@volley/recognition-client-sdk 0.1.782 → 0.1.799

package/dist/recognition-client.d.ts

@@ -89,6 +89,29 @@ export declare class RealTimeTwoWayWebSocketRecognitionClient extends WebSocketA
      */
     private connectWithRetry;
     sendAudio(audioData: ArrayBuffer | ArrayBufferView | Blob): void;
+    /**
+     * Send PCM16 mono audio captured at any sample rate. The SDK downsamples
+     * to the session's target rate (currently 16 kHz per server validator)
+     * before sending.
+     *
+     * Use this when your capture pipeline produces audio at the system's
+     * native rate — `AudioContext` defaults to 44.1 kHz or 48 kHz on most
+     * desktop/mobile hardware — and you don't want to bring your own
+     * resampler. If your audio is already at the target rate, prefer
+     * `sendAudio()` to skip the resample step.
+     *
+     * Algorithm: box-filter averaging (see audio-resampler.ts). Cheap, no
+     * dependencies, has a built-in low-pass effect so aliasing stays out of
+     * the speech band. Suitable for ASR; not a substitute for a high-quality
+     * resampler if you're doing music or full-fidelity processing.
+     *
+     * Audio must be signed 16-bit little-endian PCM, mono. Stereo must be
+     * mixed to mono by the caller.
+     *
+     * @param audioData - PCM16 mono audio at `sourceSampleRate`.
+     * @param sourceSampleRate - Source sample rate in Hz (e.g. 44100, 48000).
+     */
+    sendAudioWithSampleRate(audioData: ArrayBuffer | ArrayBufferView | Blob, sourceSampleRate: number): void;
     private sendAudioInternal;
     /**
      * Only active ehwne client is in READY state. otherwise it will return immediately.

