@alexkroman1/aai-ui 2.0.0 → 3.0.0

package/dist/session-core-types.d.ts

@@ -27,17 +27,7 @@ export type CustomEvent = {
  */
 export type SessionSnapshot = {
     readonly state: AgentState;
-    /**
-     * False when the server declared the session text-only (`tts: none()`):
-     * no audio frames will arrive and replies render as text. True until the
-     * server's `config` message says otherwise (voice is the default).
-     */
-    readonly audioOut: boolean;
-    /**
-     * True while the microphone is live and streaming to the server. Voice
-     * sessions record for their whole lifetime; text-only sessions toggle this
-     * via `startRecording()` / `stopRecording()` (the record button).
-     */
+    /** True while the microphone is live and streaming to the server. */
     readonly recording: boolean;
     /**
      * The WebSocket URL a program can connect to directly (the same endpoint
@@ -94,22 +84,6 @@ export type SessionCore = {
     start(): void;
     /** Toggle between connected and disconnected states. */
     toggle(): void;
-    /**
-     * Start streaming microphone audio (text-only sessions). Requests mic
-     * access on first use. No-op in voice sessions, where the mic is always on.
-     */
-    startRecording(): void;
-    /** Stop streaming microphone audio (text-only sessions). No-op in voice sessions. */
-    stopRecording(): void;
-    /**
-     * Decode an audio file (any format the browser can decode), resample it to
-     * the session's STT rate, and stream it to the server for transcription.
-     * Text-only sessions (`tts: none()`) only — voice sessions stream the
-     * microphone instead and reject. Also rejects when no session is connected,
-     * the mic is recording, another upload is in flight, or the file cannot be
-     * decoded. Resolves once the audio has been handed to the socket.
-     */
-    sendAudioFile(file: Blob): Promise<void>;
     /** Alias for `disconnect` for use with `using`. */
     [Symbol.dispose](): void;
 };

package/dist/session-core.js

@@ -1,2 +1,2 @@
-import { t as createSessionCore } from "./session-core-BLhiQ18c.js";
+import { t as createSessionCore } from "./session-core-BM3WHeoY.js";
 export { createSessionCore };

package/dist/types.d.ts

@@ -1,5 +1,5 @@
 import type { SessionErrorCode } from "@alexkroman1/aai/protocol";
-export { FILE_SEND_BACKOFF_MS, MIC_BUFFER_SECONDS, MIC_SEND_MAX_BUFFERED_BYTES, MIC_SILENCE_PROBE_MS, PLAYBACK_CONCEAL_FADE_MS, PLAYBACK_CONCEAL_FLOOR, PLAYBACK_JITTER_MS, PLAYBACK_REFILL_MS, } from "@alexkroman1/aai";
+export { CAPTURE_STOP_ACK_TIMEOUT_MS, DEFAULT_STT_SAMPLE_RATE, MIC_BUFFER_SECONDS, MIC_SEND_MAX_BUFFERED_BYTES, MIC_SILENCE_PROBE_MS, PLAYBACK_BUFFER_SECONDS, PLAYBACK_CONCEAL_FADE_MS, PLAYBACK_CONCEAL_FLOOR, PLAYBACK_DONE_MAX_WAIT_MS, PLAYBACK_DONE_POLL_MS, PLAYBACK_JITTER_MS, PLAYBACK_REFILL_MS, } from "@alexkroman1/aai";
 /**
  * `getUserMedia` audio constraints for every capture path in this package.
  *

package/dist/types.js

@@ -1,4 +1,4 @@
-import { FILE_SEND_BACKOFF_MS, MIC_BUFFER_SECONDS, MIC_SEND_MAX_BUFFERED_BYTES, MIC_SILENCE_PROBE_MS, PLAYBACK_CONCEAL_FADE_MS, PLAYBACK_CONCEAL_FLOOR, PLAYBACK_JITTER_MS, PLAYBACK_REFILL_MS } from "@alexkroman1/aai";
+import { CAPTURE_STOP_ACK_TIMEOUT_MS, DEFAULT_STT_SAMPLE_RATE, MIC_BUFFER_SECONDS, MIC_SEND_MAX_BUFFERED_BYTES, MIC_SILENCE_PROBE_MS, PLAYBACK_BUFFER_SECONDS, PLAYBACK_CONCEAL_FADE_MS, PLAYBACK_CONCEAL_FLOOR, PLAYBACK_DONE_MAX_WAIT_MS, PLAYBACK_DONE_POLL_MS, PLAYBACK_JITTER_MS, PLAYBACK_REFILL_MS } from "@alexkroman1/aai";
 //#region types.ts
 /**
 * `getUserMedia` audio constraints for every capture path in this package.
@@ -29,4 +29,4 @@ const VOICE_CAPTURE_CONSTRAINTS = {
 	voiceIsolation: false
 };
 //#endregion
-export { FILE_SEND_BACKOFF_MS, MIC_BUFFER_SECONDS, MIC_SEND_MAX_BUFFERED_BYTES, MIC_SILENCE_PROBE_MS, PLAYBACK_CONCEAL_FADE_MS, PLAYBACK_CONCEAL_FLOOR, PLAYBACK_JITTER_MS, PLAYBACK_REFILL_MS, VOICE_CAPTURE_CONSTRAINTS };
+export { CAPTURE_STOP_ACK_TIMEOUT_MS, DEFAULT_STT_SAMPLE_RATE, MIC_BUFFER_SECONDS, MIC_SEND_MAX_BUFFERED_BYTES, MIC_SILENCE_PROBE_MS, PLAYBACK_BUFFER_SECONDS, PLAYBACK_CONCEAL_FADE_MS, PLAYBACK_CONCEAL_FLOOR, PLAYBACK_DONE_MAX_WAIT_MS, PLAYBACK_DONE_POLL_MS, PLAYBACK_JITTER_MS, PLAYBACK_REFILL_MS, VOICE_CAPTURE_CONSTRAINTS };

