@whereby.com/assistant-sdk 0.0.0-canary-20250912102624 → 0.0.0-canary-20250912144626

package/dist/index.cjs

@@ -88,9 +88,14 @@ class AudioSink extends wrtc.nonstandard.RTCAudioSink {
     }
 }
 
+// Number of pipes in the ffmpeg process. We predefine a fixed number of slots, and then we dynamically assign
+// participants to these slots based on mute/unmute state.
 const PARTICIPANT_SLOTS = 20;
+// Each sample is 2 bytes (16 bits) for PCM audio - s16le format
+// 48000 Hz is the standard sample rate for WebRTC audio
 const STREAM_INPUT_SAMPLE_RATE_IN_HZ = 48000;
 const BYTES_PER_SAMPLE = 2;
+// 480 samples per 10ms frame at 48kHz
 const FRAME_10MS_SAMPLES = 480;
 const slotBuffers = new Map();
 function appendAndDrainTo480(slot, newSamples) {
@@ -102,10 +107,10 @@ function appendAndDrainTo480(slot, newSamples) {
     let offset = 0;
     while (merged.length - offset >= FRAME_10MS_SAMPLES) {
         const chunk = merged.subarray(offset, offset + FRAME_10MS_SAMPLES);
-        enqueueFrame(slot, chunk);
+        enqueueFrame(slot, chunk); // always 480
         offset += FRAME_10MS_SAMPLES;
     }
-    slotBuffers.set(slot, merged.subarray(offset));
+    slotBuffers.set(slot, merged.subarray(offset)); // keep remainder
 }
 ({
     enqFrames: new Array(PARTICIPANT_SLOTS).fill(0),
@@ -117,6 +122,10 @@ function appendAndDrainTo480(slot, newSamples) {
 let slots = [];
 let stopPacerFn = null;
 let outputPacerState = null;
+/**
+ * Simple linear interpolation resampler to convert audio to 48kHz.
+ * This handles the common case of 16kHz -> 48kHz (3x upsampling).
+ */
 function resampleTo48kHz(inputSamples, inputSampleRate, inputFrames) {
     const ratio = STREAM_INPUT_SAMPLE_RATE_IN_HZ / inputSampleRate;
     const outputLength = Math.floor(inputFrames * ratio);
@@ -136,11 +145,43 @@ function resampleTo48kHz(inputSamples, inputSampleRate, inputFrames) {
     }
     return output;
 }
+/**
+ * Enqueue an audio frame for paced delivery to the RTCAudioSource.
+ */
 function enqueueOutputFrame(samples) {
     if (outputPacerState) {
         outputPacerState.frameQueue.push(samples);
     }
 }
+/**
+ * Start the audio pacer loop for all input slots in an FFmpeg process.
+ *
+ * The pacer ensures each slot (pipe:3..3+N-1) is written to at a steady
+ * real-time rate (e.g. 10 ms = 480 samples @ 48kHz), even if WebRTC frames
+ * arrive jittery, bursty, or with slightly different clocks.
+ *
+ * Key behavior:
+ * - Writes exactly one frame per period, on a shared wall-clock grid.
+ * - Uses silence (zero-filled frame) if a slot's queue is empty, so timing
+ *   never stalls.
+ * - Resnaps the schedule if a slot switches between 10 ms / 20 ms frames.
+ * - Honors Node stream backpressure (`write()` return false) without breaking
+ *   the timing grid.
+ *
+ * This keeps all FFmpeg inputs phase-aligned and stable, so aresample/amix
+ * can mix them without slow-downs or drift.
+ *
+ * Call this once right after spawning FFmpeg:
+ * ```ts
+ * const ff = spawnFFmpegProcess();
+ * startPacer(ff, PARTICIPANT_SLOTS);
+ * ```
+ *
+ * When tearing down the mixer, always call `stopPacer()` before killing FFmpeg.
+ *
+ * @param ff        Child process handle from spawn("ffmpeg", ...)
+ * @param slotCount Number of participant input slots (0..N-1 → fd 3..3+N-1)
+ */
 function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     if (stopPacerFn) {
         stopPacerFn();
@@ -148,11 +189,11 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     }
     const writers = Array.from({ length: slotCount }, (_, i) => ff.stdio[3 + i]);
     const nowMs = () => Number(process.hrtime.bigint()) / 1e6;
-    const outputFrameMs = (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000;
+    const outputFrameMs = (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000; // 10ms
     const t0 = nowMs();
     slots = Array.from({ length: slotCount }, () => ({
         q: [],
-        lastFrames: FRAME_10MS_SAMPLES,
+        lastFrames: FRAME_10MS_SAMPLES, // keep constant
         nextDueMs: t0 + (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000,
     }));
     outputPacerState = {
@@ -167,10 +208,11 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
         for (let s = 0; s < slotCount; s++) {
             const st = slots[s];
             const w = writers[s];
-            const frameMs = (st.lastFrames / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000;
+            const frameMs = (st.lastFrames / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000; // 10ms if 480, 20ms if 960
             if (t >= st.nextDueMs) {
                 const buf = st.q.length ? st.q.shift() : Buffer.alloc(st.lastFrames * BYTES_PER_SAMPLE);
                 if (!w.write(buf)) {
+                    // Just continue without adding drain listener - backpressure will naturally resolve
                     const late = t - st.nextDueMs;
                     const steps = Math.max(1, Math.ceil(late / frameMs));
                     st.nextDueMs += steps * frameMs;
@@ -183,9 +225,10 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
         }
         if (!outputPacerState)
             return;
+        // Handle output pacer for RTCAudioSource
         const state = outputPacerState;
         if (t >= state.nextDueMs) {
-            const samples = state.frameQueue.length > 0 ? state.frameQueue.shift() : new Int16Array(FRAME_10MS_SAMPLES);
+            const samples = state.frameQueue.length > 0 ? state.frameQueue.shift() : new Int16Array(FRAME_10MS_SAMPLES); // silence
             if (!state.didEmitReadyEvent) {
                 state.onAudioStreamReady();
                 state.didEmitReadyEvent = true;
@@ -201,12 +244,20 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     }, 5);
     stopPacerFn = () => clearInterval(iv);
 }
+/**
+ * Stop the audio pacer loop and clear all input slots.
+ * Call this before killing the FFmpeg process to ensure clean shutdown.
+ */
 function stopPacer() {
     if (stopPacerFn)
         stopPacerFn();
     stopPacerFn = null;
     slots = [];
 }
+/**
+ * Queue a live frame for a given slot (0..N-1).
+ * Auto-resnaps the slot's schedule if the frame size (480/960) changes.
+ */
 function enqueueFrame(slot, samples, numberOfFrames) {
     const st = slots[slot];
     if (!st)
@@ -214,6 +265,10 @@ function enqueueFrame(slot, samples, numberOfFrames) {
     const buf = Buffer.from(samples.buffer, samples.byteOffset, samples.byteLength);
     st.q.push(buf);
 }
+/**
+ * Clear the audio queue for a specific slot when a participant leaves.
+ * This prevents stale audio data from continuing to play after disconnect.
+ */
 function clearSlotQueue(slot) {
     const st = slots[slot];
     if (st) {
@@ -223,6 +278,11 @@ function clearSlotQueue(slot) {
         st.nextDueMs = now + frameMs;
     }
 }
+/**
+ * Get the FFmpeg arguments for mixing audio from multiple participants.
+ * This will read from the input pipes (3..3+N-1) and output a single mixed audio stream.
+ * The output is in PCM 16-bit little-endian format at 48kHz sample rate.
+ */
 function getFFmpegArguments() {
     const N = PARTICIPANT_SLOTS;
     const SR = STREAM_INPUT_SAMPLE_RATE_IN_HZ;
@@ -240,6 +300,14 @@ function getFFmpegArguments() {
     ffArgs.push("-hide_banner", "-nostats", "-loglevel", "error", "-filter_complex", filter, "-map", "[mix]", "-f", "s16le", "-ar", String(SR), "-ac", "1", "-c:a", "pcm_s16le", "pipe:1");
     return ffArgs;
 }
+/**
+ * Spawn a new FFmpeg process for mixing audio from multiple participants.
+ * This will read from the input pipes (3..3+N-1) and output a single mixed audio stream.
+ * The output is in PCM 16-bit little-endian format at 48kHz sample rate.
+ * The process will log its output to stderr.
+ * @param rtcAudioSource The RTCAudioSource to which the mixed audio will be sent.
+ * @return The spawned FFmpeg process.
+ */
 function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     const stdio = ["ignore", "pipe", "pipe", ...Array(PARTICIPANT_SLOTS).fill("pipe")];
     const args = getFFmpegArguments();
@@ -249,7 +317,7 @@ function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     ffmpegProcess.stderr.on("data", (d) => console.error("[ffmpeg]", String(d).trim()));
     ffmpegProcess.on("error", () => console.error("FFmpeg process error: is ffmpeg installed?"));
     let audioBuffer = Buffer.alloc(0);
-    const FRAME_SIZE_BYTES = FRAME_10MS_SAMPLES * BYTES_PER_SAMPLE;
+    const FRAME_SIZE_BYTES = FRAME_10MS_SAMPLES * BYTES_PER_SAMPLE; // 480 samples * 2 bytes = 960 bytes
     ffmpegProcess.stdout.on("data", (chunk) => {
         audioBuffer = Buffer.concat([audioBuffer, chunk]);
         while (audioBuffer.length >= FRAME_SIZE_BYTES) {
@@ -264,6 +332,16 @@ function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     });
     return ffmpegProcess;
 }
+/**
+ * Write audio data from a MediaStreamTrack to the FFmpeg process.
+ * This function creates an AudioSink for the track and sets up a data handler
+ * that enqueues audio frames into the pacer.
+ *
+ * @param ffmpegProcess The FFmpeg process to which audio data will be written.
+ * @param slot The participant slot number (0..N-1) to which this track belongs.
+ * @param audioTrack The MediaStreamTrack containing the audio data.
+ * @return An object containing the AudioSink, the writable stream, and a stop function.
+ */
 function writeAudioDataToFFmpeg(ffmpegProcess, slot, audioTrack) {
     const writer = ffmpegProcess.stdio[3 + slot];
     const sink = new AudioSink(audioTrack);
@@ -288,6 +366,12 @@ function writeAudioDataToFFmpeg(ffmpegProcess, slot, audioTrack) {
     };
     return { sink, writer, stop };
 }
+/**
+ * Stop the FFmpeg process and clean up all resources.
+ * This function will unpipe the stdout, end all writable streams for each participant slot,
+ * and kill the FFmpeg process.
+ * @param ffmpegProcess The FFmpeg process to stop.
+ */
 function stopFFmpegProcess(ffmpegProcess) {
     stopPacer();
     if (ffmpegProcess && !ffmpegProcess.killed) {
@@ -341,6 +425,7 @@ class AudioMixer extends EventEmitter.EventEmitter {
         for (const p of participants)
             this.attachParticipantIfNeeded(p);
         const liveIds = new Set(participants.map((p) => p.id).filter(Boolean));
+        // eslint-disable-next-line @typescript-eslint/no-unused-vars
         for (const [slot, pid] of this.participantSlots) {
             if (pid && !liveIds.has(pid))
                 this.detachParticipant(pid);
@@ -353,6 +438,7 @@ class AudioMixer extends EventEmitter.EventEmitter {
         }
         this.participantSlots = new Map(Array.from({ length: PARTICIPANT_SLOTS }, (_, i) => [i, ""]));
         this.activeSlots = {};
+        // Recreate the media stream to avoid stale references
         this.setupMediaStream();
     }
     slotForParticipant(participantId) {
@@ -418,6 +504,7 @@ class AudioMixer extends EventEmitter.EventEmitter {
             }
             this.activeSlots[slot] = undefined;
         }
+        // Clear any queued audio data for this slot to prevent stale audio
         clearSlotQueue(slot);
         this.participantSlots.set(slot, "");
     }
@@ -545,10 +632,7 @@ dotenv__namespace.config();
 const { IS_LOCAL = "false", BIND_INTERFACE = "en0" } = process.env;
 function buildRoomUrl(roomPath, wherebySubdomain, baseDomain = "whereby.com") {
     let wherebyDomain;
-    if (!IS_LOCAL) {
-        wherebyDomain = `${wherebySubdomain}.${baseDomain}`;
-    }
-    else {
+    if (IS_LOCAL === "true") {
         const ifaceAddrs = os.networkInterfaces()[BIND_INTERFACE];
         if (!ifaceAddrs) {
             throw new Error(`Unknown interface ${BIND_INTERFACE}`);
@@ -559,6 +643,9 @@ function buildRoomUrl(roomPath, wherebySubdomain, baseDomain = "whereby.com") {
         }
         wherebyDomain = `${wherebySubdomain}-ip-${bindAddr.address.replace(/[.]/g, "-")}.hereby.dev:4443`;
     }
+    else {
+        wherebyDomain = `${wherebySubdomain}.${baseDomain}`;
+    }
     return `https://${wherebyDomain}${roomPath}`;
 }
 