package/dist/recognition-client.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"recognition-client.d.ts","sourceRoot":"","sources":["../src/recognition-client.ts"],"names":[],"mappings":"AAAA;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;GAiCG;AAGH,OAAO,EAAE,oBAAoB,EAAE,MAAM,kBAAkB,CAAC;AACxD,OAAO,EAML,KAAK,qBAAqB,EAS1B,KAAK,aAAa,EAGnB,MAAM,qBAAqB,CAAC;AAE7B,OAAO,EAAE,WAAW,EAAE,MAAM,+BAA+B,CAAC;AAC5D,OAAO,KAAK,EACV,kBAAkB,EAClB,uBAAuB,EACvB,8CAA8C,EAE/C,MAAM,+BAA+B,CAAC;AAWvC;;;;GAIG;AACH,wBAAgB,qBAAqB,CAAC,IAAI,EAAE,MAAM,GAAG,OAAO,CAE3D;AAgCD;;GAEG;AACH,MAAM,MAAM,mBAAmB,GAAG,qBAAqB,CAAC;AAGxD,YAAY,EAAE,8CAA8C,EAAE,MAAM,+BAA+B,CAAC;AAkCpG;;;;;GAKG;AACH,qBAAa,wCACX,SAAQ,oBAAoB,CAAC,MAAM,EAAE,GAAG,EAAE,GAAG,CAC7C,YAAW,kBAAkB;IAE7B,OAAO,CAAC,MAAM,CAAC,QAAQ,CAAC,gBAAgB,CAAK;IAC7C,OAAO,CAAC,MAAM,CAAC,QAAQ,CAAC,uBAAuB,CAAoB;IAEnE,OAAO,CAAC,MAAM,CAAiB;IAC/B,OAAO,CAAC,WAAW,CAAkB;IACrC,OAAO,CAAC,YAAY,CAAyC;IAC7D,OAAO,CAAC,iBAAiB,CAAK;IAC9B,OAAO,CAAC,cAAc,CAAiB;IACvC,OAAO,CAAC,KAAK,CAAoC;IACjD,OAAO,CAAC,iBAAiB,CAA4B;IAGrD,OAAO,CAAC,iBAAiB,CAAS;IAGlC,OAAO,CAAC,cAAc,CAAK;IAC3B,OAAO,CAAC,eAAe,CAAK;IAC5B,OAAO,CAAC,qBAAqB,CAAO;IACpC,OAAO,CAAC,iBAAiB,CAAK;gBAElB,MAAM,EAAE,8CAA8C;IAkGlE;;;;;;OAMG;IACH,OAAO,CAAC,GAAG;IAWX;;;OAGG;IACH,OAAO,CAAC,OAAO;IAqBA,OAAO,IAAI,OAAO,CAAC,IAAI,CAAC;IA6BvC;;;OAGG;YACW,gBAAgB;IAkIrB,SAAS,CAAC,SAAS,EAAE,WAAW,GAAG,eAAe,GAAG,IAAI,GAAG,IAAI;IAiBzE;;;;;;;;;;;;;;;;;;;;;OAqBG;IACH,uBAAuB,CACrB,SAAS,EAAE,WAAW,GAAG,eAAe,GAAG,IAAI,EAC/C,gBAAgB,EAAE,MAAM,GACvB,IAAI;IAsBP,OAAO,CAAC,iBAAiB;IAsCzB;;;OAGG;IAEG,aAAa,IAAI,OAAO,CAAC,IAAI,CAAC;IAoCpC,cAAc,IAAI,IAAI;IAwBtB,mBAAmB,IAAI,MAAM;IAI7B,MAAM,IAAI,MAAM;IAIhB,QAAQ,IAAI,WAAW;IAIvB,WAAW,IAAI,OAAO;IAItB,YAAY,IAAI,OAAO;IAIvB,UAAU,IAAI,OAAO;IAIrB,uBAAuB,IAAI,OAAO;IAIlC,mBAAmB,IAAI,OAAO;IAI9B,aAAa,IAAI,OAAO;IAIxB,eAAe,CAAC,OAAO,EAAE,aAAa,GAAG,IAAI;IAmB7C,QAAQ,IAAI,uBAAuB;IAgBnC,SAAS,CAAC,WAAW,IAAI,IAAI;IAwF7B,SAAS,CAAC,cAAc,CAAC,IAAI,EAAE,MAAM,EAAE,MAAM,EAAE,MAAM,GAAG,IAAI;IA8C5D;;OAEG;IACH,OAAO,CAAC,uBAAuB;IAwB/B,SAAS,CAAC,OAAO,CAAC,KAAK,EAAE,KAAK,GAAG,IAAI;cAYlB,SAAS,CAAC,GAAG,EAAE;QAAE,CAAC,EAAE,MAAM,CAAC;QAAC,IAAI,EAAE,MAAM,CAAC;QAAC,IAAI,EAAE,GAAG,CAAA;KAAE,GAAG,IAAI;IAQ/E;;;OAGG;IACH,OAAO,CAAC,oBAAoB;IAiC5B;;;OAGG;IACH,OAAO,CAAC,YAAY;IAwBpB;;;;;;;;;OASG;IACH,eAAe,CAAC,SAAS,EAAE,WAAW,GAAG,eAAe,GAAG,IAAI,GAAG,IAAI;IAiBtE;;;OAGG;IACH,OAAO,CAAC,uBAAuB;IAiC/B;;;;OAIG;IACH,OAAO,CAAC,kBAAkB;CA2B3B"}

package/dist/recognition-client.types.d.ts

@@ -184,6 +184,23 @@ export interface IRecognitionClient {
      * @param audioData - PCM audio data as ArrayBuffer, typed array view, or Blob
      */
     sendAudio(audioData: ArrayBuffer | ArrayBufferView | Blob): void;
+    /**
+     * Send PCM16 mono audio captured at `sourceSampleRate`; the SDK
+     * downsamples to the session's target rate (currently 16 kHz, set by the
+     * server validator) before transmitting.
+     *
+     * Use this when your capture pipeline produces audio at the system's
+     * native rate (browser `AudioContext` is typically 44.1 kHz or 48 kHz).
+     * If your audio is already at the target rate, prefer `sendAudio()` to
+     * skip the resample step.
+     *
+     * Audio must be signed 16-bit little-endian PCM, mono. Stereo must be
+     * mixed to mono by the caller.
+     *
+     * @param audioData - PCM16 mono audio at `sourceSampleRate`.
+     * @param sourceSampleRate - Source sample rate in Hz (e.g. 44100, 48000).
+     */
+    sendAudioWithSampleRate(audioData: ArrayBuffer | ArrayBufferView | Blob, sourceSampleRate: number): void;
     /**
      * Stop recording and wait for final transcript
      * The server will close the connection after sending the final transcript.