package/dist/worklets/_module-url.d.ts

@@ -0,0 +1,10 @@
+/**
+ * Blob-URL module for an inlined AudioWorklet processor source.
+ *
+ * A blob URL rather than a data URI because the agent page's CSP allows
+ * `script-src blob:` but not `data:` — a data-URI module fails `addModule`
+ * with the opaque "Unable to load a worklet's module". Every inline worklet
+ * in this package must load through this helper so that constraint lives in
+ * one place.
+ */
+export declare function workletModuleUrl(source: string): string;

package/dist/worklets/capture-processor.d.ts

@@ -1,4 +1,4 @@
 /** Raw worklet source — exported so tests can evaluate the processor directly. */
 export declare const captureProcessorSource = "\nclass CaptureProcessor extends AudioWorkletProcessor {\n  constructor(options) {\n    super();\n    this.recording = false;\n    const opts = options.processorOptions || {};\n    // The context runs at the STT rate, so this is both the input and the\n    // output rate \u2014 there is nothing to convert.\n    this.rate = opts.sampleRate || sampleRate;\n    // Int16 accumulation buffer: flushed to the main thread as one transferred\n    // ArrayBuffer once ~bufferSeconds of samples are batched. Sized 2x the\n    // flush target so a whole render quantum always fits before flushing.\n    this.targetSamples = Math.max(1, Math.round(this.rate * (opts.bufferSeconds || 0.1)));\n    this.pending = new Int16Array(this.targetSamples * 2);\n    this.pendingLen = 0;\n    // Dead-mic probe: samples left to inspect before concluding the device\n    // delivers nothing but digital silence. Only consumed while recording, so\n    // the cost disappears after the window (or after the first real sample).\n    this.probeSamplesLeft = Math.round(\n      (this.rate * (opts.silenceProbeMs ?? 1500)) / 1000,\n    );\n    this.port.onmessage = (e) => {\n      if (e.data.event === 'start') this.recording = true;\n      else if (e.data.event === 'stop') {\n        // Final flush so the tail of speech isn't dropped on close, then ack\n        // so the host knows the tail chunk (if any) has been posted and it is\n        // safe to tear the context down.\n        this.flush();\n        this.recording = false;\n        this.port.postMessage({ event: 'stopped' });\n      }\n    };\n  }\n\n  // Convert Float32 -> Int16 and append to the pending batch. Writes through\n  // an Int16Array directly (assignment truncates like DataView.setInt16).\n  accumulate(samples) {\n    let buf = this.pending;\n    if (this.pendingLen + samples.length > buf.length) {\n      // Defensive: only reachable if a render quantum outproduces the 1x\n      // headroom above the flush target (never with 128-sample quanta).\n      const grown = new Int16Array((this.pendingLen + samples.length) * 2);\n      grown.set(buf.subarray(0, this.pendingLen));\n      this.pending = grown;\n      buf = grown;\n    }\n    for (let i = 0; i < samples.length; i++) {\n      const s = Math.max(-1, Math.min(1, samples[i]));\n      buf[this.pendingLen++] = s < 0 ? s * 0x8000 : s * 0x7fff;\n    }\n  }\n\n  // Post the batched samples as one transferred ArrayBuffer and reset.\n  flush() {\n    if (this.pendingLen === 0) return;\n    const buffer = this.pending.buffer.slice(0, this.pendingLen * 2);\n    this.pendingLen = 0;\n    this.port.postMessage({ event: 'chunk', buffer }, [buffer]);\n  }\n\n  // Watch the first window of input for any nonzero sample. One is enough to\n  // prove the device is live \u2014 a real mic in a quiet room still carries a\n  // noise floor, so all-zeros means muted, wrong input, or no input at all.\n  probeForSilence(channel) {\n    for (let i = 0; i < channel.length; i++) {\n      if (channel[i] !== 0) {\n        this.probeSamplesLeft = 0;\n        return;\n      }\n    }\n    this.probeSamplesLeft -= channel.length;\n    if (this.probeSamplesLeft <= 0) {\n      this.port.postMessage({ event: 'silent' });\n    }\n  }\n\n  process(inputs) {\n    const input = inputs[0];\n    if (!input || !input[0] || !this.recording) return true;\n\n    if (this.probeSamplesLeft > 0) this.probeForSilence(input[0]);\n\n    this.accumulate(input[0]);\n    if (this.pendingLen >= this.targetSamples) this.flush();\n    return true;\n  }\n}\n\nregisterProcessor('capture-processor', CaptureProcessor);\n";
-declare const src: string;
-export default src;
+declare const _default: string;
+export default _default;

package/dist/worklets/capture-processor.js

@@ -1,4 +1,5 @@
-import { MIC_SILENCE_PROBE_MS } from "../types.js";
+import { MIC_BUFFER_SECONDS, MIC_SILENCE_PROBE_MS } from "../types.js";
+import { t as workletModuleUrl } from "../_module-url-C_4gRVL0.js";
 //#region worklets/capture-processor.ts
 const CaptureProcessorWorklet = `
 class CaptureProcessor extends AudioWorkletProcessor {
@@ -12,7 +13,7 @@ class CaptureProcessor extends AudioWorkletProcessor {
     // Int16 accumulation buffer: flushed to the main thread as one transferred
     // ArrayBuffer once ~bufferSeconds of samples are batched. Sized 2x the
     // flush target so a whole render quantum always fits before flushing.
-    this.targetSamples = Math.max(1, Math.round(this.rate * (opts.bufferSeconds || 0.1)));
+    this.targetSamples = Math.max(1, Math.round(this.rate * (opts.bufferSeconds || ${MIC_BUFFER_SECONDS})));
     this.pending = new Int16Array(this.targetSamples * 2);
     this.pendingLen = 0;
     // Dead-mic probe: samples left to inspect before concluding the device
@@ -92,7 +93,6 @@ registerProcessor('capture-processor', CaptureProcessor);
 `;
 /** Raw worklet source — exported so tests can evaluate the processor directly. */
 const captureProcessorSource = CaptureProcessorWorklet;
-const script = new Blob([CaptureProcessorWorklet], { type: "application/javascript" });
-const src = URL.createObjectURL(script);
+var capture_processor_default = workletModuleUrl(CaptureProcessorWorklet);
 //#endregion
-export { captureProcessorSource, src as default };
+export { captureProcessorSource, capture_processor_default as default };