@@ -599,6 +686,7 @@ class Trigger extends EventEmitter.EventEmitter {
         const router = webhookRouter(this.webhookTriggers, this, this.assistantKey, this.startCombinedAudioStream, this.startLocalMedia);
         app.use(router);
         const server = app.listen(this.port, () => {
+            // console.log(`Bot trigger server now running on port[${this.port}]`);
         });
         process.on("SIGTERM", () => {
             server.close();

package/dist/index.d.cts

@@ -1,9 +1,41 @@
+import * as wrtc from '@roamhq/wrtc';
+import wrtc__default from '@roamhq/wrtc';
 import { RoomConnectionClient, RemoteParticipantState, ChatMessage } from '@whereby.com/core';
 export { RemoteParticipantState } from '@whereby.com/core';
-import wrtc from '@roamhq/wrtc';
 import EventEmitter, { EventEmitter as EventEmitter$1 } from 'events';
 import { PassThrough } from 'stream';
 
+declare global {
+    interface MediaStream extends wrtc.MediaStream {
+    }
+    interface MediaStreamTrack extends wrtc.MediaStreamTrack {
+    }
+    interface RTCDataChannel extends wrtc.RTCDataChannel {
+    }
+    interface RTCDataChannelEvent extends wrtc.RTCDataChannelEvent {
+    }
+    interface RTCDtlsTransport extends wrtc.RTCDtlsTransport {
+    }
+    interface RTCIceCandidate extends wrtc.RTCIceCandidate {
+    }
+    interface RTCIceTransport extends wrtc.RTCIceTransport {
+    }
+    interface RTCPeerConnection extends wrtc.RTCPeerConnection {
+    }
+    interface RTCPeerConnectionIceEvent extends wrtc.RTCPeerConnectionIceEvent {
+    }
+    interface RTCRtpReceiver extends wrtc.RTCRtpReceiver {
+    }
+    interface RTCRtpSender extends wrtc.RTCRtpSender {
+    }
+    interface RTCRtpTransceiver extends wrtc.RTCRtpTransceiver {
+    }
+    interface RTCSctpTransport extends wrtc.RTCSctpTransport {
+    }
+    interface RTCSessionDescription extends wrtc.RTCSessionDescription {
+    }
+}
+
 declare const AUDIO_STREAM_READY = "AUDIO_STREAM_READY";
 type AssistantEvents = {
     [AUDIO_STREAM_READY]: [{
@@ -29,7 +61,7 @@ declare class Assistant extends EventEmitter<AssistantEvents> {
     joinRoom(roomUrl: string): Promise<void>;
     startLocalMedia(): void;
     getLocalMediaStream(): MediaStream | null;
-    getLocalAudioSource(): wrtc.nonstandard.RTCAudioSource | null;
+    getLocalAudioSource(): wrtc__default.nonstandard.RTCAudioSource | null;
     getRoomConnection(): RoomConnectionClient;
     getCombinedAudioStream(): MediaStream | null;
     getRemoteParticipants(): RemoteParticipantState[];
@@ -122,7 +154,7 @@ declare class Trigger extends EventEmitter$1<TriggerEvents> {
 declare class AudioSource extends PassThrough {
     constructor();
 }
-declare class AudioSink extends wrtc.nonstandard.RTCAudioSink {
+declare class AudioSink extends wrtc__default.nonstandard.RTCAudioSink {
     private _sink;
     constructor(track: MediaStreamTrack);
     subscribe(cb: (d: {

package/dist/index.d.mts

@@ -1,9 +1,41 @@
+import * as wrtc from '@roamhq/wrtc';
+import wrtc__default from '@roamhq/wrtc';
 import { RoomConnectionClient, RemoteParticipantState, ChatMessage } from '@whereby.com/core';
 export { RemoteParticipantState } from '@whereby.com/core';
-import wrtc from '@roamhq/wrtc';
 import EventEmitter, { EventEmitter as EventEmitter$1 } from 'events';
 import { PassThrough } from 'stream';
 
