@relaymessenger/livekit 0.1.0-staging.1 → 0.1.0-staging.11

package/README.md

@@ -39,10 +39,21 @@ async function answerCall(callId: string, session: AgentSession) {
 ```
 
 `RelayLiveKitCall.connect()` does not resolve until the WebRTC media peer has
-reached `connected`. `RelayAudioInput` converts the remote participant's PCM16
-audio to LiveKit frames. `RelayAudioOutput` publishes LiveKit TTS frames to
-Relay, pacing larger frames into 10 ms WebRTC source slices. `clearBuffer()`
-drops queued outbound audio that has not yet reached WebRTC.
+reached `connected`. `RelayAudioInput` hands the AgentSession the remote
+participant's audio as 24 kHz mono PCM16 frames, the format of LiveKit's own
+room input; the Opus decoder produces that format directly.
+
+`RelayAudioOutput` has the shape of LiveKit's own `ParticipantAudioOutput`:
+`captureFrame()` hands the frame to the transport and returns at once, so the
+AgentSession may push a whole reply faster than real time; the engine's 20 ms
+pump paces the wire. `flush()` closes the segment and reports
+`playbackFinished` only after the transport has drained. `clearBuffer()` drops
+audio that has not reached the wire and reports the segment as interrupted at
+the position that actually played.
+
+On the transport, `writeAudio()` resolves once its 10 ms slices are queued,
+`queuedAudioMs()` is what has not left yet, and `waitForPlayout()` resolves when
+the queue is empty and the pump is idle (early on `clearAudio()`).
 
 Use `setMuted(true)` to publish participant mute state, `end()` to end the Relay
 Call, and `close()` for local cleanup. If the signaling connection is replaced,
@@ -63,6 +74,9 @@ const transport = new RelayCallTransport({ room });
 transport.on("audio", ({ samples, sampleRate, channelCount }) => {
   // Interleaved signed PCM16 from the remote Relay participant.
 });
+// Inbound audio is 48 kHz stereo unless you pass
+// `inboundAudio: { sampleRate: 8000 | 12000 | 16000 | 24000 | 48000, channelCount: 1 | 2 }`;
+// the `"wrtc"` engine accepts only the default.
 
 await transport.connect();
 await transport.writeAudio({
@@ -70,12 +84,17 @@ await transport.writeAudio({
   sampleRate: 48_000,
   channelCount: 1,
 });
+await transport.waitForPlayout();
 ```
 
 ## ICE servers, TURN, and diagnostics
 
-By default the peer gathers host candidates only and Cloudflare's SFU supplies
-its own candidates in the answer. Inside a container or behind a firewall that
+By default the peer uses Cloudflare's STUN server
+(`stun:stun.cloudflare.com:3478`), as Cloudflare's own Realtime echo example
+does, and Cloudflare's SFU supplies its own candidates in the answer. The offer
+leaves as soon as the first local candidate exists, without waiting for ICE
+gathering to finish: the SFU is ICE-lite and learns the agent's address from
+its connectivity checks. Inside a container or behind a firewall that
 blocks outbound UDP, pass TURN servers in the standard `RTCIceServer` shape and,
 if every path must go through TURN, `iceTransportPolicy: "relay"`. Both options
 are accepted by `RelayLiveKitCall.connect()` and `RelayCallTransport`.
@@ -96,19 +115,59 @@ const call = await RelayLiveKitCall.connect({
     },
   ],
   iceTransportPolicy: "all",
-  mediaConnectTimeoutMs: 15_000,
 });
 ```
 
-`mediaConnectTimeoutMs` bounds the wait for the media peer to reach
-`connected` (15 seconds by default). When that wait expires, the error message
-carries what ICE saw, for example
-`Timed out connecting Relay WebRTC media (local: host 2, srflx 0, relay 0; remote: udp 1473; states: new→complete 0.2s, connecting 0.3s, no connected)`.
+`iceServers` may also be a function; it is called before every peer
+connection, so it can mint fresh TURN credentials for each restart.