package/dist/worklets/playback-processor.d.ts

@@ -1,4 +1,4 @@
 /** Raw worklet source — exported so tests can evaluate the processor directly. */
-export declare const playbackProcessorSource = "\nclass PlaybackProcessor extends AudioWorkletProcessor {\n  constructor(options) {\n    super();\n    const opts = options.processorOptions || {};\n    const rate = opts.sampleRate ?? 24000;\n    // Fill target for the start of a turn. If 'done' arrives first (short\n    // utterance), start immediately instead of waiting for audio that is\n    // never coming.\n    this.jitterSamples = Math.floor((rate * (opts.jitterMs ?? 400)) / 1000);\n    // Fill target after an underrun \u2014 see PLAYBACK_REFILL_MS.\n    this.refillSamples = Math.floor((rate * (opts.refillMs ?? 200)) / 1000);\n    // Concealment source: a ring of the most recently played samples, looped\n    // under a decaying gain to cover a gap. Sized to the fade window, with a\n    // per-sample decay that reaches the floor exactly at its end.\n    this.concealCapacity = Math.max(1, Math.floor((rate * 40) / 1000));\n    this.concealBuf = new Float32Array(this.concealCapacity);\n    this.concealDecay = Math.exp(Math.log(0.001) / this.concealCapacity);\n    // Float32 ring buffer \u2014 60s at the context sample rate. Allocated once for\n    // the node's lifetime; per-turn state resets via resetTurn(). writePos and\n    // readPos are absolute (monotonic) sample counts; the buffer is indexed\n    // modulo capacity so a reply longer than 60s keeps playing instead of\n    // writing past the end and going silent.\n    this.capacity = rate * 60;\n    this.samples = new Float32Array(this.capacity);\n    // Platform endianness probe: the wire format is PCM16 little-endian, so\n    // the Int16Array fast path in ingestBytes is only valid on LE hosts\n    // (every shipping browser target; the DataView path is the fallback).\n    this.littleEndian = new Uint8Array(new Uint16Array([1]).buffer)[0] === 1;\n    this.resetTurn();\n\n    this.port.onmessage = (e) => {\n      const d = e.data;\n      if (d.event === 'write') {\n        this.ingestBytes(d.buffer);\n      } else if (d.event === 'interrupt') {\n        this.interrupted = true;\n      } else if (d.event === 'done') {\n        this.isDone = true;\n      }\n    };\n  }\n\n  // Reset per-turn state so the node is reusable across replies without\n  // reallocating the sample buffer or re-instantiating the worklet.\n  resetTurn() {\n    this.interrupted = false;\n    this.isDone = false;\n    this.playing = false;\n    // Whether any real audio has been rendered this turn. Separates a turn's\n    // pre-roll (nothing to extrapolate from, and not a defect) from a\n    // mid-turn underrun.\n    this.hasPlayed = false;\n    this.fillTarget = this.jitterSamples;\n    // Carry-over byte for split samples across chunks\n    this.carry = null;\n    this.writePos = 0;\n    this.readPos = 0;\n    // Concealment ring state and the current fade position.\n    this.concealLen = 0;\n    this.concealWrite = 0;\n    this.concealPos = 0;\n    this.concealGain = 1;\n    // Episode flags, so a multi-quantum gap counts as one event.\n    this.concealing = false;\n    this.concealedSilence = false;\n    // Reported to the host on 'stop'. A fresh object per turn: the one just\n    // posted must not be mutated by the next turn.\n    this.stats = {\n      concealedSamples: 0,\n      silentConcealedSamples: 0,\n      concealmentEvents: 0,\n      silentConcealmentEvents: 0,\n    };\n  }\n\n  // End the current turn: notify the host and rearm for the next reply.\n  // Must NOT return false from process() \u2014 a processor that stops is dead\n  // for good, forcing a new node (and buffer) per reply.\n  // `reason` ('interrupt' | 'done') tells the host which turn boundary this\n  // stop belongs to: interrupt-stops are dropped host-side (flush() already\n  // settled that turn), so they can never resolve a later turn's done() early.\n  stopTurn(reason) {\n    this.port.postMessage({ event: 'stop', reason, stats: this.stats });\n    this.resetTurn();\n  }\n\n  // Cover a quantum (from `start`) where real audio should have been.\n  //\n  // Before the turn's first samples there is nothing to extrapolate from, so\n  // the gap is plain silence and counted as nothing \u2014 WebRTC likewise only\n  // counts concealment once playout has begun. After that, loop the retained\n  // tail under a decaying gain: a hard zero-fill is a discontinuity mid-word,\n  // which is the click that makes a brief stall sound like breakage.\n  coverGap(out, start) {\n    if (!this.hasPlayed) {\n      out.fill(0, start);\n      return;\n    }\n    if (!this.concealing) {\n      this.concealing = true;\n      this.concealedSilence = false;\n      this.stats.concealmentEvents++;\n    }\n    const total = out.length - start;\n    const len = this.concealLen;\n    let silent = 0;\n    if (len === 0) {\n      out.fill(0, start);\n      silent = total;\n    } else {\n      let g = this.concealGain;\n      for (let i = start; i < out.length; i++) {\n        if (g < 0.001) {\n          // The fade has run out: keep counting the gap, but stop looping a\n          // fragment that is now inaudible anyway.\n          out[i] = 0;\n          silent++;\n          continue;\n        }\n        out[i] = this.concealBuf[this.concealPos] * g;\n        this.concealPos = this.concealPos + 1 === len ? 