@volley/recognition-client-sdk 0.1.767 → 0.1.799

package/dist/index.bundled.d.ts

@@ -24,6 +24,7 @@ declare enum RecognitionProvider {
     BEDROCK = "bedrock",
     INWORLD_STT = "inworld-stt",
     AWS_TRANSCRIBE = "aws-transcribe",
+    AMAZON_NOVA_SONIC = "amazon-nova-sonic",
     TEST_ASR_PROVIDER_QUOTA = "test-asr-provider-quota",
     TEST_ASR_STREAMING = "test-asr-streaming"
 }
@@ -96,6 +97,7 @@ declare enum ElevenLabsModel {
  * @see https://platform.openai.com/docs/models/gpt-4o-transcribe
  */
 declare enum OpenAIRealtimeModel {
+    GPT_REALTIME_WHISPER = "gpt-realtime-whisper",
     GPT_4O_TRANSCRIBE = "gpt-4o-transcribe",
     GPT_4O_MINI_TRANSCRIBE = "gpt-4o-mini-transcribe"
 }
@@ -146,135 +148,28 @@ declare enum InworldSttModel {
 declare enum AwsTranscribeModel {
     DEFAULT = "default"
 }
+/**
+ * Amazon Nova Sonic bidirectional streaming model (Bedrock).
+ * Speech-to-speech model; we consume the USER FINAL transcript and discard the assistant text/audio output.
+ * @see https://docs.aws.amazon.com/nova/latest/userguide/speech-bidirection.html
+ */
+declare enum AmazonNovaSonicModel {
+    AMAZON_NOVA_SONIC_V1 = "amazon.nova-sonic-v1:0",
+    AMAZON_NOVA_2_SONIC = "amazon.nova-2-sonic-v1:0"
+}
 /**
  * Self-serve vLLM batch transcription models
- * Backed by recognition-inference / RunPod `/transcribe`
+ * Backed by recognition-inference / RunPod `/ws/transcribe`
  */
 declare enum SelfServeVllmModel {
+    QWEN3_ASR_0_6B = "qwen3-asr-0.6b",
+    QWEN3_ASR_0_6B_WOF_LETTER = "qwen3-asr-0.6b-wof-letter",
     QWEN3_ASR_1_7B = "qwen3-asr-1.7b"
 }
 /**
  * Type alias for any model from any provider
  */
-type RecognitionModel = DeepgramModel | GoogleModel | FireworksModel | GladiaModel | ElevenLabsModel | OpenAIRealtimeModel | MistralVoxtralModel | CartesiaModel | DashScopeModel | InworldSttModel | SelfServeVllmModel | BedrockModel | AwsTranscribeModel | string;
-
-/**
- * Audio encoding types
- */
-declare enum AudioEncoding {
-    ENCODING_UNSPECIFIED = 0,
-    LINEAR16 = 1,
-    OGG_OPUS = 2,
-    FLAC = 3,
-    MULAW = 4,
-    ALAW = 5
-}
-declare namespace AudioEncoding {
-    /**
-     * Convert numeric ID to AudioEncoding enum
-     * @param id - Numeric encoding identifier (0-5)
-     * @returns AudioEncoding enum value or undefined if invalid
-     */
-    function fromId(id: number): AudioEncoding | undefined;
-    /**
-     * Convert string name to AudioEncoding enum
-     * @param nameStr - String name like "linear16", "LINEAR16", "ogg_opus", "OGG_OPUS", etc. (case insensitive)
-     * @returns AudioEncoding enum value or undefined if invalid
-     */
-    function fromName(nameStr: string): AudioEncoding | undefined;
-    /**
-     * Convert AudioEncoding enum to numeric ID
-     * @param encoding - AudioEncoding enum value
-     * @returns Numeric ID (0-5)
-     */
-    function toId(encoding: AudioEncoding): number;
-    /**
-     * Convert AudioEncoding enum to string name
-     * @param encoding - AudioEncoding enum value
-     * @returns String name like "LINEAR16", "MULAW", etc.
-     */
-    function toName(encoding: AudioEncoding): string;
-    /**
-     * Check if a numeric ID is a valid encoding
-     * @param id - Numeric identifier to validate
-     * @returns true if valid encoding ID
-     */
-    function isIdValid(id: number): boolean;
-    /**
-     * Check if a string name is a valid encoding
-     * @param nameStr - String name to validate
-     * @returns true if valid encoding name
-     */
-    function isNameValid(nameStr: string): boolean;
-}
-/**
- * Common sample rates (in Hz)
- */
-declare enum SampleRate {
-    RATE_8000 = 8000,
-    RATE_16000 = 16000,
-    RATE_22050 = 22050,
-    RATE_24000 = 24000,
-    RATE_32000 = 32000,
-    RATE_44100 = 44100,
-    RATE_48000 = 48000
-}
-declare namespace SampleRate {
-    /**
-     * Convert Hz value to SampleRate enum
-     * @param hz - Sample rate in Hz (8000, 16000, etc.)