+
+`connect()` resolves when media first reaches `connected`. It has no overall
+deadline: when an SFU session is not `connected` within
+`sessionConnectTimeoutMs` (5 seconds by default) of its answer, becomes
+`failed`, or stays `disconnected` for 7 seconds, the transport closes that
+peer, waits 250 ms (x1.1 per further attempt, at most 10 s), and publishes
+from a new peer on a new session, for as long as the Call is ringing or in
+progress. Outgoing audio keeps flowing into the new peer. `connect()` rejects
+only when the Call ends, the room or transport closes, the room reports an
+error, or the `signal` passed to it aborts. Each replacement emits
+`restarted` with the reason and the replaced session's summary, for example
+`local: host 2, srflx 0, relay 0; remote: udp 1473; states: new→complete 0.2s, connecting 0.3s, no connected; …`,
+and `diagnostics().restarts` counts them. `mediaConnectTimeoutMs` and
+`connectionTimeoutMs` are deprecated names for `sessionConnectTimeoutMs`.
+
+`call.waitForPeerAudio(timeoutMs)` resolves once the person's audio has
+arrived and the room shows them connected (the transport's `peerAudio` event);
+start the agent's session after it so the first words are heard.
+
 The same facts are available at any time from `call.diagnostics()` or
 `transport.diagnostics()`: local candidate counts by type, the remote
 candidates' transport and port (never their address), the ICE gathering, ICE
 connection and peer connection state changes with their offsets from
-`connect()`, and the one-line `summary`.
+`connect()`, packet counts in both directions (`inbound`: RTP received, Opus
+decode failures, PCM frames delivered, first and last packet offsets, packets
+in the last 5 s; `outbound`: PCM frames accepted, Opus packets, RTP written
+with the caller's audio, RTP written with silence, first and last packet
+offsets, packets in the last 5 s, paced queue size, pacer state), the room frames seen (`roomState` count, pull `offer` count,
+`ended` reason, `error` messages), and the one-line `summary`, for example
+`…; in: 1234 rtp, 0 bad, 1234 frames, first 0.9s last 41.2s, 250/5s; out: 2600 frames, 1300 opus, 1300 rtp, silence 700, first 1.1s last 41.0s, 250/5s, queue 0, pacer alive; room: 3 roomState, 1 offer`.
+Packet counts come from the `werift` engine; `wrtc` reports zero packets and
+`pacer n/a`.
+
+Like a live microphone, the `werift` engine's published track sends one Opus
+packet every 20 ms from the moment media connects until the transport closes,
+paced by the monotonic clock: a timer that fires late sends every packet due
+by then, so the wire carries exactly 50 packets a second; a stall longer than
+200 ms restarts the clock instead of bursting. The track carries
+the caller's audio when some is queued, Opus silence otherwise. Cloudflare's
+SFU will not let the person's side pull a track that has carried no RTP, so a
+silent agent track would never be heard. Silence never counts toward
+`queuedAudioMs()` or `waitForPlayout()`. The `wrtc` engine sends only the
+audio written to it; write silence yourself if you use it.
+
+`onWarning` is called once per call, with the summary, when outbound audio is
+queued but no RTP packet has been written for 2 s while media is connected.
+Nothing is restarted; the callback exists so the failing direction is named in
+the agent's logs.