0 : this.concealPos + 1;\n        g *= this.concealDecay;\n      }\n      this.concealGain = g;\n    }\n    this.stats.concealedSamples += total;\n    if (silent > 0) {\n      this.stats.silentConcealedSamples += silent;\n      if (!this.concealedSilence) {\n        this.concealedSilence = true;\n        this.stats.silentConcealmentEvents++;\n      }\n    }\n  }\n\n  // Retain the tail of a rendered quantum as the next gap's concealment\n  // source, and close any episode the real audio just ended.\n  rememberTail(out, n) {\n    const cap = this.concealCapacity;\n    const take = Math.min(n, cap);\n    for (let i = n - take; i < n; i++) {\n      this.concealBuf[this.concealWrite] = out[i];\n      this.concealWrite = this.concealWrite + 1 === cap ? 0 : this.concealWrite + 1;\n    }\n    this.concealLen = Math.min(cap, this.concealLen + take);\n    // Read the loop oldest-first; once the ring is full the write cursor is\n    // the oldest retained sample.\n    this.concealPos = this.concealLen === cap ? this.concealWrite : 0;\n    this.concealing = false;\n    this.concealGain = 1;\n  }\n\n  ingestBytes(uint8) {\n    let bytes = uint8;\n\n    if (this.carry !== null) {\n      const merged = new Uint8Array(1 + bytes.length);\n      merged[0] = this.carry;\n      merged.set(bytes, 1);\n      bytes = merged;\n      this.carry = null;\n    }\n\n    if (bytes.length % 2 !== 0) {\n      this.carry = bytes[bytes.length - 1];\n      bytes = bytes.subarray(0, bytes.length - 1);\n    }\n\n    if (bytes.length === 0) return;\n    const numSamples = bytes.length / 2;\n    const cap = this.capacity;\n    const samples = this.samples;\n    if (this.littleEndian && (bytes.byteOffset & 1) === 0) {\n      // Fast path: 2-byte-aligned LE bytes wrap directly as an Int16Array;\n      // copy wrap-aware in at most two runs with no per-sample DataView call\n      // or modulo. This runs on the realtime audio thread.\n      const int16 = new Int16Array(bytes.buffer, bytes.byteOffset, numSamples);\n      let src = 0;\n      let dst = this.writePos % cap;\n      while (src < numSamples) {\n        const run = Math.min(numSamples - src, cap - dst);\n        for (let j = 0; j < run; j++) {\n          samples[dst + j] = int16[src + j] / 0x8000;\n        }\n        src += run;\n        dst = 0;\n      }\n    } else {\n      // Odd byte offset (or big-endian host): fall back to per-sample reads.\n      const view = new DataView(bytes.buffer, bytes.byteOffset, bytes.length);\n      let dst = this.writePos % cap;\n      for (let i = 0; i < numSamples; i++) {\n        samples[dst] = view.getInt16(i * 2, true) / 0x8000;\n        dst++;\n        if (dst === cap) dst = 0;\n      }\n    }\n    this.writePos += numSamples;\n    // If the producer outran the consumer by more than the buffer holds, drop\n    // the oldest unplayed audio rather than reading samples we've overwritten.\n    if (this.writePos - this.readPos > this.capacity) {\n      this.readPos = this.writePos - this.capacity;\n    }\n  }\n\n  process(inputs, outputs) {\n    // No output wired up yet \u2014 nothing to render this quantum. Throwing here\n    // would permanently kill the processor (the node is persistent per\n    // session), so guard like the capture processor does.\n    if (!outputs[0] || !outputs[0][0]) return true;\n    const out = outputs[0][0];\n    if (this.interrupted) {\n      out.fill(0);\n      this.stopTurn('interrupt');\n      return true;\n    }\n\n    const avail = this.writePos - this.readPos;\n\n    // Filling: wait for the target. 'done' short-circuits it \u2014 what is\n    // buffered is all there will be, so there is nothing left to wait for.\n    if (!this.playing) {\n      if (avail >= this.fillTarget || this.isDone) {\n        this.playing = true;\n      } else {\n        this.coverGap(out, 0);\n        return true;\n      }\n    }\n\n    // Underrun: this quantum cannot be filled and more audio is still coming.\n    // Go back to filling (at the refill target) and cover the gap, leaving\n    // readPos untouched \u2014 the fragment stays buffered and plays intact once\n    // the buffer recovers, instead of being dribbled out a few samples at a\n    // time for the rest of the turn.\n    if (avail < out.length && !this.isDone) {\n      this.playing = false;\n      this.fillTarget = this.refillSamples;\n      this.coverGap(out, 0);\n      return true;\n    }\n\n    if (avail > 0) {\n      const n = Math.min(avail, out.length);\n      // Copy from the ring buffer, splitting across the wrap boundary.\n      const start = this.readPos % this.capacity;\n      const first = Math.min(n, this.capacity - start);\n      out.set(this.samples.subarray(start, start + first), 0);\n      if (n > first) out.set(this.samples.subarray(0, n - first), first);\n      this.readPos += n;\n      // Only reachable with n < out.length on the turn's final partial\n      // quantum (the underrun branch above catches every other case).\n      out.fill(0, n);\n      this.hasPlayed = true;\n      this.rememberTail(out, n);\n      return true;\n    }\n\n    // Drained and done: end the turn. Not reachable mid-turn \u2014 an empty\n    // buffer with audio still coming is the underrun branch above.\n    out.fill(0);\n    if (this.isDone) {\n      this.stopTurn('done');\n    }\n    return true;\n  }\n}\n\nregisterProcessor('playback-processor', PlaybackProcessor);\n";
-declare const src: string;
-export default src;