+declare global {
+    interface MediaStream extends wrtc.MediaStream {
+    }
+    interface MediaStreamTrack extends wrtc.MediaStreamTrack {
+    }
+    interface RTCDataChannel extends wrtc.RTCDataChannel {
+    }
+    interface RTCDataChannelEvent extends wrtc.RTCDataChannelEvent {
+    }
+    interface RTCDtlsTransport extends wrtc.RTCDtlsTransport {
+    }
+    interface RTCIceCandidate extends wrtc.RTCIceCandidate {
+    }
+    interface RTCIceTransport extends wrtc.RTCIceTransport {
+    }
+    interface RTCPeerConnection extends wrtc.RTCPeerConnection {
+    }
+    interface RTCPeerConnectionIceEvent extends wrtc.RTCPeerConnectionIceEvent {
+    }
+    interface RTCRtpReceiver extends wrtc.RTCRtpReceiver {
+    }
+    interface RTCRtpSender extends wrtc.RTCRtpSender {
+    }
+    interface RTCRtpTransceiver extends wrtc.RTCRtpTransceiver {
+    }
+    interface RTCSctpTransport extends wrtc.RTCSctpTransport {
+    }
+    interface RTCSessionDescription extends wrtc.RTCSessionDescription {
+    }
+}
+
 declare const AUDIO_STREAM_READY = "AUDIO_STREAM_READY";
 type AssistantEvents = {
     [AUDIO_STREAM_READY]: [{
@@ -29,7 +61,7 @@ declare class Assistant extends EventEmitter<AssistantEvents> {
     joinRoom(roomUrl: string): Promise<void>;
     startLocalMedia(): void;
     getLocalMediaStream(): MediaStream | null;
-    getLocalAudioSource(): wrtc.nonstandard.RTCAudioSource | null;
+    getLocalAudioSource(): wrtc__default.nonstandard.RTCAudioSource | null;
     getRoomConnection(): RoomConnectionClient;
     getCombinedAudioStream(): MediaStream | null;
     getRemoteParticipants(): RemoteParticipantState[];
@@ -122,7 +154,7 @@ declare class Trigger extends EventEmitter$1<TriggerEvents> {
 declare class AudioSource extends PassThrough {
     constructor();
 }
-declare class AudioSink extends wrtc.nonstandard.RTCAudioSink {
+declare class AudioSink extends wrtc__default.nonstandard.RTCAudioSink {
     private _sink;
     constructor(track: MediaStreamTrack);
     subscribe(cb: (d: {

package/dist/index.d.ts

@@ -1,9 +1,41 @@
+import * as wrtc from '@roamhq/wrtc';
+import wrtc__default from '@roamhq/wrtc';
 import { RoomConnectionClient, RemoteParticipantState, ChatMessage } from '@whereby.com/core';
 export { RemoteParticipantState } from '@whereby.com/core';
-import wrtc from '@roamhq/wrtc';
 import EventEmitter, { EventEmitter as EventEmitter$1 } from 'events';
 import { PassThrough } from 'stream';
 
+declare global {
+    interface MediaStream extends wrtc.MediaStream {
+    }
+    interface MediaStreamTrack extends wrtc.MediaStreamTrack {
+    }
+    interface RTCDataChannel extends wrtc.RTCDataChannel {
+    }
+    interface RTCDataChannelEvent extends wrtc.RTCDataChannelEvent {
+    }
+    interface RTCDtlsTransport extends wrtc.RTCDtlsTransport {
+    }
+    interface RTCIceCandidate extends wrtc.RTCIceCandidate {
+    }
+    interface RTCIceTransport extends wrtc.RTCIceTransport {
+    }
+    interface RTCPeerConnection extends wrtc.RTCPeerConnection {
+    }
+    interface RTCPeerConnectionIceEvent extends wrtc.RTCPeerConnectionIceEvent {
+    }
+    interface RTCRtpReceiver extends wrtc.RTCRtpReceiver {
+    }
+    interface RTCRtpSender extends wrtc.RTCRtpSender {
+    }
+    interface RTCRtpTransceiver extends wrtc.RTCRtpTransceiver {
+    }
+    interface RTCSctpTransport extends wrtc.RTCSctpTransport {
+    }
+    interface RTCSessionDescription extends wrtc.RTCSessionDescription {
+    }
+}
+
 declare const AUDIO_STREAM_READY = "AUDIO_STREAM_READY";
 type AssistantEvents = {
     [AUDIO_STREAM_READY]: [{
@@ -29,7 +61,7 @@ declare class Assistant extends EventEmitter<AssistantEvents> {
     joinRoom(roomUrl: string): Promise<void>;
     startLocalMedia(): void;
     getLocalMediaStream(): MediaStream | null;
-    getLocalAudioSource(): wrtc.nonstandard.RTCAudioSource | null;
+    getLocalAudioSource(): wrtc__default.nonstandard.RTCAudioSource | null;
     getRoomConnection(): RoomConnectionClient;
     getCombinedAudioStream(): MediaStream | null;
     getRemoteParticipants(): RemoteParticipantState[];
@@ -122,7 +154,7 @@ declare class Trigger extends EventEmitter$1<TriggerEvents> {
 declare class AudioSource extends PassThrough {
     constructor();
 }
-declare class AudioSink extends wrtc.nonstandard.RTCAudioSink {
+declare class AudioSink extends wrtc__default.nonstandard.RTCAudioSink {
     private _sink;
     constructor(track: MediaStreamTrack);
     subscribe(cb: (d: {

package/dist/index.mjs

@@ -67,9 +67,14 @@ class AudioSink extends wrtc.nonstandard.RTCAudioSink {
     }
 }
 