package/dist/recognition-client.types.d.ts.map

@@ -1 +1 @@
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package/dist/simplified-vgf-recognition-client.d.ts

@@ -9,7 +9,7 @@
  */
 import { RecognitionState } from './vgf-recognition-state.js';
 import { IRecognitionClientConfig, ClientState } from './recognition-client.types.js';
-import { type GameContextV1 } from '@recog/shared-types';
+import type { GameContextV1 } from '@recog/shared-types';
 /**
  * Configuration for SimplifiedVGFRecognitionClient
  */
@@ -42,6 +42,14 @@ export interface ISimplifiedVGFRecognitionClient {
      * @param audioData - PCM audio data as ArrayBuffer, typed array, or Blob
      */
     sendAudio(audioData: ArrayBuffer | ArrayBufferView | Blob): void;
+    /**
+     * Send PCM16 mono audio captured at `sourceSampleRate`; the SDK
+     * downsamples to the session's target rate before transmitting. Use
+     * when capture is at the system's native rate (browser AudioContext is
+     * typically 44.1 kHz or 48 kHz). Audio must be signed 16-bit
+     * little-endian PCM, mono.
+     */
+    sendAudioWithSampleRate(audioData: ArrayBuffer | ArrayBufferView | Blob, sourceSampleRate: number): void;
     /**
      * Stop recording and wait for final transcription
      * @returns Promise that resolves when transcription is complete
@@ -132,6 +140,13 @@ export declare class SimplifiedVGFRecognitionClient implements ISimplifiedVGFRec
     constructor(config: SimplifiedVGFClientConfig);
     connect(): Promise<void>;
     sendAudio(audioData: ArrayBuffer | ArrayBufferView | Blob): void;
+    sendAudioWithSampleRate(audioData: ArrayBuffer | ArrayBufferView | Blob, sourceSampleRate: number): void;
+    /**
+     * Set VGF recording status to RECORDING on the first audio chunk.
+     * Idempotent — subsequent calls are no-ops until disconnect/stop resets
+     * `isRecordingAudio`.
+     */
+    private markRecordingStarted;
     stopRecording(): Promise<void>;
     stopAbnormally(): void;
     getAudioUtteranceId(): string;

package/dist/simplified-vgf-recognition-client.d.ts.map

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package/dist/utils/audio-resampler.d.ts

@@ -0,0 +1,32 @@
+/**
+ * Downsample PCM16 mono audio to a target sample rate.
+ *
+ * Used by `sendAudioWithSampleRate()` so integrators whose capture pipeline
+ * produces audio at the system's native rate (AudioContext defaults to
+ * 44.1 kHz or 48 kHz on most desktop/mobile hardware) can hand raw bytes
+ * to the SDK without having to bring in their own resampler. The
+ * recognition-service `SampleRateValidator` accepts only 16 kHz, so the SDK
+ * resamples on the client side before sending.
+ *
+ * Algorithm: box-filter averaging. For each output sample we average the
+ * source samples that fall into its time window. This is the cheapest
+ * correct approach for speech ASR — it has a built-in low-pass effect that
+ * suppresses aliasing far better than naive decimation or linear
+ * interpolation, while staying O(n) with no FFT and no dependencies.
+ * For integer ratios (e.g. 48000 → 16000, ratio = 3) it degenerates to a
+ * plain 3-sample average; for fractional ratios (e.g. 44100 → 16000) the
+ * window count varies by ±1 across output samples.
+ *
+ * Assumes the input is signed 16-bit little-endian PCM (the SDK's
+ * documented `AudioEncoding.LINEAR16` input format). Mono only. Stereo
+ * audio must be mixed to mono by the caller.
+ *
+ * @param input - Source PCM16 audio (ArrayBuffer or any ArrayBufferView).
+ * @param srcRate - Source sample rate in Hz (e.g. 44100, 48000).
+ * @param targetRate - Target sample rate in Hz. Must be ≤ srcRate.
+ * @returns A new ArrayBuffer of PCM16 samples at `targetRate`.
+ * @throws Error if `targetRate > srcRate` (upsampling is not supported —
+ *   capture at ≥ targetRate instead).
+ */
+export declare function downsamplePcm16(input: ArrayBuffer | ArrayBufferView, srcRate: number, targetRate: number): ArrayBuffer;
+//# sourceMappingURL=audio-resampler.d.ts.map

package/dist/utils/audio-resampler.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"audio-resampler.d.ts","sourceRoot":"","sources":["../../src/utils/audio-resampler.ts"],"names":[],"mappings":"AAAA;;;;;;;;;;;;;;;;;;;;;;;;;;;;;GA6BG;AACH,wBAAgB,eAAe,CAC7B,KAAK,EAAE,WAAW,GAAG,eAAe,EACpC,OAAO,EAAE,MAAM,EACf,UAAU,EAAE,MAAM,GACjB,WAAW,CA4Cb"}