+export declare const playbackProcessorSource = "\nclass PlaybackProcessor extends AudioWorkletProcessor {\n  constructor(options) {\n    super();\n    const opts = options.processorOptions || {};\n    // The node's context IS the playback context (audio.ts asserts its rate),\n    // so the worklet-global sampleRate is authoritative; the option exists\n    // for the node-less test harness.\n    const rate = opts.sampleRate ?? sampleRate;\n    // Fill target for the start of a turn. If 'done' arrives first (short\n    // utterance), start immediately instead of waiting for audio that is\n    // never coming.\n    this.jitterSamples = Math.floor((rate * (opts.jitterMs ?? 400)) / 1000);\n    // Fill target after an underrun \u2014 see PLAYBACK_REFILL_MS.\n    this.refillSamples = Math.floor((rate * (opts.refillMs ?? 200)) / 1000);\n    // Concealment source: a ring of the most recently played samples, looped\n    // under a decaying gain to cover a gap. Sized to the fade window, with a\n    // per-sample decay that reaches the floor exactly at its end.\n    this.concealCapacity = Math.max(1, Math.floor((rate * 40) / 1000));\n    this.concealBuf = new Float32Array(this.concealCapacity);\n    this.concealDecay = Math.exp(Math.log(0.001) / this.concealCapacity);\n    // Float32 ring buffer \u2014 PLAYBACK_BUFFER_SECONDS at the context sample\n    // rate. Allocated once for the node's lifetime; per-turn state resets via\n    // resetTurn(). writePos and readPos are absolute (monotonic) sample\n    // counts; the buffer is indexed modulo capacity so a longer reply keeps\n    // playing instead of writing past the end and going silent.\n    this.capacity = rate * 60;\n    this.samples = new Float32Array(this.capacity);\n    // Platform endianness probe: the wire format is PCM16 little-endian, so\n    // the Int16Array fast path in ingestBytes is only valid on LE hosts\n    // (every shipping browser target; the DataView path is the fallback).\n    this.littleEndian = new Uint8Array(new Uint16Array([1]).buffer)[0] === 1;\n    this.resetTurn();\n\n    this.port.onmessage = (e) => {\n      const d = e.data;\n      if (d.event === 'write') {\n        this.ingestBytes(d.buffer);\n      } else if (d.event === 'interrupt') {\n        // Applied eagerly, not deferred to the next process(): onmessage and\n        // process() never interleave (one audio thread), and a deferred\n        // interrupt would let 'write'/'done' frames for the NEXT turn\n        // coalesce in ahead of it and be wiped by the reset along with the\n        // cancelled turn's audio.\n        this.stopTurn('interrupt');\n      } else if (d.event === 'done') {\n        this.isDone = true;\n      }\n    };\n  }\n\n  // Reset per-turn state so the node is reusable across replies without\n  // reallocating the sample buffer or re-instantiating the worklet.\n  resetTurn() {\n    this.isDone = false;\n    this.playing = false;\n    // Whether any real audio has been rendered this turn. Separates a turn's\n    // pre-roll (nothing to extrapolate from, and not a defect) from a\n    // mid-turn underrun.\n    this.hasPlayed = false;\n    this.fillTarget = this.jitterSamples;\n    // Carry-over byte for split samples across chunks\n    this.carry = null;\n    this.writePos = 0;\n    this.readPos = 0;\n    // Concealment ring state and the current fade position.\n    this.concealLen = 0;\n    this.concealWrite = 0;\n    this.concealPos = 0;\n    this.concealGain = 1;\n    // Episode flags, so a multi-quantum gap counts as one event.\n    this.concealing = false;\n    this.concealedSilence = false;\n    // Reported to the host on 'stop'. A fresh object per turn: the one just\n    // posted must not be mutated by the next turn.\n    this.stats = {\n      concealedSamples: 0,\n      silentConcealedSamples: 0,\n      concealmentEvents: 0,\n      silentConcealmentEvents: 0,\n    };\n  }\n\n  // End the current turn: notify the host and rearm for the next reply.\n  // Must NOT return false from process() \u2014 a processor that stops is dead\n  // for good, forcing a new node (and buffer) per reply.\n  // `reason` ('interrupt' | 'done') tells the host which turn boundary this\n  // stop belongs to: interrupt-stops are dropped host-side (flush() already\n  // settled that turn), so they can never resolve a later turn's done() early.\n  stopTurn(reason) {\n    this.port.postMessage({ event: 'stop', reason, stats: this.stats });\n    this.resetTurn();\n  }\n\n  // Cover a quantum (from `start`) where real audio should have been.\n  //\n  // Before the turn's first samples there is nothing to extrapolate from, so\n  // the gap is plain silence and counted as nothing \u2014 WebRTC likewise only\n  // counts concealment once playout has begun. After that, loop the retained\n  // tail under a decaying gain: a hard zero-fill is a discontinuity mid-word,\n  // which is the click that makes a brief stall sound like breakage.\n  coverGap(out, start) {\n    if (!this.hasPlayed) {\n      out.fill(0, start);\n      return;\n    }\n    if (!this.concealing) {\n      this.concealing = true;\n      this.concealedSilence = false;\n      this.stats.concealmentEvents++;\n    }\n    const total = out.length - start;\n    const len = this.concealLen;\n    let silent = 0;\n    // Second condition: once the fade has decayed to the floor, every sample\n    // the loop would emit is 0 anyway \u2014 bulk-fill instead of running 128\n    // branchy iterations per quantum for the whole tail of a long stall.\n    if (len === 0 || this.concealGain < 0.001) {\n      out.fill(0, start);\n      silent = total;\n    } else {\n      let g = this.concealGain;\n      for (let i = start; i < out.length; i++) {\n        if (g < 0.001) {\n          // The fade has run out: keep counting the gap, but stop looping a\n          // fragment that is now inaudible anyway.\n          out[i] = 0;\n          silent++;\n          continue;\n        }\n        out[i] = this.concealBuf[this.concealPos] * g;\n        this.concealPos = this.concealPos + 1 === len ? 