 
 `RelayCallTransport` consumes the SDK's `CallRoom`; it does not duplicate the
 room protocol. The default engine is `werift` (pure TypeScript WebRTC) with

package/dist/engine-werift.d.ts

@@ -6,6 +6,37 @@ export declare const WERIFT_CHANNEL_COUNT = 2;
 /** One RTP packet carries 20 ms: 960 frames at 48 kHz, timestamp advances by 960. */
 export declare const WERIFT_PACKET_MS = 20;
 export type WeriftAudioSourceData = Parameters<RelayAudioSourceLike["onData"]>[0];
+/** Packets counted for `diagnostics()`; the recent window is the last 5 s. */
+export declare const STATS_WINDOW_MS = 5000;
+/**
+ * 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 declare 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 declare class RtpAudioPacer {
+    #private;
+    readonly packetMs: number;
+    readonly maxCatchUpMs: number;
+    constructor(packetMs?: number, maxCatchUpMs?: number);
+    /** Packets due by `now` and not yet sent; the first call of a run sends one at once. */
+    take(now: number): number;
+    /** When the next packet is due; `-Infinity` before the first `take` of a run. */
+    nextDueAt(): number;
+    /** Forget the run; the next `take` starts a new clock. */
+    reset(): void;
+}
 /** Convert any PCM16 frame to interleaved 48 kHz stereo by linear interpolation. */
 export declare const toWireFormat: (data: {
     samples: Int16Array;

package/dist/engine-werift.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"engine-werift.d.ts","sourceRoot":"","sources":["../src/engine-werift.ts"],"names":[],"mappings":"AAqCA,OAAO,EAGL,iBAAiB,EAIlB,MAAM,QAAQ,CAAC;AAChB,OAAO,KAAK,EAGV,oBAAoB,EAIpB,yBAAyB,EAEzB,kBAAkB,EACnB,MAAM,gBAAgB,CAAC;AAExB,sGAAsG;AACtG,eAAO,MAAM,kBAAkB,QAAS,CAAC;AACzC,eAAO,MAAM,oBAAoB,IAAI,CAAC;AACtC,qFAAqF;AACrF,eAAO,MAAM,gBAAgB,KAAK,CAAC;AAMnC,MAAM,MAAM,qBAAqB,GAAG,UAAU,CAAC,oBAAoB,CAAC,QAAQ,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;AAElF,8EAA8E;AAC9E,eAAO,MAAM,eAAe,OAAQ,CAAC;AA8BrC;;;;;;;;;;GAUG;AACH,eAAO,MAAM,sBAAsB,MAAM,CAAC;AAE1C;;;;GAIG;AACH,qBAAa,aAAa;;IAKtB,QAAQ,CAAC,QAAQ;IACjB,QAAQ,CAAC,YAAY;gBADZ,QAAQ,SAAmB,EAC3B,YAAY,SAAyB;IAGhD,wFAAwF;IACxF,IAAI,CAAC,GAAG,EAAE,MAAM,GAAG,MAAM;IAWzB,iFAAiF;IACjF,SAAS,IAAI,MAAM;IAInB,0DAA0D;IAC1D,KAAK,IAAI,IAAI;CAId;AAED,oFAAoF;AACpF,eAAO,MAAM,YAAY,GAAI,MAAM;IACjC,OAAO,EAAE,UAAU,CAAC;IACpB,UAAU,EAAE,MAAM,CAAC;IACnB,YAAY,EAAE,MAAM,CAAC;CACtB,KAAG,UAwBH,CAAC;AA4QF,eAAO,MAAM,0BAA0B,GACrC,SAAQ,yBAAyE,KAChF,iBAeC,CAAC;AAEL,eAAO,MAAM,yBAAyB,QAAO,kBAI3C,CAAC"}

package/dist/engine-werift.js

@@ -21,7 +21,11 @@
  *   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.
+ *   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
@@ -41,6 +45,82 @@ 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;
@@ -69,8 +149,25 @@ export const toWireFormat = (data) => {
 };
 /**
  * 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.
+ * 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({
@@ -78,26 +175,70 @@ class WeriftAudioSource {
         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.#packets.length = 0;
-            this.#pending = new Int16Array(0);
+            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;
@@ -109,34 +250,85 @@ class WeriftAudioSource {
         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.#packets.length === 0)
+        if (this.#pump || this.#stopped)
             return;
-        this.#sendNext();
-        if (this.#packets.length === 0)
+        if (!this.#started && this.#packets.length === 0)
+            return;
+        this.#tick();