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@volley/recognition-client-sdk",
-  "version": "0.1.782",
+  "version": "0.1.799",
   "description": "Recognition Service TypeScript/Node.js Client SDK",
   "type": "module",
   "main": "dist/index.js",
@@ -55,9 +55,9 @@
     "ts-jest": "29.2.5",
     "typescript": "5.1.6",
     "@recog/shared-config": "1.0.0",
-    "@recog/shared-types": "1.0.0",
     "@recog/shared-utils": "1.0.0",
-    "@recog/websocket": "1.0.0"
+    "@recog/websocket": "1.0.0",
+    "@recog/shared-types": "1.0.0"
   },
   "keywords": [
     "recognition",

package/src/index.spec.ts

@@ -6,6 +6,8 @@ describe('SDK top-level exports', () => {
     expect(RecognitionProvider.SELF_SERVE_VLLM).toBe('self-serve-vllm');
     expect(CartesiaModel.INK_WHISPER).toBe('ink-whisper');
     expect(CartesiaModel.INK_WHISPER_20250604).toBe('ink-whisper-2025-06-04');
+    expect(SelfServeVllmModel.QWEN3_ASR_0_6B).toBe('qwen3-asr-0.6b');
+    expect(SelfServeVllmModel.QWEN3_ASR_0_6B_WOF_LETTER).toBe('qwen3-asr-0.6b-wof-letter');
     expect(SelfServeVllmModel.QWEN3_ASR_1_7B).toBe('qwen3-asr-1.7b');
   });
 });

package/src/recognition-client.ts

@@ -65,6 +65,7 @@ import type {
 import { buildWebSocketUrl } from './utils/url-builder.js';
 import { AudioRingBuffer } from './utils/audio-ring-buffer.js';
 import { MessageHandler } from './utils/message-handler.js';
+import { downsamplePcm16 } from './utils/audio-resampler.js';
 import { ConnectionError } from './errors.js';
 
 // ============================================================================
@@ -215,11 +216,24 @@ export class RealTimeTwoWayWebSocketRecognitionClient
     const maxAttempts = Math.max(1, Math.min(5, retryConfig.maxAttempts ?? 4)); // Default: 4 attempts (3 retries), clamp 1-5
     const delayMs = retryConfig.delayMs ?? 200; // Fast retry for short audio sessions
 
+    // Normalize encoding to AudioEncoding enum at construction so the binary
+    // frame header (uint32) and ASRRequest serialization both get a number.
+    // String inputs ('linear16', 'LINEAR16') are accepted with a warn-level
+    // log; invalid strings throw via AudioEncoding.coerce().
+    const normalizedASRConfig = config.asrRequestConfig
+      ? {
+          ...config.asrRequestConfig,
+          encoding: AudioEncoding.coerce(config.asrRequestConfig.encoding, (warning) =>
+            config.logger?.('warn', warning)
+          ),
+        }
+      : undefined;
+
     // Process config with defaults
     this.config = {
       url,
       audioUtteranceId,
-      ...(config.asrRequestConfig && { asrRequestConfig: config.asrRequestConfig }),
+      ...(normalizedASRConfig && { asrRequestConfig: normalizedASRConfig }),
       ...(config.gameContext && { gameContext: config.gameContext }),
       ...(config.callbackUrls && { callbackUrls: config.callbackUrls }),
       onTranscript: config.onTranscript || (() => {}),
@@ -488,6 +502,53 @@ export class RealTimeTwoWayWebSocketRecognitionClient
     this.sendAudioInternal(audioData);
   }
 
