@relaymessenger/livekit 0.1.0-staging.2 → 0.1.0-staging.21

package/dist/engine-werift.js

@@ -1,263 +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.
- * 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, RtpHeader, RtpPacket, } from "werift";
-/** 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);
-    }
-}
-/** 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; a timer drains one packet every 20 ms so the wire sees a steady
- * cadence regardless of how the adapter chunks its writes.
- */
-class WeriftAudioSource {
-    #encoder = new Encoder({
-        channels: WERIFT_CHANNEL_COUNT,
-        sample_rate: WERIFT_SAMPLE_RATE,
-        application: "voip",
-    });
-    #track = new MediaStreamTrack({ kind: "audio", id: "microphone", streamId: "relay-call" });
-    #packets = [];
-    #rtp = new PacketClock();
-    #opusPackets = 0;
-    #pending = new Int16Array(0);
-    #sequenceNumber = 1;
-    #timestamp = 0;
-    #first = true;
-    #pump;
-    #stopped = false;
-    stats() {
-        return {
-            opusPackets: this.#opusPackets,
-            rtpPackets: this.#rtp.count,
-            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.#packets.length = 0;
-            this.#pending = new Int16Array(0);
-            this.#stopPump();
-            stop();
-        };
-        return track;
-    }
-    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();
-    }
-    #startPump() {
-        if (this.#pump || this.#packets.length === 0)
-            return;
-        this.#sendNext();
-        if (this.#packets.length === 0)
-            return;
-        this.#pump = setInterval(() => {
-            this.#sendNext();
-            if (this.#packets.length === 0)
-                this.#stopPump();
-        }, WERIFT_PACKET_MS);
-        this.#pump.unref?.();
-    }
-    #stopPump() {
-        if (!this.#pump)
-            return;
-        clearInterval(this.#pump);
-        this.#pump = undefined;
-    }
-    #sendNext() {
-        const payload = this.#packets.shift();
-        if (!payload || this.#stopped)
-            return;
-        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, payload));
-        this.#rtp.mark();
-    }
-}
-/** Opus RTP in, PCM16 (48 kHz stereo) out, one `ondata` per packet. */
-class WeriftAudioSink {
-    ondata = null;
-    #decoder = new Decoder({ channels: WERIFT_CHANNEL_COUNT, sample_rate: WERIFT_SAMPLE_RATE });
-    #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) {
-        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: WERIFT_SAMPLE_RATE,
-                bitsPerSample: 16,
-                channelCount: WERIFT_CHANNEL_COUNT,
-                numberOfFrames: samples.length / WERIFT_CHANNEL_COUNT,
-            });
-        });
-        this.#unsubscribe = unSubscribe;
-    }
-    stop() {
-        this.ondata = null;
-        this.#unsubscribe();
-    }
-}
-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,
-            }),
-        ],
-    },
-});
-export const createWeriftWebRTCFactory = () => ({
-    createPeerConnection: (config) => createWeriftPeerConnection(config),
-    createAudioSource: () => new WeriftAudioSource(),
-    createAudioSink: (track) => new WeriftAudioSink(track),
-});
-//# sourceMappingURL=engine-werift.js.map

package/dist/engine-werift.js.map

@@ -1 +0,0 @@
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package/dist/transport.d.ts