+    }
+    #tick() {
+        if (this.#stopped)
             return;
-        this.#pump = setInterval(() => {
-            this.#sendNext();
-            if (this.#packets.length === 0)
+        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();
-        }, WERIFT_PACKET_MS);
+                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;
-        clearInterval(this.#pump);
+        clearTimeout(this.#pump);
         this.#pump = undefined;
     }
+    /** Writes one packet, queued audio first, else silence once started; false when nothing was written. */
     #sendNext() {
-        const payload = this.#packets.shift();
-        if (!payload || this.#stopped)
-            return;
+        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,
@@ -147,16 +339,36 @@ class WeriftAudioSource {
         this.#first = false;
         this.#sequenceNumber = (this.#sequenceNumber + 1) & 0xffff;
         this.#timestamp = (this.#timestamp + PACKET_FRAMES) >>> 0;
-        this.#track.writeRtp(new RtpPacket(header, payload));
+        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 (48 kHz stereo) out, one `ondata` per packet. */
+/** Opus RTP in, PCM16 in the requested format (default 48 kHz stereo) out, one `ondata` per packet. */
 class WeriftAudioSink {
     ondata = null;
-    #decoder = new Decoder({ channels: WERIFT_CHANNEL_COUNT, sample_rate: WERIFT_SAMPLE_RATE });
+    #decoder;
+    #rtp = new PacketClock();
+    #decodeFailures = 0;
     #unsubscribe;
-    constructor(track) {
+    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;
@@ -166,16 +378,17 @@ class WeriftAudioSink {
                 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,
+                sampleRate,
                 bitsPerSample: 16,
-                channelCount: WERIFT_CHANNEL_COUNT,
-                numberOfFrames: samples.length / WERIFT_CHANNEL_COUNT,
+                channelCount,
+                numberOfFrames: samples.length / channelCount,
             });
         });
         this.#unsubscribe = unSubscribe;
@@ -185,6 +398,15 @@ class WeriftAudioSink {
         this.#unsubscribe();
     }
 }
+/**
+ * Video codecs the peer can accept in a pull offer. Nothing decodes video: the
+ * transport only answers the person's `video` m-line receive-only and ignores
+ * the track (PROTOCOL.md section 5). 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 an audio-only codec list cannot answer a video m-line.
+ */
+const RECEIVE_VIDEO_MIME_TYPES = ["video/VP8", "video/VP9", "video/H264", "video/AV1"];
 export const createWeriftPeerConnection = (config = { iceServers: [], iceTransportPolicy: "all" }) => new RTCPeerConnection({
     bundlePolicy: "max-bundle",
     iceServers: config.iceServers,
@@ -197,11 +419,12 @@ export const createWeriftPeerConnection = (config = { iceServers: [], iceTranspo
                 channels: WERIFT_CHANNEL_COUNT,
             }),
         ],
+        video: RECEIVE_VIDEO_MIME_TYPES.map((mimeType) => new RTCRtpCodecParameters({ mimeType, clockRate: 90_000 })),
     },
 });
 export const createWeriftWebRTCFactory = () => ({
     createPeerConnection: (config) => createWeriftPeerConnection(config),
     createAudioSource: () => new WeriftAudioSource(),
-    createAudioSink: (track) => new WeriftAudioSink(track),
+    createAudioSink: (track, format) => new WeriftAudioSink(track, format),
 });
 //# sourceMappingURL=engine-werift.js.map

package/dist/engine-werift.js.map

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package/dist/index.d.ts