+  /**
+   * Send PCM16 mono audio captured at any sample rate. The SDK downsamples
+   * to the session's target rate (currently 16 kHz per server validator)
+   * before sending.
+   *
+   * Use this when your capture pipeline produces audio at the system's
+   * native rate — `AudioContext` defaults to 44.1 kHz or 48 kHz on most
+   * desktop/mobile hardware — and you don't want to bring your own
+   * resampler. If your audio is already at the target rate, prefer
+   * `sendAudio()` to skip the resample step.
+   *
+   * Algorithm: box-filter averaging (see audio-resampler.ts). Cheap, no
+   * dependencies, has a built-in low-pass effect so aliasing stays out of
+   * the speech band. Suitable for ASR; not a substitute for a high-quality
+   * resampler if you're doing music or full-fidelity processing.
+   *
+   * Audio must be signed 16-bit little-endian PCM, mono. Stereo must be
+   * mixed to mono by the caller.
+   *
+   * @param audioData - PCM16 mono audio at `sourceSampleRate`.
+   * @param sourceSampleRate - Source sample rate in Hz (e.g. 44100, 48000).
+   */
+  sendAudioWithSampleRate(
+    audioData: ArrayBuffer | ArrayBufferView | Blob,
+    sourceSampleRate: number
+  ): void {
+    const targetRate =
+      typeof this.config.asrRequestConfig?.sampleRate === 'number'
+        ? this.config.asrRequestConfig.sampleRate
+        : SampleRate.RATE_16000;
+
+    if (audioData instanceof Blob) {
+      blobToArrayBuffer(audioData)
+        .then((arrayBuffer) => {
+          this.sendAudioInternal(
+            downsamplePcm16(arrayBuffer, sourceSampleRate, targetRate)
+          );
+        })
+        .catch((error) => {
+          this.log('warn', 'Failed to convert Blob to ArrayBuffer', error);
+        });
+      return;
+    }
+
+    this.sendAudioInternal(downsamplePcm16(audioData, sourceSampleRate, targetRate));
+  }
+
   private sendAudioInternal(audioData: ArrayBuffer | ArrayBufferView): void {
     const bytes = ArrayBuffer.isView(audioData) ? audioData.byteLength : audioData.byteLength;
     if (bytes === 0) return;
@@ -692,10 +753,7 @@ export class RealTimeTwoWayWebSocketRecognitionClient
           typeof this.config.asrRequestConfig.sampleRate === 'number'
             ? this.config.asrRequestConfig.sampleRate
             : SampleRate.RATE_16000,
-        encoding:
-          typeof this.config.asrRequestConfig.encoding === 'number'
-            ? this.config.asrRequestConfig.encoding
-            : AudioEncoding.LINEAR16,
+        encoding: this.config.asrRequestConfig.encoding as AudioEncoding,
         interimResults: this.config.asrRequestConfig.interimResults ?? false,
         // Auto-enable useContext if gameContext is provided, or use explicit value if set
         useContext: this.config.asrRequestConfig.useContext ?? !!this.config.gameContext,
@@ -892,7 +950,7 @@ export class RealTimeTwoWayWebSocketRecognitionClient
       ? audioData.byteLength
       : audioData.byteLength;
 
-    const encodingId = (this.config.asrRequestConfig?.encoding ||
+    const encodingId = (this.config.asrRequestConfig?.encoding ??
       AudioEncoding.LINEAR16) as AudioEncoding;
 
     const sampleRate =
@@ -987,7 +1045,7 @@ export class RealTimeTwoWayWebSocketRecognitionClient
 
     if (byteLength === 0) return;
 
-    const baseEncodingId = (this.config.asrRequestConfig?.encoding ||
+    const baseEncodingId = (this.config.asrRequestConfig?.encoding ??
       AudioEncoding.LINEAR16) as AudioEncoding;
 
     // Add offset to mark as prefix audio

package/src/recognition-client.types.ts

@@ -239,6 +239,27 @@ export interface IRecognitionClient {
    */
   sendAudio(audioData: ArrayBuffer | ArrayBufferView | Blob): void;
 