+// Number of pipes in the ffmpeg process. We predefine a fixed number of slots, and then we dynamically assign
+// participants to these slots based on mute/unmute state.
 const PARTICIPANT_SLOTS = 20;
+// Each sample is 2 bytes (16 bits) for PCM audio - s16le format
+// 48000 Hz is the standard sample rate for WebRTC audio
 const STREAM_INPUT_SAMPLE_RATE_IN_HZ = 48000;
 const BYTES_PER_SAMPLE = 2;
+// 480 samples per 10ms frame at 48kHz
 const FRAME_10MS_SAMPLES = 480;
 const slotBuffers = new Map();
 function appendAndDrainTo480(slot, newSamples) {
@@ -81,10 +86,10 @@ function appendAndDrainTo480(slot, newSamples) {
     let offset = 0;
     while (merged.length - offset >= FRAME_10MS_SAMPLES) {
         const chunk = merged.subarray(offset, offset + FRAME_10MS_SAMPLES);
-        enqueueFrame(slot, chunk);
+        enqueueFrame(slot, chunk); // always 480
         offset += FRAME_10MS_SAMPLES;
     }
-    slotBuffers.set(slot, merged.subarray(offset));
+    slotBuffers.set(slot, merged.subarray(offset)); // keep remainder
 }
 ({
     enqFrames: new Array(PARTICIPANT_SLOTS).fill(0),
@@ -96,6 +101,10 @@ function appendAndDrainTo480(slot, newSamples) {
 let slots = [];
 let stopPacerFn = null;
 let outputPacerState = null;
+/**
+ * Simple linear interpolation resampler to convert audio to 48kHz.
+ * This handles the common case of 16kHz -> 48kHz (3x upsampling).
+ */
 function resampleTo48kHz(inputSamples, inputSampleRate, inputFrames) {
     const ratio = STREAM_INPUT_SAMPLE_RATE_IN_HZ / inputSampleRate;
     const outputLength = Math.floor(inputFrames * ratio);
@@ -115,11 +124,43 @@ function resampleTo48kHz(inputSamples, inputSampleRate, inputFrames) {
     }
     return output;
 }
+/**
+ * Enqueue an audio frame for paced delivery to the RTCAudioSource.
+ */
 function enqueueOutputFrame(samples) {
     if (outputPacerState) {
         outputPacerState.frameQueue.push(samples);
     }
 }
+/**
+ * Start the audio pacer loop for all input slots in an FFmpeg process.
+ *
+ * The pacer ensures each slot (pipe:3..3+N-1) is written to at a steady
+ * real-time rate (e.g. 10 ms = 480 samples @ 48kHz), even if WebRTC frames
+ * arrive jittery, bursty, or with slightly different clocks.
+ *
+ * Key behavior:
+ * - Writes exactly one frame per period, on a shared wall-clock grid.
+ * - Uses silence (zero-filled frame) if a slot's queue is empty, so timing
+ *   never stalls.
+ * - Resnaps the schedule if a slot switches between 10 ms / 20 ms frames.
+ * - Honors Node stream backpressure (`write()` return false) without breaking
+ *   the timing grid.
+ *
+ * This keeps all FFmpeg inputs phase-aligned and stable, so aresample/amix
+ * can mix them without slow-downs or drift.
+ *
+ * Call this once right after spawning FFmpeg:
+ * ```ts
+ * const ff = spawnFFmpegProcess();
+ * startPacer(ff, PARTICIPANT_SLOTS);
+ * ```
+ *
+ * When tearing down the mixer, always call `stopPacer()` before killing FFmpeg.
+ *
+ * @param ff        Child process handle from spawn("ffmpeg", ...)
+ * @param slotCount Number of participant input slots (0..N-1 → fd 3..3+N-1)
+ */
 function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     if (stopPacerFn) {
         stopPacerFn();
@@ -127,11 +168,11 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     }
     const writers = Array.from({ length: slotCount }, (_, i) => ff.stdio[3 + i]);
     const nowMs = () => Number(process.hrtime.bigint()) / 1e6;
-    const outputFrameMs = (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000;
+    const outputFrameMs = (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000; // 10ms
     const t0 = nowMs();
     slots = Array.from({ length: slotCount }, () => ({
         q: [],
-        lastFrames: FRAME_10MS_SAMPLES,
+        lastFrames: FRAME_10MS_SAMPLES, // keep constant
         nextDueMs: t0 + (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000,
     }));
     outputPacerState = {
@@ -146,10 +187,11 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
         for (let s = 0; s < slotCount; s++) {
             const st = slots[s];
             const w = writers[s];
-            const frameMs = (st.lastFrames / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000;
+            const frameMs = (st.lastFrames / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000; // 10ms if 480, 20ms if 960
             if (t >= st.nextDueMs) {
                 const buf = st.q.length ? st.q.shift() : Buffer.alloc(st.lastFrames * BYTES_PER_SAMPLE);
                 if (!w.write(buf)) {
+                    // Just continue without adding drain listener - backpressure will naturally resolve
                     const late = t - st.nextDueMs;
                     const steps = Math.max(1, Math.ceil(late / frameMs));
                     st.nextDueMs += steps * frameMs;
@@ -162,9 +204,10 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
         }
         if (!outputPacerState)
             return;
+        // Handle output pacer for RTCAudioSource
         const state = outputPacerState;
         if (t >= state.nextDueMs) {
-            const samples = state.frameQueue.length > 0 ? state.frameQueue.shift() : new Int16Array(FRAME_10MS_SAMPLES);
+            const samples = state.frameQueue.length > 0 ? state.frameQueue.shift() : new Int16Array(FRAME_10MS_SAMPLES); // silence
             if (!state.didEmitReadyEvent) {
                 state.onAudioStreamReady();
                 state.didEmitReadyEvent = true;
@@ -180,12 +223,20 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     }, 5);
     stopPacerFn = () => clearInterval(iv);
 }
+/**
+ * Stop the audio pacer loop and clear all input slots.
+ * Call this before killing the FFmpeg process to ensure clean shutdown.
+ */
 function stopPacer() {
     if (stopPacerFn)
         stopPacerFn();
     stopPacerFn = null;
     slots = [];
 }
+/**
+ * Queue a live frame for a given slot (0..N-1).
+ * Auto-resnaps the slot's schedule if the frame size (480/960) changes.
+ */
 function enqueueFrame(slot, samples, numberOfFrames) {
     const st = slots[slot];
     if (!st)
@@ -193,6 +244,10 @@ function enqueueFrame(slot, samples, numberOfFrames) {
     const buf = Buffer.from(samples.buffer, samples.byteOffset, samples.byteLength);
     st.q.push(buf);
 }
+/**
+ * Clear the audio queue for a specific slot when a participant leaves.
+ * This prevents stale audio data from continuing to play after disconnect.
+ */
 function clearSlotQueue(slot) {
     const st = slots[slot];
     if (st) {
@@ -202,6 +257,11 @@ function clearSlotQueue(slot) {
         st.nextDueMs = now + frameMs;
     }
 }
+/**
+ * Get the FFmpeg arguments for mixing audio from multiple participants.
+ * This will read from the input pipes (3..3+N-1) and output a single mixed audio stream.
+ * The output is in PCM 16-bit little-endian format at 48kHz sample rate.
+ */
 function getFFmpegArguments() {
     const N = PARTICIPANT_SLOTS;
     const SR = STREAM_INPUT_SAMPLE_RATE_IN_HZ;
@@ -219,6 +279,14 @@ function getFFmpegArguments() {
     ffArgs.push("-hide_banner", "-nostats", "-loglevel", "error", "-filter_complex", filter, "-map", "[mix]", "-f", "s16le", "-ar", String(SR), "-ac", "1", "-c:a", "pcm_s16le", "pipe:1");
     return ffArgs;
 }
+/**
+ * Spawn a new FFmpeg process for mixing audio from multiple participants.
+ * This will read from the input pipes (3..3+N-1) and output a single mixed audio stream.
+ * The output is in PCM 16-bit little-endian format at 48kHz sample rate.
+ * The process will log its output to stderr.
+ * @param rtcAudioSource The RTCAudioSource to which the mixed audio will be sent.
+ * @return The spawned FFmpeg process.
+ */
 function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     const stdio = ["ignore", "pipe", "pipe", ...Array(PARTICIPANT_SLOTS).fill("pipe")];
     const args = getFFmpegArguments();
@@ -228,7 +296,7 @@ function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     ffmpegProcess.stderr.on("data", (d) => console.error("[ffmpeg]", String(d).trim()));
     ffmpegProcess.on("error", () => console.error("FFmpeg process error: is ffmpeg installed?"));
     let audioBuffer = Buffer.alloc(0);
-    const FRAME_SIZE_BYTES = FRAME_10MS_SAMPLES * BYTES_PER_SAMPLE;
+    const FRAME_SIZE_BYTES = FRAME_10MS_SAMPLES * BYTES_PER_SAMPLE; // 480 samples * 2 bytes = 960 bytes
     ffmpegProcess.stdout.on("data", (chunk) => {
         audioBuffer = Buffer.concat([audioBuffer, chunk]);
         while (audioBuffer.length >= FRAME_SIZE_BYTES) {
@@ -243,6 +311,16 @@ function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     });
     return ffmpegProcess;
 }
+/**
+ * Write audio data from a MediaStreamTrack to the FFmpeg process.
+ * This function creates an AudioSink for the track and sets up a data handler
+ * that enqueues audio frames into the pacer.
+ *
+ * @param ffmpegProcess The FFmpeg process to which audio data will be written.
+ * @param slot The participant slot number (0..N-1) to which this track belongs.
+ * @param audioTrack The MediaStreamTrack containing the audio data.
+ * @return An object containing the AudioSink, the writable stream, and a stop function.
+ */
 function writeAudioDataToFFmpeg(ffmpegProcess, slot, audioTrack) {
     const writer = ffmpegProcess.stdio[3 + slot];
     const sink = new AudioSink(audioTrack);
@@ -267,6 +345,12 @@ function writeAudioDataToFFmpeg(ffmpegProcess, slot, audioTrack) {
     };
     return { sink, writer, stop };
 }
+/**
+ * Stop the FFmpeg process and clean up all resources.
+ * This function will unpipe the stdout, end all writable streams for each participant slot,
+ * and kill the FFmpeg process.
+ * @param ffmpegProcess The FFmpeg process to stop.
+ */
 function stopFFmpegProcess(ffmpegProcess) {
     stopPacer();
     if (ffmpegProcess && !ffmpegProcess.killed) {
@@ -320,6 +404,7 @@ class AudioMixer extends EventEmitter {
         for (const p of participants)
             this.attachParticipantIfNeeded(p);
         const liveIds = new Set(participants.map((p) => p.id).filter(Boolean));
+        // eslint-disable-next-line @typescript-eslint/no-unused-vars
         for (const [slot, pid] of this.participantSlots) {
             if (pid && !liveIds.has(pid))
                 this.detachParticipant(pid);
@@ -332,6 +417,7 @@ class AudioMixer extends EventEmitter {
         }
         this.participantSlots = new Map(Array.from({ length: PARTICIPANT_SLOTS }, (_, i) => [i, ""]));
         this.activeSlots = {};
+        // Recreate the media stream to avoid stale references
         this.setupMediaStream();
     }
     slotForParticipant(participantId) {
@@ -397,6 +483,7 @@ class AudioMixer extends EventEmitter {
             }
             this.activeSlots[slot] = undefined;
         }
+        // Clear any queued audio data for this slot to prevent stale audio
         clearSlotQueue(slot);
         this.participantSlots.set(slot, "");
     }
@@ -524,10 +611,7 @@ dotenv.config();
 const { IS_LOCAL = "false", BIND_INTERFACE = "en0" } = process.env;
 function buildRoomUrl(roomPath, wherebySubdomain, baseDomain = "whereby.com") {
     let wherebyDomain;
-    if (!IS_LOCAL) {
-        wherebyDomain = `${wherebySubdomain}.${baseDomain}`;
-    }
-    else {
+    if (IS_LOCAL === "true") {
         const ifaceAddrs = networkInterfaces()[BIND_INTERFACE];
         if (!ifaceAddrs) {
             throw new Error(`Unknown interface ${BIND_INTERFACE}`);
@@ -538,6 +622,9 @@ function buildRoomUrl(roomPath, wherebySubdomain, baseDomain = "whereby.com") {
         }
         wherebyDomain = `${wherebySubdomain}-ip-${bindAddr.address.replace(/[.]/g, "-")}.hereby.dev:4443`;
     }
+    else {
+        wherebyDomain = `${wherebySubdomain}.${baseDomain}`;
+    }
     return `https://${wherebyDomain}${roomPath}`;
 }
 