-     * @returns SampleRate enum value or undefined if invalid
-     */
-    function fromHz(hz: number): SampleRate | undefined;
-    /**
-     * Convert string name to SampleRate enum
-     * @param nameStr - String name like "rate_8000", "RATE_16000", etc. (case insensitive)
-     * @returns SampleRate enum value or undefined if invalid
-     */
-    function fromName(nameStr: string): SampleRate | undefined;
-    /**
-     * Convert SampleRate enum to Hz value
-     * @param rate - SampleRate enum value
-     * @returns Hz value (8000, 16000, etc.)
-     */
-    function toHz(rate: SampleRate): number;
-    /**
-     * Convert SampleRate enum to string name
-     * @param rate - SampleRate enum value
-     * @returns String name like "RATE_8000", "RATE_16000", etc.
-     */
-    function toName(rate: SampleRate): string;
-    /**
-     * Check if a numeric Hz value is a valid sample rate
-     * @param hz - Hz value to validate
-     * @returns true if valid sample rate
-     */
-    function isHzValid(hz: number): boolean;
-    /**
-     * Check if a string name is a valid sample rate
-     * @param nameStr - String name to validate
-     * @returns true if valid sample rate name
-     */
-    function isNameValid(nameStr: string): boolean;
-}
-/**
- * Supported languages for recognition
- * Using BCP-47 language tags
- */
-declare enum Language {
-    ENGLISH_US = "en-US",
-    ENGLISH_GB = "en-GB",
-    SPANISH_ES = "es-ES",
-    SPANISH_MX = "es-MX",
-    FRENCH_FR = "fr-FR",
-    GERMAN_DE = "de-DE",
-    ITALIAN_IT = "it-IT",
-    PORTUGUESE_BR = "pt-BR",
-    JAPANESE_JP = "ja-JP",
-    KOREAN_KR = "ko-KR",
-    CHINESE_CN = "zh-CN",
-    CHINESE_TW = "zh-TW"
-}
+type RecognitionModel = DeepgramModel | GoogleModel | FireworksModel | GladiaModel | ElevenLabsModel | OpenAIRealtimeModel | MistralVoxtralModel | CartesiaModel | DashScopeModel | InworldSttModel | SelfServeVllmModel | BedrockModel | AwsTranscribeModel | AmazonNovaSonicModel | string;
 
 /**
  * Recognition Result Types V1
@@ -294,6 +189,16 @@ declare enum RecognitionResultTypeV1 {
     AUDIO_METRICS = "AudioMetrics",
     SESSION_CONFIGURED = "SessionConfigured"
 }
+/**
+ * Source of a phrase detection — what kind of provider feature produced
+ * the hit. Currently only Deepgram's `search` parameter is wired up, so
+ * this enum has one value. New entries (e.g. KEYWORDS, KEYTERMS,
+ * SPEECH_CONTEXTS) get added when other providers join.
+ */
+declare enum DetectionTypeV1 {
+    /** Deepgram phonetic phrase match via the `search=…` request parameter */
+    SEARCH = "search"
+}
 /**
  * Transcription result V1 - contains transcript message
  * In the long run game side should not need to know it. In the short run it is send back to client.