+  /**
+   * Send PCM16 mono audio captured at `sourceSampleRate`; the SDK
+   * downsamples to the session's target rate (currently 16 kHz, set by the
+   * server validator) before transmitting.
+   *
+   * Use this when your capture pipeline produces audio at the system's
+   * native rate (browser `AudioContext` is typically 44.1 kHz or 48 kHz).
+   * If your audio is already at the target rate, prefer `sendAudio()` to
+   * skip the resample step.
+   *
+   * Audio must be signed 16-bit little-endian PCM, mono. Stereo must be
+   * mixed to mono by the caller.
+   *
+   * @param audioData - PCM16 mono audio at `sourceSampleRate`.
+   * @param sourceSampleRate - Source sample rate in Hz (e.g. 44100, 48000).
+   */
+  sendAudioWithSampleRate(
+    audioData: ArrayBuffer | ArrayBufferView | Blob,
+    sourceSampleRate: number
+  ): void;
+
   /**
    * Stop recording and wait for final transcript
    * The server will close the connection after sending the final transcript.

package/src/simplified-vgf-recognition-client.ts

@@ -27,7 +27,7 @@ import {
     updateStateOnStop,
     resetRecognitionVGFState
 } from './vgf-recognition-mapper.js';
-import { RecognitionContextTypeV1, type GameContextV1 } from '@recog/shared-types';
+import type { GameContextV1 } from '@recog/shared-types';
 
 /**
  * Configuration for SimplifiedVGFRecognitionClient
@@ -66,6 +66,18 @@ export interface ISimplifiedVGFRecognitionClient {
      */
     sendAudio(audioData: ArrayBuffer | ArrayBufferView | Blob): void;
 
+    /**
+     * Send PCM16 mono audio captured at `sourceSampleRate`; the SDK
+     * downsamples to the session's target rate before transmitting. Use
+     * when capture is at the system's native rate (browser AudioContext is
+     * typically 44.1 kHz or 48 kHz). Audio must be signed 16-bit
+     * little-endian PCM, mono.
+     */
+    sendAudioWithSampleRate(
+        audioData: ArrayBuffer | ArrayBufferView | Blob,
+        sourceSampleRate: number
+    ): void;
+
     /**
      * Stop recording and wait for final transcription
      * @returns Promise that resolves when transcription is complete
@@ -254,7 +266,7 @@ export class SimplifiedVGFRecognitionClient implements ISimplifiedVGFRecognition
             ...clientConfig,
 
             // These callbacks ONLY update the VGF state sink
-            onTranscript: (result) => {
+            onTranscript: (result): void => {
                 // Skip update if UUID doesn't match (stale callback from previous session)
                 if (result.audioUtteranceId && result.audioUtteranceId !== this.expectedUuid) {
                     if (this.logger) {
@@ -275,7 +287,7 @@ export class SimplifiedVGFRecognitionClient implements ISimplifiedVGFRecognition
                 }
             },
 
-            onMetadata: (metadata) => {
+            onMetadata: (metadata): void => {
                 // Skip update if UUID doesn't match (stale callback from previous session)
                 if (metadata.audioUtteranceId && metadata.audioUtteranceId !== this.expectedUuid) {
                     if (this.logger) {
@@ -291,14 +303,14 @@ export class SimplifiedVGFRecognitionClient implements ISimplifiedVGFRecognition
                 }
             },
 
-            onFunctionCall: (result) => {
+            onFunctionCall: (result): void => {
                 // Pass through function call - no VGF state changes needed for P2 feature
                 if (clientConfig.onFunctionCall) {
                     clientConfig.onFunctionCall(result);
                 }
             },
 
-            onError: (error) => {
+            onError: (error): void => {
                 // Skip update if UUID doesn't match (stale callback from previous session)
                 if (error.audioUtteranceId && error.audioUtteranceId !== this.expectedUuid) {
                     if (this.logger) {
@@ -318,14 +330,14 @@ export class SimplifiedVGFRecognitionClient implements ISimplifiedVGFRecognition
                 }
             },
 
-            onConnected: () => {
+            onConnected: (): void => {
                 // Don't update READY here - client can accept audio before connection
                 if (clientConfig.onConnected) {
                     clientConfig.onConnected();
                 }
             },
 
-            onDisconnected: (code, reason) => {
+            onDisconnected: (code, reason): void => {
                 this.isRecordingAudio = false; // Reset on disconnect
                 if (clientConfig.onDisconnected) {
                     clientConfig.onDisconnected(code, reason);
@@ -343,19 +355,34 @@ export class SimplifiedVGFRecognitionClient implements ISimplifiedVGFRecognition
     }
 
     sendAudio(audioData: ArrayBuffer | ArrayBufferView | Blob): void {
-        // Track recording for state updates
-        if (!this.isRecordingAudio) {
-            this.isRecordingAudio = true;
-            this.state = {
-                ...this.state,
-                startRecordingStatus: 'RECORDING',
-                startRecordingTimestamp: new Date().toISOString()
-            };
-            this.notifyStateChange();
-        }
+        this.markRecordingStarted();
         this.client.sendAudio(audioData);
     }
 