@@ -578,6 +665,7 @@ class Trigger extends EventEmitter {
         const router = webhookRouter(this.webhookTriggers, this, this.assistantKey, this.startCombinedAudioStream, this.startLocalMedia);
         app.use(router);
         const server = app.listen(this.port, () => {
+            // console.log(`Bot trigger server now running on port[${this.port}]`);
         });
         process.on("SIGTERM", () => {
             server.close();

package/dist/legacy-esm.js

@@ -67,9 +67,14 @@ class AudioSink extends wrtc.nonstandard.RTCAudioSink {
     }
 }
 
+// Number of pipes in the ffmpeg process. We predefine a fixed number of slots, and then we dynamically assign
+// participants to these slots based on mute/unmute state.
 const PARTICIPANT_SLOTS = 20;
+// Each sample is 2 bytes (16 bits) for PCM audio - s16le format
+// 48000 Hz is the standard sample rate for WebRTC audio
 const STREAM_INPUT_SAMPLE_RATE_IN_HZ = 48000;
 const BYTES_PER_SAMPLE = 2;
+// 480 samples per 10ms frame at 48kHz
 const FRAME_10MS_SAMPLES = 480;
 const slotBuffers = new Map();
 function appendAndDrainTo480(slot, newSamples) {
@@ -81,10 +86,10 @@ function appendAndDrainTo480(slot, newSamples) {
     let offset = 0;
     while (merged.length - offset >= FRAME_10MS_SAMPLES) {
         const chunk = merged.subarray(offset, offset + FRAME_10MS_SAMPLES);
-        enqueueFrame(slot, chunk);
+        enqueueFrame(slot, chunk); // always 480
         offset += FRAME_10MS_SAMPLES;
     }
-    slotBuffers.set(slot, merged.subarray(offset));
+    slotBuffers.set(slot, merged.subarray(offset)); // keep remainder
 }
 ({
     enqFrames: new Array(PARTICIPANT_SLOTS).fill(0),
@@ -96,6 +101,10 @@ function appendAndDrainTo480(slot, newSamples) {
 let slots = [];
 let stopPacerFn = null;
 let outputPacerState = null;
+/**
+ * Simple linear interpolation resampler to convert audio to 48kHz.
+ * This handles the common case of 16kHz -> 48kHz (3x upsampling).
+ */
 function resampleTo48kHz(inputSamples, inputSampleRate, inputFrames) {
     const ratio = STREAM_INPUT_SAMPLE_RATE_IN_HZ / inputSampleRate;
     const outputLength = Math.floor(inputFrames * ratio);
@@ -115,11 +124,43 @@ function resampleTo48kHz(inputSamples, inputSampleRate, inputFrames) {
     }
     return output;
 }
+/**
+ * Enqueue an audio frame for paced delivery to the RTCAudioSource.
+ */
 function enqueueOutputFrame(samples) {
     if (outputPacerState) {
         outputPacerState.frameQueue.push(samples);
     }
 }
+/**
+ * Start the audio pacer loop for all input slots in an FFmpeg process.
+ *
+ * The pacer ensures each slot (pipe:3..3+N-1) is written to at a steady
+ * real-time rate (e.g. 10 ms = 480 samples @ 48kHz), even if WebRTC frames
+ * arrive jittery, bursty, or with slightly different clocks.
+ *
+ * Key behavior:
+ * - Writes exactly one frame per period, on a shared wall-clock grid.
+ * - Uses silence (zero-filled frame) if a slot's queue is empty, so timing
+ *   never stalls.
+ * - Resnaps the schedule if a slot switches between 10 ms / 20 ms frames.
+ * - Honors Node stream backpressure (`write()` return false) without breaking
+ *   the timing grid.
+ *
+ * This keeps all FFmpeg inputs phase-aligned and stable, so aresample/amix
+ * can mix them without slow-downs or drift.
+ *
+ * Call this once right after spawning FFmpeg:
+ * ```ts
+ * const ff = spawnFFmpegProcess();
+ * startPacer(ff, PARTICIPANT_SLOTS);
+ * ```
+ *
+ * When tearing down the mixer, always call `stopPacer()` before killing FFmpeg.
+ *
+ * @param ff        Child process handle from spawn("ffmpeg", ...)
+ * @param slotCount Number of participant input slots (0..N-1 → fd 3..3+N-1)
+ */
 function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     if (stopPacerFn) {
         stopPacerFn();
@@ -127,11 +168,11 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     }
     const writers = Array.from({ length: slotCount }, (_, i) => ff.stdio[3 + i]);
     const nowMs = () => Number(process.hrtime.bigint()) / 1e6;
-    const outputFrameMs = (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000;
+    const outputFrameMs = (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000; // 10ms
     const t0 = nowMs();
     slots = Array.from({ length: slotCount }, () => ({
         q: [],
-        lastFrames: FRAME_10MS_SAMPLES,
+        lastFrames: FRAME_10MS_SAMPLES, // keep constant
         nextDueMs: t0 + (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000,
     }));
     outputPacerState = {
@@ -146,10 +187,11 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
         for (let s = 0; s < slotCount; s++) {
             const st = slots[s];
             const w = writers[s];
-            const frameMs = (st.lastFrames / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000;
+            const frameMs = (st.lastFrames / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000; // 10ms if 480, 20ms if 960
             if (t >= st.nextDueMs) {
                 const buf = st.q.length ? st.q.shift() : Buffer.alloc(st.lastFrames * BYTES_PER_SAMPLE);
                 if (!w.write(buf)) {
+                    // Just continue without adding drain listener - backpressure will naturally resolve
                     const late = t - st.nextDueMs;
                     const steps = Math.max(1, Math.ceil(late / frameMs));
                     st.nextDueMs += steps * frameMs;
@@ -162,9 +204,10 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
         }
         if (!outputPacerState)
             return;
+        // Handle output pacer for RTCAudioSource
         const state = outputPacerState;
         if (t >= state.nextDueMs) {
-            const samples = state.frameQueue.length > 0 ? state.frameQueue.shift() : new Int16Array(FRAME_10MS_SAMPLES);
+            const samples = state.frameQueue.length > 0 ? state.frameQueue.shift() : new Int16Array(FRAME_10MS_SAMPLES); // silence
             if (!state.didEmitReadyEvent) {
                 state.onAudioStreamReady();
                 state.didEmitReadyEvent = true;
@@ -180,12 +223,20 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     }, 5);
     stopPacerFn = () => clearInterval(iv);
 }
+/**