@@ -318,6 +223,25 @@ declare const TranscriptionResultSchemaV1: z.ZodObject<{
     receivedAtMs: z.ZodOptional<z.ZodNumber>;
     accumulatedAudioTimeMs: z.ZodOptional<z.ZodNumber>;
     rawAudioTimeMs: z.ZodOptional<z.ZodNumber>;
+    detections: z.ZodOptional<z.ZodArray<z.ZodObject<{
+        type: z.ZodNativeEnum<typeof DetectionTypeV1>;
+        query: z.ZodString;
+        score: z.ZodNumber;
+        startMs: z.ZodOptional<z.ZodNumber>;
+        endMs: z.ZodOptional<z.ZodNumber>;
+    }, "strip", z.ZodTypeAny, {
+        type: DetectionTypeV1;
+        query: string;
+        score: number;
+        startMs?: number | undefined;
+        endMs?: number | undefined;
+    }, {
+        type: DetectionTypeV1;
+        query: string;
+        score: number;
+        startMs?: number | undefined;
+        endMs?: number | undefined;
+    }>, "many">>;
 }, "strip", z.ZodTypeAny, {
     type: RecognitionResultTypeV1.TRANSCRIPTION;
     audioUtteranceId: string;
@@ -337,6 +261,13 @@ declare const TranscriptionResultSchemaV1: z.ZodObject<{
     receivedAtMs?: number | undefined;
     accumulatedAudioTimeMs?: number | undefined;
     rawAudioTimeMs?: number | undefined;
+    detections?: {
+        type: DetectionTypeV1;
+        query: string;
+        score: number;
+        startMs?: number | undefined;
+        endMs?: number | undefined;
+    }[] | undefined;
 }, {
     type: RecognitionResultTypeV1.TRANSCRIPTION;
     audioUtteranceId: string;
@@ -356,6 +287,13 @@ declare const TranscriptionResultSchemaV1: z.ZodObject<{
     receivedAtMs?: number | undefined;
     accumulatedAudioTimeMs?: number | undefined;
     rawAudioTimeMs?: number | undefined;
+    detections?: {
+        type: DetectionTypeV1;
+        query: string;
+        score: number;
+        startMs?: number | undefined;
+        endMs?: number | undefined;
+    }[] | undefined;
 }>;
 type TranscriptionResultV1 = z.infer<typeof TranscriptionResultSchemaV1>;
 /**
@@ -676,6 +614,138 @@ declare const AudioMetricsResultSchemaV1: z.ZodObject<{
 }>;
 type AudioMetricsResultV1 = z.infer<typeof AudioMetricsResultSchemaV1>;
 
+/**
+ * Audio encoding types
+ */
+declare enum AudioEncoding {
+    ENCODING_UNSPECIFIED = 0,
+    LINEAR16 = 1,
+    OGG_OPUS = 2,
+    FLAC = 3,
+    MULAW = 4,
+    ALAW = 5
+}
+declare namespace AudioEncoding {
+    /**
+     * Convert numeric ID to AudioEncoding enum
+     * @param id - Numeric encoding identifier (0-5)
+     * @returns AudioEncoding enum value or undefined if invalid
+     */
+    function fromId(id: number): AudioEncoding | undefined;
+    /**
+     * Convert string name to AudioEncoding enum
+     * @param nameStr - String name like "linear16", "LINEAR16", "ogg_opus", "OGG_OPUS", etc. (case insensitive)
+     * @returns AudioEncoding enum value or undefined if invalid
+     */
+    function fromName(nameStr: string): AudioEncoding | undefined;
+    /**
+     * Convert AudioEncoding enum to numeric ID
+     * @param encoding - AudioEncoding enum value
+     * @returns Numeric ID (0-5)
+     */
+    function toId(encoding: AudioEncoding): number;
+    /**
+     * Convert AudioEncoding enum to string name
+     * @param encoding - AudioEncoding enum value
+     * @returns String name like "LINEAR16", "MULAW", etc.
+     */
+    function toName(encoding: AudioEncoding): string;
+    /**
+     * Check if a numeric ID is a valid encoding
+     * @param id - Numeric identifier to validate
+     * @returns true if valid encoding ID
+     */
+    function isIdValid(id: number): boolean;
+    /**
+     * Check if a string name is a valid encoding
+     * @param nameStr - String name to validate
+     * @returns true if valid encoding name
+     */
+    function isNameValid(nameStr: string): boolean;
+    /**
+     * Coerce a possibly-stringly-typed encoding value into the AudioEncoding enum.
+     *
+     * - enum / number → returned as-is (already AudioEncoding-shaped)
+     * - string (case-insensitive, e.g. 'linear16', 'LINEAR16') → converted via {@link fromName}.
+     *   Invokes `onStringInput` with a warning message so callers can route it
+     *   to their preferred logger.
+     * - invalid string → throws (preferred over silent fallback so typos surface)
+     * - undefined → defaults to {@link AudioEncoding.LINEAR16}
+     *
+     * Always normalize at the SDK / server boundary so downstream code can rely
+     * on a numeric AudioEncoding (the wire-level binary frame header is uint32).
+     */
+    function coerce(value: AudioEncoding | string | number | undefined, onStringInput?: (warning: string) => void): AudioEncoding;
+}
+/**
+ * Common sample rates (in Hz)
+ */
+declare enum SampleRate {
+    RATE_8000 = 8000,
+    RATE_16000 = 16000,
+    RATE_22050 = 22050,
+    RATE_24000 = 24000,
+    RATE_32000 = 32000,
+    RATE_44100 = 44100,
+    RATE_48000 = 48000
+}
+declare namespace SampleRate {
+    /**
+     * Convert Hz value to SampleRate enum
+     * @param hz - Sample rate in Hz (8000, 16000, etc.)