+    sendAudioWithSampleRate(
+        audioData: ArrayBuffer | ArrayBufferView | Blob,
+        sourceSampleRate: number
+    ): void {
+        this.markRecordingStarted();
+        this.client.sendAudioWithSampleRate(audioData, sourceSampleRate);
+    }
+
+    /**
+     * Set VGF recording status to RECORDING on the first audio chunk.
+     * Idempotent — subsequent calls are no-ops until disconnect/stop resets
+     * `isRecordingAudio`.
+     */
+    private markRecordingStarted(): void {
+        if (this.isRecordingAudio) return;
+        this.isRecordingAudio = true;
+        this.state = {
+            ...this.state,
+            startRecordingStatus: 'RECORDING',
+            startRecordingTimestamp: new Date().toISOString()
+        };
+        this.notifyStateChange();
+    }
+
     async stopRecording(): Promise<void> {
         this.isRecordingAudio = false;
         this.state = updateStateOnStop(this.state);

package/src/utils/audio-resampler.spec.ts

@@ -0,0 +1,69 @@
+import { downsamplePcm16 } from './audio-resampler.js';
+
+// Helper: build an Int16Array from numbers and return its backing buffer.
+const buf = (samples: number[]): ArrayBuffer => new Int16Array(samples).slice().buffer;
+const samples = (b: ArrayBuffer): number[] => Array.from(new Int16Array(b));
+
+describe('downsamplePcm16', () => {
+  it('returns a defensive copy when srcRate === targetRate', () => {
+    const input = new Int16Array([100, 200, 300, 400]);
+    const out = downsamplePcm16(input, 16000, 16000);
+    expect(samples(out)).toEqual([100, 200, 300, 400]);
+    // Mutating the input must not affect the returned buffer.
+    input[0] = 9999;
+    expect(samples(out)[0]).toBe(100);
+  });
+
+  it('integer ratio 48000 → 16000 averages every 3 source samples', () => {
+    // 9 source samples → 3 output samples (ratio = 3)
+    const input = buf([0, 6, 12, 30, 60, 90, -3, -6, -9]);
+    const out = downsamplePcm16(input, 48000, 16000);
+    expect(samples(out)).toEqual([
+      Math.round((0 + 6 + 12) / 3), // 6
+      Math.round((30 + 60 + 90) / 3), // 60
+      Math.round((-3 + -6 + -9) / 3), // -6
+    ]);
+  });
+
+  it('fractional ratio 44100 → 16000 produces ~44100/16000 output samples', () => {
+    // 441 source samples at 44.1kHz ≈ 10ms; expect ~160 output samples at 16kHz.
+    const input = new Int16Array(441).fill(1000);
+    const out = downsamplePcm16(input, 44100, 16000);
+    const outArr = new Int16Array(out);
+    expect(outArr.length).toBe(Math.floor(441 / (44100 / 16000))); // = 160
+    // Constant input should produce constant output (within rounding).
+    for (let i = 0; i < outArr.length; i++) {
+      expect(outArr[i]).toBe(1000);
+    }
+  });
+
+  it('accepts ArrayBuffer input as well as typed-array view', () => {
+    const view = new Int16Array([10, 20, 30, 40, 50, 60]);
+    const fromView = samples(downsamplePcm16(view, 48000, 16000));
+    const fromBuf = samples(downsamplePcm16(view.buffer, 48000, 16000));
+    expect(fromBuf).toEqual(fromView);
+  });
+
+  it('handles a typed-array view that shares a larger backing buffer', () => {
+    // Simulate a slice from a bigger capture buffer — only the view's bytes
+    // should be considered, not the rest of the underlying ArrayBuffer.
+    const big = new Int16Array([99, 99, 0, 6, 12, 30, 60, 90, 99, 99]);
+    const slice = new Int16Array(big.buffer, 2 * Int16Array.BYTES_PER_ELEMENT, 6);
+    const out = downsamplePcm16(slice, 48000, 16000);
+    expect(samples(out)).toEqual([
+      Math.round((0 + 6 + 12) / 3),
+      Math.round((30 + 60 + 90) / 3),
+    ]);
+  });
+
+  it('returns an empty buffer for empty input', () => {
+    const out = downsamplePcm16(new Int16Array(0), 48000, 16000);
+    expect(new Int16Array(out).length).toBe(0);
+  });
+
+  it('throws when asked to upsample', () => {
+    expect(() => downsamplePcm16(new Int16Array([1, 2]), 8000, 16000)).toThrow(
+      /cannot upsample/i
+    );
+  });
+});