+ * Stop the audio pacer loop and clear all input slots.
+ * Call this before killing the FFmpeg process to ensure clean shutdown.
+ */
 function stopPacer() {
     if (stopPacerFn)
         stopPacerFn();
     stopPacerFn = null;
     slots = [];
 }
+/**
+ * Queue a live frame for a given slot (0..N-1).
+ * Auto-resnaps the slot's schedule if the frame size (480/960) changes.
+ */
 function enqueueFrame(slot, samples, numberOfFrames) {
     const st = slots[slot];
     if (!st)
@@ -193,6 +244,10 @@ function enqueueFrame(slot, samples, numberOfFrames) {
     const buf = Buffer.from(samples.buffer, samples.byteOffset, samples.byteLength);
     st.q.push(buf);
 }
+/**
+ * Clear the audio queue for a specific slot when a participant leaves.
+ * This prevents stale audio data from continuing to play after disconnect.
+ */
 function clearSlotQueue(slot) {
     const st = slots[slot];
     if (st) {
@@ -202,6 +257,11 @@ function clearSlotQueue(slot) {
         st.nextDueMs = now + frameMs;
     }
 }
+/**
+ * Get the FFmpeg arguments for mixing audio from multiple participants.
+ * This will read from the input pipes (3..3+N-1) and output a single mixed audio stream.
+ * The output is in PCM 16-bit little-endian format at 48kHz sample rate.
+ */
 function getFFmpegArguments() {
     const N = PARTICIPANT_SLOTS;
     const SR = STREAM_INPUT_SAMPLE_RATE_IN_HZ;
@@ -219,6 +279,14 @@ function getFFmpegArguments() {
     ffArgs.push("-hide_banner", "-nostats", "-loglevel", "error", "-filter_complex", filter, "-map", "[mix]", "-f", "s16le", "-ar", String(SR), "-ac", "1", "-c:a", "pcm_s16le", "pipe:1");
     return ffArgs;
 }
+/**
+ * Spawn a new FFmpeg process for mixing audio from multiple participants.
+ * This will read from the input pipes (3..3+N-1) and output a single mixed audio stream.
+ * The output is in PCM 16-bit little-endian format at 48kHz sample rate.
+ * The process will log its output to stderr.
+ * @param rtcAudioSource The RTCAudioSource to which the mixed audio will be sent.
+ * @return The spawned FFmpeg process.
+ */
 function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     const stdio = ["ignore", "pipe", "pipe", ...Array(PARTICIPANT_SLOTS).fill("pipe")];
     const args = getFFmpegArguments();
@@ -228,7 +296,7 @@ function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     ffmpegProcess.stderr.on("data", (d) => console.error("[ffmpeg]", String(d).trim()));
     ffmpegProcess.on("error", () => console.error("FFmpeg process error: is ffmpeg installed?"));
     let audioBuffer = Buffer.alloc(0);
-    const FRAME_SIZE_BYTES = FRAME_10MS_SAMPLES * BYTES_PER_SAMPLE;
+    const FRAME_SIZE_BYTES = FRAME_10MS_SAMPLES * BYTES_PER_SAMPLE; // 480 samples * 2 bytes = 960 bytes
     ffmpegProcess.stdout.on("data", (chunk) => {
         audioBuffer = Buffer.concat([audioBuffer, chunk]);
         while (audioBuffer.length >= FRAME_SIZE_BYTES) {
@@ -243,6 +311,16 @@ function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     });
     return ffmpegProcess;
 }
+/**
+ * Write audio data from a MediaStreamTrack to the FFmpeg process.
+ * This function creates an AudioSink for the track and sets up a data handler
+ * that enqueues audio frames into the pacer.
+ *
+ * @param ffmpegProcess The FFmpeg process to which audio data will be written.
+ * @param slot The participant slot number (0..N-1) to which this track belongs.
+ * @param audioTrack The MediaStreamTrack containing the audio data.
+ * @return An object containing the AudioSink, the writable stream, and a stop function.
+ */
 function writeAudioDataToFFmpeg(ffmpegProcess, slot, audioTrack) {
     const writer = ffmpegProcess.stdio[3 + slot];
     const sink = new AudioSink(audioTrack);
@@ -267,6 +345,12 @@ function writeAudioDataToFFmpeg(ffmpegProcess, slot, audioTrack) {
     };
     return { sink, writer, stop };
 }
+/**
+ * Stop the FFmpeg process and clean up all resources.
+ * This function will unpipe the stdout, end all writable streams for each participant slot,
+ * and kill the FFmpeg process.
+ * @param ffmpegProcess The FFmpeg process to stop.
+ */
 function stopFFmpegProcess(ffmpegProcess) {
     stopPacer();
     if (ffmpegProcess && !ffmpegProcess.killed) {
@@ -320,6 +404,7 @@ class AudioMixer extends EventEmitter {
         for (const p of participants)
             this.attachParticipantIfNeeded(p);
         const liveIds = new Set(participants.map((p) => p.id).filter(Boolean));
+        // eslint-disable-next-line @typescript-eslint/no-unused-vars
         for (const [slot, pid] of this.participantSlots) {
             if (pid && !liveIds.has(pid))
                 this.detachParticipant(pid);
@@ -332,6 +417,7 @@ class AudioMixer extends EventEmitter {
         }
         this.participantSlots = new Map(Array.from({ length: PARTICIPANT_SLOTS }, (_, i) => [i, ""]));
         this.activeSlots = {};
+        // Recreate the media stream to avoid stale references
         this.setupMediaStream();
     }
     slotForParticipant(participantId) {
@@ -397,6 +483,7 @@ class AudioMixer extends EventEmitter {
             }
             this.activeSlots[slot] = undefined;
         }
+        // Clear any queued audio data for this slot to prevent stale audio
         clearSlotQueue(slot);
         this.participantSlots.set(slot, "");
     }
@@ -524,10 +611,7 @@ dotenv.config();
 const { IS_LOCAL = "false", BIND_INTERFACE = "en0" } = process.env;
 function buildRoomUrl(roomPath, wherebySubdomain, baseDomain = "whereby.com") {
     let wherebyDomain;
-    if (!IS_LOCAL) {
-        wherebyDomain = `${wherebySubdomain}.${baseDomain}`;
-    }
-    else {
+    if (IS_LOCAL === "true") {
         const ifaceAddrs = networkInterfaces()[BIND_INTERFACE];
         if (!ifaceAddrs) {
             throw new Error(`Unknown interface ${BIND_INTERFACE}`);
@@ -538,6 +622,9 @@ function buildRoomUrl(roomPath, wherebySubdomain, baseDomain = "whereby.com") {
         }
         wherebyDomain = `${wherebySubdomain}-ip-${bindAddr.address.replace(/[.]/g, "-")}.hereby.dev:4443`;
     }
+    else {
+        wherebyDomain = `${wherebySubdomain}.${baseDomain}`;
+    }
     return `https://${wherebyDomain}${roomPath}`;
 }
 