0 : this.concealPos + 1;\n        g *= this.concealDecay;\n      }\n      this.concealGain = g;\n    }\n    this.stats.concealedSamples += total;\n    if (silent > 0) {\n      this.stats.silentConcealedSamples += silent;\n      if (!this.concealedSilence) {\n        this.concealedSilence = true;\n        this.stats.silentConcealmentEvents++;\n      }\n    }\n  }\n\n  // Retain the tail of a rendered quantum as the next gap's concealment\n  // source, and close any episode the real audio just ended.\n  rememberTail(out, n) {\n    const cap = this.concealCapacity;\n    const take = Math.min(n, cap);\n    // Bulk copies (this runs on every cleanly rendered quantum): the tail is\n    // one contiguous source run, landing in at most two ring runs.\n    const tail = out.subarray(n - take, n);\n    const first = Math.min(take, cap - this.concealWrite);\n    this.concealBuf.set(tail.subarray(0, first), this.concealWrite);\n    if (take > first) this.concealBuf.set(tail.subarray(first), 0);\n    this.concealWrite = (this.concealWrite + take) % cap;\n    this.concealLen = Math.min(cap, this.concealLen + take);\n    // Read the loop oldest-first; once the ring is full the write cursor is\n    // the oldest retained sample.\n    this.concealPos = this.concealLen === cap ? this.concealWrite : 0;\n    this.concealing = false;\n    this.concealGain = 1;\n  }\n\n  ingestBytes(uint8) {\n    let bytes = uint8;\n\n    if (this.carry !== null) {\n      const merged = new Uint8Array(1 + bytes.length);\n      merged[0] = this.carry;\n      merged.set(bytes, 1);\n      bytes = merged;\n      this.carry = null;\n    }\n\n    if (bytes.length % 2 !== 0) {\n      this.carry = bytes[bytes.length - 1];\n      bytes = bytes.subarray(0, bytes.length - 1);\n    }\n\n    if (bytes.length === 0) return;\n    const numSamples = bytes.length / 2;\n    const cap = this.capacity;\n    const samples = this.samples;\n    if (this.littleEndian && (bytes.byteOffset & 1) === 0) {\n      // Fast path: 2-byte-aligned LE bytes wrap directly as an Int16Array;\n      // copy wrap-aware in at most two runs with no per-sample DataView call\n      // or modulo. This runs on the realtime audio thread.\n      const int16 = new Int16Array(bytes.buffer, bytes.byteOffset, numSamples);\n      let src = 0;\n      let dst = this.writePos % cap;\n      while (src < numSamples) {\n        const run = Math.min(numSamples - src, cap - dst);\n        for (let j = 0; j < run; j++) {\n          samples[dst + j] = int16[src + j] / 0x8000;\n        }\n        src += run;\n        dst = 0;\n      }\n    } else {\n      // Odd byte offset (or big-endian host): fall back to per-sample reads.\n      const view = new DataView(bytes.buffer, bytes.byteOffset, bytes.length);\n      let dst = this.writePos % cap;\n      for (let i = 0; i < numSamples; i++) {\n        samples[dst] = view.getInt16(i * 2, true) / 0x8000;\n        dst++;\n        if (dst === cap) dst = 0;\n      }\n    }\n    this.writePos += numSamples;\n    // If the producer outran the consumer by more than the buffer holds, drop\n    // the oldest unplayed audio rather than reading samples we've overwritten.\n    if (this.writePos - this.readPos > this.capacity) {\n      this.readPos = this.writePos - this.capacity;\n    }\n  }\n\n  process(inputs, outputs) {\n    // No output wired up yet \u2014 nothing to render this quantum. Throwing here\n    // would permanently kill the processor (the node is persistent per\n    // session), so guard like the capture processor does.\n    if (!outputs[0] || !outputs[0][0]) return true;\n    const out = outputs[0][0];\n    const avail = this.writePos - this.readPos;\n\n    // Filling: wait for the target. 'done' short-circuits it \u2014 what is\n    // buffered is all there will be, so there is nothing left to wait for.\n    if (!this.playing) {\n      if (avail >= this.fillTarget || this.isDone) {\n        this.playing = true;\n      } else {\n        this.coverGap(out, 0);\n        return true;\n      }\n    }\n\n    // Underrun: this quantum cannot be filled and more audio is still coming.\n    // Go back to filling (at the refill target) and cover the gap, leaving\n    // readPos untouched \u2014 the fragment stays buffered and plays intact once\n    // the buffer recovers, instead of being dribbled out a few samples at a\n    // time for the rest of the turn.\n    if (avail < out.length && !this.isDone) {\n      this.playing = false;\n      this.fillTarget = this.refillSamples;\n      this.coverGap(out, 0);\n      return true;\n    }\n\n    if (avail > 0) {\n      const n = Math.min(avail, out.length);\n      // Copy from the ring buffer, splitting across the wrap boundary.\n      const start = this.readPos % this.capacity;\n      const first = Math.min(n, this.capacity - start);\n      out.set(this.samples.subarray(start, start + first), 0);\n      if (n > first) out.set(this.samples.subarray(0, n - first), first);\n      this.readPos += n;\n      // Only reachable with n < out.length on the turn's final partial\n      // quantum (the underrun branch above catches every other case).\n      out.fill(0, n);\n      this.hasPlayed = true;\n      this.rememberTail(out, n);\n      return true;\n    }\n\n    // Drained and done: end the turn. Not reachable mid-turn \u2014 an empty\n    // buffer with audio still coming is the underrun branch above.\n    out.fill(0);\n    if (this.isDone) {\n      this.stopTurn('done');\n    }\n    return true;\n  }\n}\n\nregisterProcessor('playback-processor', PlaybackProcessor);\n";
+declare const _default: string;
+export default _default;

