };
-        return track;
-    }
-    start() {
-        if (this.#started || this.#stopped)
-            return;
-        this.#started = true;
-        this.#startPump();
-    }
-    queuedMs() {
-        return this.#packets.length * WERIFT_PACKET_MS
-            + (this.#pending.length / WERIFT_CHANNEL_COUNT / WERIFT_SAMPLE_RATE) * 1_000;
-    }
-    waitForDrain() {
-        this.#flushPending();
-        if (this.#drained())
-            return Promise.resolve();
-        return new Promise((resolve) => this.#drainWaiters.add(resolve));
-    }
-    clear() {
-        this.#packets.length = 0;
-        this.#pending = new Int16Array(0);
-        if (!this.#started)
-            this.#stopPump();
-        this.#notifyDrained();
-    }
-    onData(data) {
-        if (this.#stopped)
-            return;
-        const wire = toWireFormat(data);
-        const merged = new Int16Array(this.#pending.length + wire.length);
-        merged.set(this.#pending);
-        merged.set(wire, this.#pending.length);
-        let offset = 0;
-        while (merged.length - offset >= PACKET_SAMPLES) {
-            const frame = merged.subarray(offset, offset + PACKET_SAMPLES);
-            this.#packets.push(Buffer.from(this.#encoder.encode(frame)));
-            this.#opusPackets += 1;
-            offset += PACKET_SAMPLES;
-        }
-        this.#pending = merged.slice(offset);
-        this.#startPump();
-    }
-    /** Pad a sub-packet remainder with silence so the tail of a segment reaches the wire. */
-    #flushPending() {
-        if (this.#stopped || this.#pending.length === 0)
-            return;
-        const padded = new Int16Array(PACKET_SAMPLES);
-        padded.set(this.#pending);
-        this.#pending = new Int16Array(0);
-        this.#packets.push(Buffer.from(this.#encoder.encode(padded)));
-        this.#opusPackets += 1;
-        this.#startPump();
-    }
-    /** Application audio only: the silence the started pump keeps writing never counts. */
-    #drained() {
-        return this.#packets.length === 0
-            && this.#pending.length === 0
-            && (this.#started || this.#pump === undefined);
-    }
-    #notifyDrained() {
-        if (!this.#drained())
-            return;
-        for (const resolve of [...this.#drainWaiters]) {
-            this.#drainWaiters.delete(resolve);
-            resolve();
-        }
-    }
-    /**
-     * The first packet of a run leaves at once; after that each wake sends
-     * every packet the pacer says is due and sleeps until the next one is due,
-     * the absolute schedule of libwebrtc's `RepeatingTask`. Before `start()` the
-     * pump stops once the queue is empty; after it, the pump runs until the
-     * track stops.
-     */
-    #startPump() {
-        if (this.#pump || this.#stopped)
-            return;
-        if (!this.#started && this.#packets.length === 0)
-            return;
-        this.#tick();
-    }
-    #tick() {
-        if (this.#stopped)
-            return;
-        const due = this.#pacer.take(performance.now());
-        for (let sent = 0; sent < due; sent += 1) {
-            if (!this.#sendNext())
-                break;
-        }
-        if (this.#packets.length === 0) {
-            if (!this.#started) {
-                this.#stopPump();
-                this.#notifyDrained();
-                return;
-            }
-            this.#notifyDrained();
-        }
-        const delay = Math.max(0, Math.ceil(this.#pacer.nextDueAt() - performance.now()));
-        this.#pump = setTimeout(() => this.#tick(), delay);
-        this.#pump.unref?.();
-    }
-    #stopPump() {
-        this.#pacer.reset();
-        if (!this.#pump)
-            return;
-        clearTimeout(this.#pump);
-        this.#pump = undefined;
-    }
-    /** Writes one packet, queued audio first, else silence once started; false when nothing was written. */
-    #sendNext() {
-        if (this.#stopped)
-            return false;
-        const application = this.#packets.shift();
-        if (!application && !this.#started)
-            return false;
-        const header = new RtpHeader({
-            payloadType: LOCAL_PAYLOAD_TYPE,
-            sequenceNumber: this.#sequenceNumber,
-            timestamp: this.#timestamp,
-            ssrc: 0,
-            marker: this.#first,
-        });
-        this.#first = false;
-        this.#sequenceNumber = (this.#sequenceNumber + 1) & 0xffff;
-        this.#timestamp = (this.#timestamp + PACKET_FRAMES) >>> 0;
-        this.#track.writeRtp(new RtpPacket(header, application ?? this.#silence));
-        if (application)
-            this.#applicationRtpPackets += 1;
-        else