@@ -578,6 +665,7 @@ class Trigger extends EventEmitter {
         const router = webhookRouter(this.webhookTriggers, this, this.assistantKey, this.startCombinedAudioStream, this.startLocalMedia);
         app.use(router);
         const server = app.listen(this.port, () => {
+            // console.log(`Bot trigger server now running on port[${this.port}]`);
         });
         process.on("SIGTERM", () => {
             server.close();

package/dist/polyfills.cjs

@@ -38,8 +38,10 @@ typeof SuppressedError === "function" ? SuppressedError : function (error, suppr
 
 function setWebsocketOrigin(roomUrl) {
     try {
+        // add pathname needed for parsing in rtcstats-server.
         const url = new URL(roomUrl);
         global.window.location.pathname = url.pathname;
+        // fix origin header needed for parsing in rtcstats-server.
         const defaultClientOptions = {
             origin: url.origin,
         };
@@ -90,6 +92,10 @@ class RTCPeerConnection extends wrtc.RTCPeerConnection {
     }
     getStats(arg) {
         return __awaiter(this, void 0, void 0, function* () {
+            /**
+             * node-wrtc seems to expect an Object argument, and doesn't handle the null arg we pass, so we
+             * wrap the call and filter the arg
+             **/
             arg = arg instanceof Object ? arg : undefined;
             const stats = yield this.wrappedGetStats(arg);
             return stats;
@@ -112,6 +118,6 @@ global.RTCRtpSender = wrtc.RTCRtpSender;
 global.RTCRtpTransceiver = wrtc.RTCRtpTransceiver;
 global.RTCSctpTransport = wrtc.RTCSctpTransport;
 global.RTCSessionDescription = wrtc.RTCSessionDescription;
-global.window = Object.assign(Object.assign({}, global), { location: { pathname: "" }, screen: { width: 0 }, setInterval: global.setInterval });
+global.window = Object.assign(Object.assign({}, global), { location: { pathname: "" }, screen: { width: 0 }, setInterval: global.setInterval }); // make sure all the classes / setInterval are available on window for rtcstats
 
 exports.setWebsocketOrigin = setWebsocketOrigin;

package/dist/tools.cjs

@@ -19,9 +19,14 @@ class AudioSink extends wrtc.nonstandard.RTCAudioSink {
     }
 }
 