package/dist/worklets/playback-processor.js

@@ -1,11 +1,15 @@
-import { PLAYBACK_CONCEAL_FADE_MS, PLAYBACK_CONCEAL_FLOOR, PLAYBACK_JITTER_MS, PLAYBACK_REFILL_MS } from "../types.js";
+import { PLAYBACK_BUFFER_SECONDS, PLAYBACK_CONCEAL_FADE_MS, PLAYBACK_CONCEAL_FLOOR, PLAYBACK_JITTER_MS, PLAYBACK_REFILL_MS } from "../types.js";
+import { t as workletModuleUrl } from "../_module-url-C_4gRVL0.js";
 //#region worklets/playback-processor.ts
 const PlaybackProcessorWorklet = `
 class PlaybackProcessor extends AudioWorkletProcessor {
   constructor(options) {
     super();
     const opts = options.processorOptions || {};
-    const rate = opts.sampleRate ?? 24000;
+    // The node's context IS the playback context (audio.ts asserts its rate),
+    // so the worklet-global sampleRate is authoritative; the option exists
+    // for the node-less test harness.
+    const rate = opts.sampleRate ?? sampleRate;
     // Fill target for the start of a turn. If 'done' arrives first (short
     // utterance), start immediately instead of waiting for audio that is
     // never coming.
@@ -18,12 +22,12 @@ class PlaybackProcessor extends AudioWorkletProcessor {
     this.concealCapacity = Math.max(1, Math.floor((rate * ${PLAYBACK_CONCEAL_FADE_MS}) / 1000));
     this.concealBuf = new Float32Array(this.concealCapacity);
     this.concealDecay = Math.exp(Math.log(${PLAYBACK_CONCEAL_FLOOR}) / this.concealCapacity);
-    // Float32 ring buffer — 60s at the context sample rate. Allocated once for
-    // the node's lifetime; per-turn state resets via resetTurn(). writePos and
-    // readPos are absolute (monotonic) sample counts; the buffer is indexed
-    // modulo capacity so a reply longer than 60s keeps playing instead of
-    // writing past the end and going silent.
-    this.capacity = rate * 60;
+    // Float32 ring buffer — PLAYBACK_BUFFER_SECONDS at the context sample
+    // rate. Allocated once for the node's lifetime; per-turn state resets via
+    // resetTurn(). writePos and readPos are absolute (monotonic) sample
+    // counts; the buffer is indexed modulo capacity so a longer reply keeps
+    // playing instead of writing past the end and going silent.
+    this.capacity = rate * ${PLAYBACK_BUFFER_SECONDS};
     this.samples = new Float32Array(this.capacity);
     // Platform endianness probe: the wire format is PCM16 little-endian, so
     // the Int16Array fast path in ingestBytes is only valid on LE hosts
@@ -36,7 +40,12 @@ class PlaybackProcessor extends AudioWorkletProcessor {
       if (d.event === 'write') {
         this.ingestBytes(d.buffer);
       } else if (d.event === 'interrupt') {
-        this.interrupted = true;
+        // Applied eagerly, not deferred to the next process(): onmessage and
+        // process() never interleave (one audio thread), and a deferred
+        // interrupt would let 'write'/'done' frames for the NEXT turn
+        // coalesce in ahead of it and be wiped by the reset along with the
+        // cancelled turn's audio.
+        this.stopTurn('interrupt');
       } else if (d.event === 'done') {
         this.isDone = true;
       }
@@ -46,7 +55,6 @@ class PlaybackProcessor extends AudioWorkletProcessor {
   // Reset per-turn state so the node is reusable across replies without
   // reallocating the sample buffer or re-instantiating the worklet.
   resetTurn() {
-    this.interrupted = false;
     this.isDone = false;
     this.playing = false;
     // Whether any real audio has been rendered this turn. Separates a turn's
@@ -107,7 +115,10 @@ class PlaybackProcessor extends AudioWorkletProcessor {
     const total = out.length - start;
     const len = this.concealLen;
     let silent = 0;
-    if (len === 0) {
+    // Second condition: once the fade has decayed to the floor, every sample
+    // the loop would emit is 0 anyway — bulk-fill instead of running 128
+    // branchy iterations per quantum for the whole tail of a long stall.
+    if (len === 0 || this.concealGain < ${PLAYBACK_CONCEAL_FLOOR}) {
       out.fill(0, start);
       silent = total;
     } else {
@@ -141,10 +152,13 @@ class PlaybackProcessor extends AudioWorkletProcessor {
   rememberTail(out, n) {
     const cap = this.concealCapacity;
     const take = Math.min(n, cap);
-    for (let i = n - take; i < n; i++) {
-      this.concealBuf[this.concealWrite] = out[i];
-      this.concealWrite = this.concealWrite + 1 === cap ? 0 : this.concealWrite + 1;
-    }
+    // Bulk copies (this runs on every cleanly rendered quantum): the tail is
+    // one contiguous source run, landing in at most two ring runs.
+    const tail = out.subarray(n - take, n);
+    const first = Math.min(take, cap - this.concealWrite);
+    this.concealBuf.set(tail.subarray(0, first), this.concealWrite);
+    if (take > first) this.concealBuf.set(tail.subarray(first), 0);
+    this.concealWrite = (this.concealWrite + take) % cap;
     this.concealLen = Math.min(cap, this.concealLen + take);
     // Read the loop oldest-first; once the ring is full the write cursor is
     // the oldest retained sample.
@@ -212,12 +226,6 @@ class PlaybackProcessor extends AudioWorkletProcessor {
     // session), so guard like the capture processor does.
     if (!outputs[0] || !outputs[0][0]) return true;
     const out = outputs[0][0];
-    if (this.interrupted) {
-      out.fill(0);
-      this.stopTurn('interrupt');
-      return true;
-    }
-
     const avail = this.writePos - this.readPos;
 
     // Filling: wait for the target. 'done' short-circuits it — what is
@@ -273,7 +281,6 @@ registerProcessor('playback-processor', PlaybackProcessor);
 `;
 /** Raw worklet source — exported so tests can evaluate the processor directly. */
 const playbackProcessorSource = PlaybackProcessorWorklet;
-const script = new Blob([PlaybackProcessorWorklet], { type: "application/javascript" });
-const src = URL.createObjectURL(script);
+var playback_processor_default = workletModuleUrl(PlaybackProcessorWorklet);
 //#endregion
-export { src as default, playbackProcessorSource };
+export { playback_processor_default as default, playbackProcessorSource };

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@alexkroman1/aai-ui",
-  "version": "2.0.0",
+  "version": "3.0.0",
   "type": "module",
   "files": [
     "dist",
@@ -20,7 +20,7 @@
     "clsx": "^2.1.1",
     "partysocket": "^1.3.0",
     "use-sync-external-store": "^1.6.0",
-    "@alexkroman1/aai": "2.0.0"
+    "@alexkroman1/aai": "3.0.0"
   },
   "peerDependencies": {
     "react": "^19.0.0",

package/dist/components/workflow-view.d.ts

@@ -1,13 +0,0 @@
-/**
- * Workflow run surface: push-to-talk or audio upload stages one instruction
- * clip; **Go** runs it as a single sync turn and shows what was heard plus
- * the tool calls that executed — no greeting, no assistant messages.
- *
- * @public
- */
-export declare function WorkflowView({ syncUrl, title, }: {
-    /** The workflow server's sync endpoint, e.g. `https://host/slug/sync`. */
-    syncUrl: string;
-    /** Workflow name shown in the header. */
-    title?: string | undefined;
-}): import("react").JSX.Element;