-            this.#silencePackets += 1;
-        this.#rtp.mark();
-        return true;
-    }
-}
-/** Opus RTP in, PCM16 in the requested format (default 48 kHz stereo) out, one `ondata` per packet. */
-class WeriftAudioSink {
-    ondata = null;
-    #decoder;
-    #rtp = new PacketClock();
-    #decodeFailures = 0;
-    #unsubscribe;
-    stats() {
-        return {
-            rtpPackets: this.#rtp.count,
-            decodeFailures: this.#decodeFailures,
-            firstRtpAt: this.#rtp.firstAt,
-            lastRtpAt: this.#rtp.lastAt,
-            recentRtpPackets: this.#rtp.recent(),
-        };
-    }
-    constructor(track, format = { sampleRate: WERIFT_SAMPLE_RATE, channelCount: WERIFT_CHANNEL_COUNT }) {
-        const { sampleRate, channelCount } = format;
-        this.#decoder = new Decoder({ channels: channelCount, sample_rate: sampleRate });
-        const { unSubscribe } = track.onReceiveRtp.subscribe((rtp) => {
-            this.#rtp.mark();
-            const handler = this.ondata;
-            if (!handler || !rtp.payload?.length)
-                return;
-            const packet = OpusRtpPayload.deSerialize(rtp.payload);
-            let bytes;
-            try {
-                bytes = this.#decoder.decode(packet.payload);
-            }
-            catch {
-                this.#decodeFailures += 1;
-                return;
-            }
-            const samples = new Int16Array(bytes.byteLength >> 1);
-            new Uint8Array(samples.buffer).set(bytes.subarray(0, samples.byteLength));
-            handler({
-                samples,
-                sampleRate,
-                bitsPerSample: 16,
-                channelCount,
-                numberOfFrames: samples.length / channelCount,
-            });
-        });
-        this.#unsubscribe = unSubscribe;
-    }
-    stop() {
-        this.ondata = null;
-        this.#unsubscribe();
-    }
-}
-/**
- * Video codecs. werift matches a remote codec by MIME type alone
- * (`findCodecByMimeType` in `TransceiverManager.setRemoteRTP`) and throws
- * "negotiate codecs failed." for a media section with no local codec of its
- * kind, so every codec a phone may publish is listed. H.264 and VP8 are the
- * ones this engine encodes and decodes (engine-werift-video.ts): H.264 as
- * constrained baseline `42e01f`, the only H.264 profile Cloudflare's SFU
- * accepts, with packetization-mode 1; both ask for NACK, PLI and FIR, without
- * which the SFU sends no keyframe request. VP9 and AV1 are receive-only: a
- * pull offer carrying them is answered, the track is not decoded.
- */
-const videoCodecs = () => {
-    const feedback = () => [useNACK(), usePLI(), useFIR()];
-    return [
-        new RTCRtpCodecParameters({
-            mimeType: "video/H264",
-            clockRate: 90_000,
-            rtcpFeedback: feedback(),
-            parameters: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
-        }),
-        new RTCRtpCodecParameters({ mimeType: "video/VP8", clockRate: 90_000, rtcpFeedback: feedback() }),
-        new RTCRtpCodecParameters({ mimeType: "video/VP9", clockRate: 90_000, direction: "recvonly" }),
-        new RTCRtpCodecParameters({ mimeType: "video/AV1", clockRate: 90_000, direction: "recvonly" }),
-    ];
-};
-/**
- * The 3GPP coordination-of-video-orientation extension: with it negotiated a
- * phone sends its camera frames unrotated and says how to turn them, and the
- * receiver reports that as `VideoFrameEvent.rotation`, as LiveKit does.
- */
-const VIDEO_ORIENTATION_URI = "urn:3gpp:video-orientation";
-export const createWeriftPeerConnection = (config = { iceServers: [], iceTransportPolicy: "all" }) => new RTCPeerConnection({
-    bundlePolicy: "max-bundle",
-    iceServers: config.iceServers,
-    iceTransportPolicy: config.iceTransportPolicy,
-    codecs: {
-        audio: [
-            new RTCRtpCodecParameters({
-                mimeType: "audio/opus",
-                clockRate: WERIFT_SAMPLE_RATE,
-                channels: WERIFT_CHANNEL_COUNT,
-            }),
-        ],
-        video: videoCodecs(),
-    },
-    headerExtensions: {
-        video: [new RTCRtpHeaderExtensionParameters({ id: 1, uri: VIDEO_ORIENTATION_URI })],
-    },
-});
-export const createWeriftWebRTCFactory = () => ({
-    createPeerConnection: (config) => createWeriftPeerConnection(config),
-    createAudioSource: () => new WeriftAudioSource(),
-    createAudioSink: (track, format) => new WeriftAudioSink(track, format),
-    ...createWeriftVideoFactory(),
-});
-//# sourceMappingURL=engine-werift.js.map

package/dist/engine-werift.js.map

@@ -1 +0,0 @@
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