+     * @returns SampleRate enum value or undefined if invalid
+     */
+    function fromHz(hz: number): SampleRate | undefined;
+    /**
+     * Convert string name to SampleRate enum
+     * @param nameStr - String name like "rate_8000", "RATE_16000", etc. (case insensitive)
+     * @returns SampleRate enum value or undefined if invalid
+     */
+    function fromName(nameStr: string): SampleRate | undefined;
+    /**
+     * Convert SampleRate enum to Hz value
+     * @param rate - SampleRate enum value
+     * @returns Hz value (8000, 16000, etc.)
+     */
+    function toHz(rate: SampleRate): number;
+    /**
+     * Convert SampleRate enum to string name
+     * @param rate - SampleRate enum value
+     * @returns String name like "RATE_8000", "RATE_16000", etc.
+     */
+    function toName(rate: SampleRate): string;
+    /**
+     * Check if a numeric Hz value is a valid sample rate
+     * @param hz - Hz value to validate
+     * @returns true if valid sample rate
+     */
+    function isHzValid(hz: number): boolean;
+    /**
+     * Check if a string name is a valid sample rate
+     * @param nameStr - String name to validate
+     * @returns true if valid sample rate name
+     */
+    function isNameValid(nameStr: string): boolean;
+}
+/**
+ * Supported languages for recognition
+ * Using BCP-47 language tags
+ */
+declare enum Language {
+    ENGLISH_US = "en-US",
+    ENGLISH_GB = "en-GB",
+    SPANISH_ES = "es-ES",
+    SPANISH_MX = "es-MX",
+    FRENCH_FR = "fr-FR",
+    GERMAN_DE = "de-DE",
+    ITALIAN_IT = "it-IT",
+    PORTUGUESE_BR = "pt-BR",
+    JAPANESE_JP = "ja-JP",
+    KOREAN_KR = "ko-KR",
+    CHINESE_CN = "zh-CN",
+    CHINESE_TW = "zh-TW"
+}
+
 /**
  * Error Exception Types
  *
@@ -1371,6 +1441,7 @@ declare const ASRRequestSchemaV1: z.ZodObject<{
     prefixId: z.ZodOptional<z.ZodString>;
     prefixTextToRemove: z.ZodOptional<z.ZodArray<z.ZodString, "many">>;
     audioMetricsIntervalMs: z.ZodOptional<z.ZodNumber>;
+    appendSearch: z.ZodOptional<z.ZodBoolean>;
     debugCommand: z.ZodOptional<z.ZodObject<{
         enableDebugLog: z.ZodDefault<z.ZodOptional<z.ZodBoolean>>;
         enableAudioStorage: z.ZodDefault<z.ZodOptional<z.ZodBoolean>>;
@@ -1413,6 +1484,7 @@ declare const ASRRequestSchemaV1: z.ZodObject<{
     prefixId?: string | undefined;
     prefixTextToRemove?: string[] | undefined;
     audioMetricsIntervalMs?: number | undefined;
+    appendSearch?: boolean | undefined;
     debugCommand?: {
         enableDebugLog: boolean;
         enableAudioStorage: boolean;
@@ -1445,6 +1517,7 @@ declare const ASRRequestSchemaV1: z.ZodObject<{
     prefixId?: string | undefined;
     prefixTextToRemove?: string[] | undefined;
     audioMetricsIntervalMs?: number | undefined;
+    appendSearch?: boolean | undefined;
     debugCommand?: {
         enableDebugLog?: boolean | undefined;
         enableAudioStorage?: boolean | undefined;
@@ -1579,10 +1652,10 @@ interface ASRRequestConfig {
      * doesn't respond with is_final=true after stopRecording().
      *
      * - aggressive: 100ms - fast response, may cut off slow providers
-     * - balanced: 500ms - current default, good for most cases
-     * - conservative: 1000ms - wait longer for complex utterances
+     * - balanced: 500ms - good for most cases
+     * - conservative: 1000ms - current default, wait longer for complex utterances
      *
-     * @default 'balanced'
+     * @default 'conservative'
      * @see FinalTranscriptStability enum for detailed descriptions
      */
     finalTranscriptStability?: FinalTranscriptStability | string;
@@ -1676,6 +1749,29 @@ interface ASRRequestConfig {
      * @example 500
      */
     audioMetricsIntervalMs?: number;
+    /**
+     * Opt-in: round-trip Deepgram `search` phrase hits into the transcript.