@@ -1,4 +1,4 @@
-export { RelayAudioInput, RelayAudioOutput, RelayLiveKitCall, createRelayLiveKitAudio, type RelayLiveKitAudio, type RelayLiveKitAudioOptions, type RelayLiveKitConnectOptions, } from "./livekit.js";
-export { RelayCallTransport, RelayCallTransportError, type RelayAudioFrame, type RelayCallEngine, type RelayCallIceDiagnostics, type RelayCallTransportCloseEvent, type RelayCallTransportOptions, type RelayIceServer, type RelayIceTransportPolicy, } from "./transport.js";
+export { RelayAudioInput, RelayAudioOutput, LIVEKIT_ROOM_INPUT_AUDIO, RelayLiveKitCall, createRelayLiveKitAudio, type RelayLiveKitAudio, type RelayLiveKitAudioOptions, type RelayLiveKitConnectOptions, } from "./livekit.js";
+export { RelayCallTransport, RelayCallTransportError, type RelayAudioFrame, type RelayCallEngine, type RelayCallIceDiagnostics, type RelayCallInboundDiagnostics, type RelayCallOutboundDiagnostics, type RelayCallRestartEvent, type RelayCallRestartReason, type RelayCallRoomDiagnostics, type RelayCallTransportCloseEvent, type RelayCallTransportOptions, type RelayIceServer, type RelayIceServersProvider, type RelayIceTransportPolicy, type RelayInboundAudioFormat, } from "./transport.js";
 export { createWeriftWebRTCFactory } from "./engine-werift.js";
 //# sourceMappingURL=index.d.ts.map

package/dist/index.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"index.d.ts","sourceRoot":"","sources":["../src/index.ts"],"names":[],"mappings":"AAAA,OAAO,EACL,eAAe,EACf,gBAAgB,EAChB,gBAAgB,EAChB,uBAAuB,EACvB,KAAK,iBAAiB,EACtB,KAAK,wBAAwB,EAC7B,KAAK,0BAA0B,GAChC,MAAM,cAAc,CAAC;AACtB,OAAO,EACL,kBAAkB,EAClB,uBAAuB,EACvB,KAAK,eAAe,EACpB,KAAK,eAAe,EACpB,KAAK,uBAAuB,EAC5B,KAAK,4BAA4B,EACjC,KAAK,yBAAyB,EAC9B,KAAK,cAAc,EACnB,KAAK,uBAAuB,GAC7B,MAAM,gBAAgB,CAAC;AACxB,OAAO,EAAE,yBAAyB,EAAE,MAAM,oBAAoB,CAAC"}
+{"version":3,"file":"index.d.ts","sourceRoot":"","sources":["../src/index.ts"],"names":[],"mappings":"AAAA,OAAO,EACL,eAAe,EACf,gBAAgB,EAChB,wBAAwB,EACxB,gBAAgB,EAChB,uBAAuB,EACvB,KAAK,iBAAiB,EACtB,KAAK,wBAAwB,EAC7B,KAAK,0BAA0B,GAChC,MAAM,cAAc,CAAC;AACtB,OAAO,EACL,kBAAkB,EAClB,uBAAuB,EACvB,KAAK,eAAe,EACpB,KAAK,eAAe,EACpB,KAAK,uBAAuB,EAC5B,KAAK,2BAA2B,EAChC,KAAK,4BAA4B,EACjC,KAAK,qBAAqB,EAC1B,KAAK,sBAAsB,EAC3B,KAAK,wBAAwB,EAC7B,KAAK,4BAA4B,EACjC,KAAK,yBAAyB,EAC9B,KAAK,cAAc,EACnB,KAAK,uBAAuB,EAC5B,KAAK,uBAAuB,EAC5B,KAAK,uBAAuB,GAC7B,MAAM,gBAAgB,CAAC;AACxB,OAAO,EAAE,yBAAyB,EAAE,MAAM,oBAAoB,CAAC"}

package/dist/index.js

@@ -1,4 +1,4 @@
-export { RelayAudioInput, RelayAudioOutput, RelayLiveKitCall, createRelayLiveKitAudio, } from "./livekit.js";
+export { RelayAudioInput, RelayAudioOutput, LIVEKIT_ROOM_INPUT_AUDIO, RelayLiveKitCall, createRelayLiveKitAudio, } from "./livekit.js";
 export { RelayCallTransport, RelayCallTransportError, } from "./transport.js";
 export { createWeriftWebRTCFactory } from "./engine-werift.js";
 //# sourceMappingURL=index.js.map

package/dist/index.js.map

@@ -1 +1 @@
-{"version":3,"file":"index.js","sourceRoot":"","sources":["../src/index.ts"],"names":[],"mappings":"AAAA,OAAO,EACL,eAAe,EACf,gBAAgB,EAChB,gBAAgB,EAChB,uBAAuB,GAIxB,MAAM,cAAc,CAAC;AACtB,OAAO,EACL,kBAAkB,EAClB,uBAAuB,GAQxB,MAAM,gBAAgB,CAAC;AACxB,OAAO,EAAE,yBAAyB,EAAE,MAAM,oBAAoB,CAAC"}
+{"version":3,"file":"index.js","sourceRoot":"","sources":["../src/index.ts"],"names":[],"mappings":"AAAA,OAAO,EACL,eAAe,EACf,gBAAgB,EAChB,wBAAwB,EACxB,gBAAgB,EAChB,uBAAuB,GAIxB,MAAM,cAAc,CAAC;AACtB,OAAO,EACL,kBAAkB,EAClB,uBAAuB,GAexB,MAAM,gBAAgB,CAAC;AACxB,OAAO,EAAE,yBAAyB,EAAE,MAAM,oBAAoB,CAAC"}