package/dist/default-client/assets/audio-Cgviqo9t.js

@@ -1 +0,0 @@
-import{n as e}from"./rolldown-runtime-BpQH8Ho1.js";import{d as t,t as n}from"./types-Bpg3ZIZK.js";var r=e({createVoiceIO:()=>l,decodeAudioToPcm16:()=>s}),i=1e3,a=65e3,o=250;async function s(e,t){let n=await new OfflineAudioContext(1,1,t).decodeAudioData(e),r=Math.ceil(n.duration*t),i=new OfflineAudioContext(1,r,t),a=i.createBufferSource();a.buffer=n,a.connect(i.destination),a.start();let o=(await i.startRendering()).getChannelData(0),s=new Int16Array(o.length),c=0;for(let e of o){let t=Math.max(-1,Math.min(1,e));s[c++]=t<0?t*32768:t*32767}return s}function c(e,t,n){if(e!==t)throw Error(`Browser refused the ${n} sample rate: asked for ${t} Hz, got ${e} Hz`)}async function l(e){let{sttSampleRate:r,ttsSampleRate:s,captureWorkletSrc:l,playbackWorkletSrc:u,onMicData:d,onError:f,onPlaybackStats:p,onMicSilent:m}=e,h=new AudioContext({sampleRate:s,latencyHint:`playback`}),g=r===s,_=g?h:new AudioContext({sampleRate:r,latencyHint:`interactive`});async function v(){let e=g?[h]:[h,_];await Promise.all(e.map(e=>e.close().catch(e=>{console.warn(`AudioContext close failed:`,e)})))}let y=navigator.mediaDevices.getUserMedia({audio:{deviceId:{ideal:`default`},...n}}),b;try{[b]=await Promise.all([y,h.resume(),_.resume(),_.audioWorklet.addModule(l),h.audioWorklet.addModule(u)]),c(_.sampleRate,r,`capture`),c(h.sampleRate,s,`playback`)}catch(e){throw y.then(e=>{for(let t of e.getTracks())t.stop()}).catch(()=>{}),await v(),e}let x=_.createMediaStreamSource(b),S=new AudioWorkletNode(_,`capture-processor`,{channelCount:1,channelCountMode:`explicit`,processorOptions:{sampleRate:r,bufferSeconds:t}});x.connect(S),S.onprocessorerror=()=>{let e=Error(`Audio capture worklet crashed`);console.error(`[aai-ui]`,e.message),f?.(e)},S.port.postMessage({event:`start`});let C=null;S.port.onmessage=e=>{e.data.event===`chunk`?d(e.data.buffer):e.data.event===`silent`?m?.():e.data.event===`stopped`&&(C?.(),C=null)};let w=null,T=null,E=new AbortController;function D(){if(w)return w;let e=new AudioWorkletNode(h,`playback-processor`,{processorOptions:{sampleRate:h.sampleRate}});return e.connect(h.destination),e.port.onmessage=e=>{if(e.data.event===`stop`){let t=e.data.stats;if(t&&t.concealedSamples>0&&p?.(t),e.data.reason===`interrupt`)return;T?.(),T=null}},e.onprocessorerror=()=>{let e=Error(`Audio playback worklet crashed`);console.error(`[aai-ui]`,e.message),T?.(),T=null,f?.(e)},w=e,e}let O={enqueue(e){E.signal.aborted||e.byteLength!==0&&D().port.postMessage({event:`write`,buffer:new Uint8Array(e)},[e])},done(){return!w||h.state!==`running`?Promise.resolve():new Promise(e=>{T?.();let t=()=>{clearInterval(n),clearTimeout(r),T===t&&(T=null),e()},n=setInterval(()=>{h.state!==`running`&&t()},i),r=setTimeout(t,a);T=t,w?.port.postMessage({event:`done`})})},flush(){w&&(T?.(),T=null,w.port.postMessage({event:`interrupt`}))},async close(){if(!E.signal.aborted){E.abort(),await new Promise(e=>{let t=setTimeout(e,o);C=()=>{clearTimeout(t),e()},S.port.postMessage({event:`stop`})}),x.disconnect(),S.disconnect(),w&&w.disconnect();for(let e of b.getTracks())e.stop();await v()}},async[Symbol.asyncDispose](){await O.close()}};return O}export{s as n,r as t};

package/dist/default-client/assets/index-BgbIWnfG.css

@@ -1,2 +0,0 @@
-/*! tailwindcss v4.3.3 | MIT License | https://tailwindcss.com */
-@layer properties{@supports (((-webkit-hyphens:none)) and (not (margin-trim:inline))) or ((-moz-orient:inline) and (not (color:rgb(from red r g b)))){*,:before,:after,::backdrop{--tw-border-style:solid;--tw-leading:initial;--tw-font-weight:initial;--tw-tracking:initial;--tw-shadow:0 0 #0000;--tw-shadow-color:initial;--tw-shadow-alpha:100%;--tw-inset-shadow:0 0 #0000;--tw-inset-shadow-color:initial;--tw-inset-shadow-alpha:100%;--tw-ring-color:initial;--tw-ring-shadow:0 0 #0000;--tw-inset-ring-color:initial;--tw-inset-ring-shadow:0 0 #0000;--tw-ring-inset:initial;--tw-ring-offset-width:0px;--tw-ring-offset-color:#fff;--tw-ring-offset-shadow:0 0 #0000;--tw-blur:initial;--tw-brightness:initial;--tw-contrast:initial;--tw-grayscale:initial;--tw-hue-rotate:initial;--tw-invert:initial;--tw-opacity:initial;--tw-saturate:initial;--tw-sepia:initial;--tw-drop-shadow:initial;--tw-drop-shadow-color:initial;--tw-drop-shadow-alpha:100%;--tw-drop-shadow-size:initial;--tw-duration:initial;--tw-ease:initial}}}@layer theme{:root,:host{--font-sans:-apple-system, BlinkMacSystemFont, "Segoe UI", Roboto, "Helvetica Neue", "Noto Sans", Arial, sans-serif, "Apple Color Emoji", "Segoe UI Emoji", "Segoe UI Symbol", "Noto Color Emoji";--font-mono:ui-monospace, SFMono-Regular, Menlo, Monaco, Consolas, "Liberation Mono", "Courier New", monospace;--spacing:.25rem;--text-xs:.75rem;--text-xs--line-height:calc(1 / .75);--text-sm:.875rem;--text-sm--line-height:calc(1.25 / .875);--font-weight-normal:400;--font-weight-medium:500;--tracking-wide:.025em;--radius-md:.375rem;--radius-lg:.5rem;--radius-xl:.75rem;--ease-in-out:cubic-bezier(.4, 0, .2, 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