+// Number of pipes in the ffmpeg process. We predefine a fixed number of slots, and then we dynamically assign
+// participants to these slots based on mute/unmute state.
 const PARTICIPANT_SLOTS = 20;
+// Each sample is 2 bytes (16 bits) for PCM audio - s16le format
+// 48000 Hz is the standard sample rate for WebRTC audio
 const STREAM_INPUT_SAMPLE_RATE_IN_HZ = 48000;
 const BYTES_PER_SAMPLE = 2;
+// 480 samples per 10ms frame at 48kHz
 const FRAME_10MS_SAMPLES = 480;
 const slotBuffers = new Map();
 function appendAndDrainTo480(slot, newSamples) {
@@ -33,10 +38,10 @@ function appendAndDrainTo480(slot, newSamples) {
     let offset = 0;
     while (merged.length - offset >= FRAME_10MS_SAMPLES) {
         const chunk = merged.subarray(offset, offset + FRAME_10MS_SAMPLES);
-        enqueueFrame(slot, chunk);
+        enqueueFrame(slot, chunk); // always 480
         offset += FRAME_10MS_SAMPLES;
     }
-    slotBuffers.set(slot, merged.subarray(offset));
+    slotBuffers.set(slot, merged.subarray(offset)); // keep remainder
 }
 ({
     enqFrames: new Array(PARTICIPANT_SLOTS).fill(0),
@@ -48,6 +53,10 @@ function appendAndDrainTo480(slot, newSamples) {
 let slots = [];
 let stopPacerFn = null;
 let outputPacerState = null;
+/**
+ * Simple linear interpolation resampler to convert audio to 48kHz.
+ * This handles the common case of 16kHz -> 48kHz (3x upsampling).
+ */
 function resampleTo48kHz(inputSamples, inputSampleRate, inputFrames) {
     const ratio = STREAM_INPUT_SAMPLE_RATE_IN_HZ / inputSampleRate;
     const outputLength = Math.floor(inputFrames * ratio);
@@ -67,11 +76,43 @@ function resampleTo48kHz(inputSamples, inputSampleRate, inputFrames) {
     }
     return output;
 }
+/**
+ * Enqueue an audio frame for paced delivery to the RTCAudioSource.
+ */
 function enqueueOutputFrame(samples) {
     if (outputPacerState) {
         outputPacerState.frameQueue.push(samples);
     }
 }
+/**
+ * Start the audio pacer loop for all input slots in an FFmpeg process.
+ *
+ * The pacer ensures each slot (pipe:3..3+N-1) is written to at a steady
+ * real-time rate (e.g. 10 ms = 480 samples @ 48kHz), even if WebRTC frames
+ * arrive jittery, bursty, or with slightly different clocks.
+ *
+ * Key behavior:
+ * - Writes exactly one frame per period, on a shared wall-clock grid.
+ * - Uses silence (zero-filled frame) if a slot's queue is empty, so timing
+ *   never stalls.
+ * - Resnaps the schedule if a slot switches between 10 ms / 20 ms frames.
+ * - Honors Node stream backpressure (`write()` return false) without breaking
+ *   the timing grid.
+ *
+ * This keeps all FFmpeg inputs phase-aligned and stable, so aresample/amix
+ * can mix them without slow-downs or drift.
+ *
+ * Call this once right after spawning FFmpeg:
+ * ```ts
+ * const ff = spawnFFmpegProcess();
+ * startPacer(ff, PARTICIPANT_SLOTS);
+ * ```
+ *
+ * When tearing down the mixer, always call `stopPacer()` before killing FFmpeg.
+ *
+ * @param ff        Child process handle from spawn("ffmpeg", ...)
+ * @param slotCount Number of participant input slots (0..N-1 → fd 3..3+N-1)
+ */
 function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     if (stopPacerFn) {
         stopPacerFn();
@@ -79,11 +120,11 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     }
     const writers = Array.from({ length: slotCount }, (_, i) => ff.stdio[3 + i]);
     const nowMs = () => Number(process.hrtime.bigint()) / 1e6;
-    const outputFrameMs = (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000;
+    const outputFrameMs = (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000; // 10ms
     const t0 = nowMs();
     slots = Array.from({ length: slotCount }, () => ({
         q: [],
-        lastFrames: FRAME_10MS_SAMPLES,
+        lastFrames: FRAME_10MS_SAMPLES, // keep constant
         nextDueMs: t0 + (FRAME_10MS_SAMPLES / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000,
     }));
     outputPacerState = {
@@ -98,10 +139,11 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
         for (let s = 0; s < slotCount; s++) {
             const st = slots[s];
             const w = writers[s];
-            const frameMs = (st.lastFrames / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000;
+            const frameMs = (st.lastFrames / STREAM_INPUT_SAMPLE_RATE_IN_HZ) * 1000; // 10ms if 480, 20ms if 960
             if (t >= st.nextDueMs) {
                 const buf = st.q.length ? st.q.shift() : Buffer.alloc(st.lastFrames * BYTES_PER_SAMPLE);
                 if (!w.write(buf)) {
+                    // Just continue without adding drain listener - backpressure will naturally resolve
                     const late = t - st.nextDueMs;
                     const steps = Math.max(1, Math.ceil(late / frameMs));
                     st.nextDueMs += steps * frameMs;
@@ -114,9 +156,10 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
         }
         if (!outputPacerState)
             return;
+        // Handle output pacer for RTCAudioSource
         const state = outputPacerState;
         if (t >= state.nextDueMs) {
-            const samples = state.frameQueue.length > 0 ? state.frameQueue.shift() : new Int16Array(FRAME_10MS_SAMPLES);
+            const samples = state.frameQueue.length > 0 ? state.frameQueue.shift() : new Int16Array(FRAME_10MS_SAMPLES); // silence
             if (!state.didEmitReadyEvent) {
                 state.onAudioStreamReady();
                 state.didEmitReadyEvent = true;
@@ -132,12 +175,20 @@ function startPacer(ff, slotCount, rtcAudioSource, onAudioStreamReady) {
     }, 5);
     stopPacerFn = () => clearInterval(iv);
 }
+/**
+ * Stop the audio pacer loop and clear all input slots.
+ * Call this before killing the FFmpeg process to ensure clean shutdown.
+ */
 function stopPacer() {
     if (stopPacerFn)
         stopPacerFn();
     stopPacerFn = null;
     slots = [];
 }
+/**
+ * Queue a live frame for a given slot (0..N-1).
+ * Auto-resnaps the slot's schedule if the frame size (480/960) changes.
+ */
 function enqueueFrame(slot, samples, numberOfFrames) {
     const st = slots[slot];
     if (!st)
@@ -145,6 +196,10 @@ function enqueueFrame(slot, samples, numberOfFrames) {
     const buf = Buffer.from(samples.buffer, samples.byteOffset, samples.byteLength);
     st.q.push(buf);
 }
+/**
+ * Clear the audio queue for a specific slot when a participant leaves.
+ * This prevents stale audio data from continuing to play after disconnect.
+ */
 function clearSlotQueue(slot) {
     const st = slots[slot];
     if (st) {
@@ -154,6 +209,11 @@ function clearSlotQueue(slot) {
         st.nextDueMs = now + frameMs;
     }
 }
+/**
+ * Get the FFmpeg arguments for mixing audio from multiple participants.
+ * This will read from the input pipes (3..3+N-1) and output a single mixed audio stream.
+ * The output is in PCM 16-bit little-endian format at 48kHz sample rate.
+ */
 function getFFmpegArguments() {
     const N = PARTICIPANT_SLOTS;
     const SR = STREAM_INPUT_SAMPLE_RATE_IN_HZ;
@@ -171,6 +231,14 @@ function getFFmpegArguments() {
     ffArgs.push("-hide_banner", "-nostats", "-loglevel", "error", "-filter_complex", filter, "-map", "[mix]", "-f", "s16le", "-ar", String(SR), "-ac", "1", "-c:a", "pcm_s16le", "pipe:1");
     return ffArgs;
 }
+/**
+ * Spawn a new FFmpeg process for mixing audio from multiple participants.
+ * This will read from the input pipes (3..3+N-1) and output a single mixed audio stream.
+ * The output is in PCM 16-bit little-endian format at 48kHz sample rate.
+ * The process will log its output to stderr.
+ * @param rtcAudioSource The RTCAudioSource to which the mixed audio will be sent.
+ * @return The spawned FFmpeg process.
+ */
 function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     const stdio = ["ignore", "pipe", "pipe", ...Array(PARTICIPANT_SLOTS).fill("pipe")];
     const args = getFFmpegArguments();
@@ -180,7 +248,7 @@ function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     ffmpegProcess.stderr.on("data", (d) => console.error("[ffmpeg]", String(d).trim()));
     ffmpegProcess.on("error", () => console.error("FFmpeg process error: is ffmpeg installed?"));
     let audioBuffer = Buffer.alloc(0);
-    const FRAME_SIZE_BYTES = FRAME_10MS_SAMPLES * BYTES_PER_SAMPLE;
+    const FRAME_SIZE_BYTES = FRAME_10MS_SAMPLES * BYTES_PER_SAMPLE; // 480 samples * 2 bytes = 960 bytes
     ffmpegProcess.stdout.on("data", (chunk) => {
         audioBuffer = Buffer.concat([audioBuffer, chunk]);
         while (audioBuffer.length >= FRAME_SIZE_BYTES) {
@@ -195,6 +263,16 @@ function spawnFFmpegProcess(rtcAudioSource, onAudioStreamReady) {
     });
     return ffmpegProcess;
 }
+/**
+ * Write audio data from a MediaStreamTrack to the FFmpeg process.
+ * This function creates an AudioSink for the track and sets up a data handler
+ * that enqueues audio frames into the pacer.
+ *
+ * @param ffmpegProcess The FFmpeg process to which audio data will be written.
+ * @param slot The participant slot number (0..N-1) to which this track belongs.
+ * @param audioTrack The MediaStreamTrack containing the audio data.
+ * @return An object containing the AudioSink, the writable stream, and a stop function.
+ */
 function writeAudioDataToFFmpeg(ffmpegProcess, slot, audioTrack) {
     const writer = ffmpegProcess.stdio[3 + slot];
     const sink = new AudioSink(audioTrack);
@@ -219,6 +297,12 @@ function writeAudioDataToFFmpeg(ffmpegProcess, slot, audioTrack) {
     };
     return { sink, writer, stop };
 }
+/**
+ * Stop the FFmpeg process and clean up all resources.
+ * This function will unpipe the stdout, end all writable streams for each participant slot,
+ * and kill the FFmpeg process.
+ * @param ffmpegProcess The FFmpeg process to stop.
+ */
 function stopFFmpegProcess(ffmpegProcess) {
     stopPacer();
     if (ffmpegProcess && !ffmpegProcess.killed) {
@@ -272,6 +356,7 @@ class AudioMixer extends events.EventEmitter {
         for (const p of participants)
             this.attachParticipantIfNeeded(p);
         const liveIds = new Set(participants.map((p) => p.id).filter(Boolean));
+        // eslint-disable-next-line @typescript-eslint/no-unused-vars
         for (const [slot, pid] of this.participantSlots) {
             if (pid && !liveIds.has(pid))
                 this.detachParticipant(pid);
@@ -284,6 +369,7 @@ class AudioMixer extends events.EventEmitter {
         }
         this.participantSlots = new Map(Array.from({ length: PARTICIPANT_SLOTS }, (_, i) => [i, ""]));
         this.activeSlots = {};
+        // Recreate the media stream to avoid stale references
         this.setupMediaStream();
     }
     slotForParticipant(participantId) {
@@ -349,6 +435,7 @@ class AudioMixer extends events.EventEmitter {
             }
             this.activeSlots[slot] = undefined;
         }
+        // Clear any queued audio data for this slot to prevent stale audio
         clearSlotQueue(slot);
         this.participantSlots.set(slot, "");
     }

package/package.json

@@ -2,7 +2,7 @@
   "name": "@whereby.com/assistant-sdk",
   "description": "Assistant SDK for whereby.com",
   "author": "Whereby AS",
-  "version": "0.0.0-canary-20250912102624",
+  "version": "0.0.0-canary-20250912144626",
   "license": "MIT",
   "files": [
     "dist",
@@ -63,7 +63,7 @@
     "dotenv": "^16.4.5",
     "uuid": "^11.0.3",
     "ws": "^8.18.0",
-    "@whereby.com/core": "0.0.0-canary-20250912102624"
+    "@whereby.com/core": "0.0.0-canary-20250912144626"
   },
   "prettier": "@whereby.com/prettier-config",
   "scripts": {