@@ -1,159 +0,0 @@
-import type { CallRoomEndedFrame, CallRoomOptions, CallRoomStateFrame, Relay } from "@relaymessenger/sdk";
-export interface RelayAudioFrame {
-    /** Interleaved signed PCM16 samples. */
-    samples: Int16Array;
-    sampleRate: number;
-    channelCount: number;
-}
-/** Standard `RTCIceServer` shape: STUN or TURN URLs with optional credentials. */
-export interface RelayIceServer {
-    urls: string | string[];
-    username?: string;
-    credential?: string;
-}
-export type RelayIceTransportPolicy = "all" | "relay";
-export type RelayCallEngine = "werift" | "wrtc";
-export interface RelayCallTransportOptions {
-    relay: Relay;
-    callId: string;
-    room?: CallRoomOptions;
-    /**
-     * WebRTC engine. `"werift"` (default) is pure TypeScript plus prebuilt Opus
-     * and loads no native binding. `"wrtc"` loads `@roamhq/wrtc` on demand.
-     */
-    engine?: RelayCallEngine;
-    /**
-     * STUN and TURN servers handed to the engine's `RTCPeerConnection`. Defaults
-     * to none: Cloudflare's SFU answers with its own host candidates. Set TURN
-     * servers when the agent runs behind a NAT or firewall that blocks UDP.
-     */
-    iceServers?: RelayIceServer[];
-    /** `"relay"` forces every candidate through TURN. Defaults to `"all"`. */
-    iceTransportPolicy?: RelayIceTransportPolicy;
-    /** Maximum wait for the WebRTC peer to become connected. Defaults to 15 seconds. */
-    mediaConnectTimeoutMs?: number;
-    /** @deprecated Use `mediaConnectTimeoutMs`. */
-    connectionTimeoutMs?: number;
-    /**
-     * Called once per call when outbound audio is queued but no RTP packet has
-     * been written for 2 s while media is connected. Receives the diagnostics
-     * summary. Defaults to a no-op; nothing is restarted.
-     */
-    onWarning?: (message: string) => void;
-}
-/** Packet counts for the other participant's track, as seen by this participant. */
-export interface RelayCallInboundDiagnostics {
-    /** RTP packets received on the subscribed track. */
-    rtpPackets: number;
-    /** Opus packets the decoder rejected. */
-    decodeFailures: number;
-    /** PCM frames handed to the `audio` listeners. */
-    frames: number;
-    /** ms since `connect()` for the first and last RTP packet; undefined until one arrives. */
-    firstPacketAtMs: number | undefined;
-    lastPacketAtMs: number | undefined;
-    /** RTP packets received in the 5 s before `diagnostics()` was called. */
-    recentRtpPackets: number;
-}
-/** Packet counts for this participant's published track. */
-export interface RelayCallOutboundDiagnostics {
-    /** PCM slices accepted from the caller and handed to the engine source. */
-    frames: number;
-    opusPackets: number;
-    rtpPackets: number;
-    firstPacketAtMs: number | undefined;
-    lastPacketAtMs: number | undefined;
-    /** RTP packets written in the 5 s before `diagnostics()` was called. */
-    recentRtpPackets: number;
-    /** Encoded packets waiting for the 20 ms pacer. */
-    queued: number;
-    /** `undefined` when the engine does not expose its pacer (`wrtc`). */
-    pacerAlive: boolean | undefined;
-}
-/** Room signaling frames counted since `connect()`. */
-export interface RelayCallRoomDiagnostics {
-    roomStates: number;
-    /** Subscription (pull) offers received from the room. */
-    offers: number;
-    endedReason: string | undefined;
-    /** `error` frame messages, in order. */
-    errors: string[];
-}
-/** ICE facts recorded for one call, for logs and for the connect timeout error. */
-export interface RelayCallIceDiagnostics {
-    /** Local candidates gathered, counted by type. */
-    local: {
-        host: number;
-        srflx: number;
-        relay: number;
-        other: number;
-    };
-    /** Remote candidates from the SFU answer: transport and port only, never the address. */
-    remote: Array<{
-        transport: string;
-        port: number;
-    }>;
-    /** ICE gathering, ICE connection and peer connection state changes since `connect()`. */
-    transitions: Array<{
-        kind: "gathering" | "ice" | "connection";
-        state: string;
-        atMs: number;
-    }>;
-    connected: boolean;
-    inbound: RelayCallInboundDiagnostics;
-    outbound: RelayCallOutboundDiagnostics;
-    room: RelayCallRoomDiagnostics;
-    /** One-line rendering of the fields above. */
-    summary: string;
-}
-export interface RelayCallTransportCloseEvent {
-    code: number;
-    reason: string;
-    wasClean: boolean;
-}
-type TransportEventMap = {
-    audio: [RelayAudioFrame];
-    connected: [];
-    roomState: [CallRoomStateFrame];
-    ended: [CallRoomEndedFrame];
-    error: [Error];
-    close: [RelayCallTransportCloseEvent];
-};
-type TransportEvent = keyof TransportEventMap;
-type TransportListener<K extends TransportEvent> = (...args: TransportEventMap[K]) => void;
-export declare class RelayCallTransportError extends Error {
-    readonly code?: string;
-    constructor(message: string, code?: string);
-}
-/**
- * Provider-neutral Node WebRTC bridge for Relay Call rooms.
- *
- * The class owns Relay media negotiation internally. Higher-level adapters only
- * exchange PCM16 frames and call lifecycle events.
- */
-export declare class RelayCallTransport {
-    #private;
-    constructor(options: RelayCallTransportOptions);
-    on<K extends TransportEvent>(event: K, listener: TransportListener<K>): this;
-    off<K extends TransportEvent>(event: K, listener: TransportListener<K>): this;
-    connect(): Promise<void>;
-    /** Replace only the room signaling socket and replay the identical publication. */
-    reconnect(): Promise<void>;
-    /**
-     * ICE candidates, state transitions, packet counts in both directions and
-     * room frame counts recorded for this call, with a one-line summary.
-     */
-    diagnostics(): RelayCallIceDiagnostics;
-    /**
-     * Feed interleaved PCM16 audio into Relay. Frames are split into 10 ms WebRTC
-     * source slices and paced in real time, so adapters may push larger chunks.
-     */
-    writeAudio(frame: RelayAudioFrame): Promise<void>;
-    /** Drop queued outgoing PCM. At most one already-submitted 10 ms slice remains. */
-    clearAudio(): void;
-    setMuted(muted: boolean): void;
-    end(): void;
-    close(): void;
-}
-export {};
-//# sourceMappingURL=transport.d.ts.map

package/dist/transport.d.ts.map

@@ -1 +0,0 @@
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