package/src/utils/audio-resampler.ts

@@ -0,0 +1,79 @@
+/**
+ * Downsample PCM16 mono audio to a target sample rate.
+ *
+ * Used by `sendAudioWithSampleRate()` so integrators whose capture pipeline
+ * produces audio at the system's native rate (AudioContext defaults to
+ * 44.1 kHz or 48 kHz on most desktop/mobile hardware) can hand raw bytes
+ * to the SDK without having to bring in their own resampler. The
+ * recognition-service `SampleRateValidator` accepts only 16 kHz, so the SDK
+ * resamples on the client side before sending.
+ *
+ * Algorithm: box-filter averaging. For each output sample we average the
+ * source samples that fall into its time window. This is the cheapest
+ * correct approach for speech ASR — it has a built-in low-pass effect that
+ * suppresses aliasing far better than naive decimation or linear
+ * interpolation, while staying O(n) with no FFT and no dependencies.
+ * For integer ratios (e.g. 48000 → 16000, ratio = 3) it degenerates to a
+ * plain 3-sample average; for fractional ratios (e.g. 44100 → 16000) the
+ * window count varies by ±1 across output samples.
+ *
+ * Assumes the input is signed 16-bit little-endian PCM (the SDK's
+ * documented `AudioEncoding.LINEAR16` input format). Mono only. Stereo
+ * audio must be mixed to mono by the caller.
+ *
+ * @param input - Source PCM16 audio (ArrayBuffer or any ArrayBufferView).
+ * @param srcRate - Source sample rate in Hz (e.g. 44100, 48000).
+ * @param targetRate - Target sample rate in Hz. Must be ≤ srcRate.
+ * @returns A new ArrayBuffer of PCM16 samples at `targetRate`.
+ * @throws Error if `targetRate > srcRate` (upsampling is not supported —
+ *   capture at ≥ targetRate instead).
+ */
+export function downsamplePcm16(
+  input: ArrayBuffer | ArrayBufferView,
+  srcRate: number,
+  targetRate: number
+): ArrayBuffer {
+  if (targetRate > srcRate) {
+    throw new Error(
+      `downsamplePcm16: cannot upsample from ${srcRate}Hz to ${targetRate}Hz; ` +
+        `capture audio at ≥ ${targetRate}Hz instead.`
+    );
+  }
+
+  // Wrap input as Int16Array regardless of how it was passed in. Slice the
+  // backing ArrayBuffer to the exact byte range so a typed-array view that
+  // shares a larger buffer doesn't pull in neighbouring bytes.
+  const buffer = ArrayBuffer.isView(input)
+    ? input.buffer.slice(input.byteOffset, input.byteOffset + input.byteLength)
+    : input;
+  const src = new Int16Array(buffer);
+
+  if (srcRate === targetRate || src.length === 0) {
+    // Return a defensive copy so the caller's typed-array view can't be
+    // mutated through the returned buffer.
+    return src.slice().buffer;
+  }
+
+  const ratio = srcRate / targetRate;
+  const dstLen = Math.floor(src.length / ratio);
+  const dst = new Int16Array(dstLen);
+
+  for (let i = 0; i < dstLen; i++) {
+    const startPos = i * ratio;
+    const endPos = (i + 1) * ratio;
+    const startIdx = Math.floor(startPos);
+    const endIdx = Math.min(Math.ceil(endPos), src.length);
+
+    let sum = 0;
+    let count = 0;
+    for (let j = startIdx; j < endIdx; j++) {
+      // Bounds are guaranteed by the surrounding floor/ceil/min math, but
+      // noUncheckedIndexedAccess still narrows to `number | undefined`.
+      sum += src[j] ?? 0;
+      count++;
+    }
+    dst[i] = count > 0 ? Math.round(sum / count) : 0;
+  }
+
+  return dst.buffer;
+}