+     *
+     * When `true` AND the resolved provider/model is **deepgram nova-2** AND the
+     * GameContext `gamePhase` is `'Solve Puzzle'`, every Deepgram Results event
+     * with a `channel.search` hit at confidence ≥ 0.6 has the original query
+     * prepended to the transcript text delivered to the client. This restores
+     * parity with the legacy Roku→Deepgram WoF Puzzle-Solve path where the
+     * phrase round-trip lets downstream NLU match multi-word puzzle solutions
+     * even when nova-2's primary transcription drifts.
+     *
+     * Default: `false` (no prepend; transcript is whatever nova-2 produces).
+     *
+     * Scope guard rationale:
+     * - nova-2 only: nova-3 / flux do not need this (they handle phrase
+     *   spotting differently and the prepend would only add noise).
+     * - Solve-Puzzle scene only: other WoF scenes (Letter-Guess,
+     *   Bonus-Round, etc.) do NOT want the slotMap phrase prepended — only
+     *   Puzzle-Solve depends on the phrase round-trip.
+     *
+     * @default false
+     */
+    appendSearch?: boolean;
     /**
      * Optional fallback ASR configurations
      *
@@ -1736,7 +1832,10 @@ declare enum GeminiModel {
 }
 
 /**
- * OpenAI Model Types
+ * OpenAI Batch API Model Types (HTTP API)
+ * @see https://platform.openai.com/docs/guides/speech-to-text
+ *
+ * Note: For openai-realtime provider models, see OpenAIRealtimeModel in provider.types.ts
  */
 declare enum OpenAIModel {
     WHISPER_1 = "whisper-1"
@@ -2090,6 +2189,23 @@ 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.
@@ -2295,6 +2411,29 @@ declare class RealTimeTwoWayWebSocketRecognitionClient extends WebSocketAudioCli
      */
     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.
@@ -2718,6 +2857,14 @@ 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
@@ -2808,6 +2955,13 @@ declare class SimplifiedVGFRecognitionClient implements ISimplifiedVGFRecognitio
     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;
@@ -2934,5 +3088,5 @@ declare function getRecognitionConductorHttpBase(stage?: Stage | string | null |
 declare function getRecognitionConductorWsBase(stage?: Stage | string | null | undefined): string;
 declare function getRecognitionConductorHost(stage?: Stage | string | null | undefined): string;
 
-export { AudioEncoding, AwsTranscribeModel, BedrockModel, CartesiaModel, ClientControlActionV1, ClientState, ConfigBuilder, ConnectionError, ControlSignalTypeV1 as ControlSignal, ControlSignalTypeV1, DashScopeModel, DeepgramModel, ElevenLabsModel, ErrorTypeV1, FinalTranscriptStability, FireworksModel, GeminiModel, GladiaModel, GoogleModel, Language, MistralVoxtralModel, OpenAIModel, OpenAIRealtimeModel, RECOGNITION_CONDUCTOR_BASES, RECOGNITION_SERVICE_BASES, RealTimeTwoWayWebSocketRecognitionClient, RecognitionContextTypeV1, RecognitionError, RecognitionProvider, RecognitionResultTypeV1, RecognitionVGFStateSchema, RecordingStatus, STAGES, SampleRate, SelfServeVllmModel, SimplifiedVGFRecognitionClient, TimeoutError, TranscriptionStatus, ValidationError, createClient, createClientWithBuilder, createDefaultASRConfig, createInitialRecognitionState, createSimplifiedVGFClient, getRecognitionConductorBase, getRecognitionConductorHost, getRecognitionConductorHttpBase, getRecognitionConductorWsBase, getRecognitionServiceBase, getRecognitionServiceHost, getRecognitionServiceHttpBase, getRecognitionServiceWsBase, getUserFriendlyMessage, isExceptionImmediatelyAvailable, isNormalDisconnection, isValidRecordingStatusTransition, normalizeStage, resetRecognitionVGFState };
+export { AmazonNovaSonicModel, AudioEncoding, AwsTranscribeModel, BedrockModel, CartesiaModel, ClientControlActionV1, ClientState, ConfigBuilder, ConnectionError, ControlSignalTypeV1 as ControlSignal, ControlSignalTypeV1, DashScopeModel, DeepgramModel, ElevenLabsModel, ErrorTypeV1, FinalTranscriptStability, FireworksModel, GeminiModel, GladiaModel, GoogleModel, Language, MistralVoxtralModel, OpenAIModel, OpenAIRealtimeModel, RECOGNITION_CONDUCTOR_BASES, RECOGNITION_SERVICE_BASES, RealTimeTwoWayWebSocketRecognitionClient, RecognitionContextTypeV1, RecognitionError, RecognitionProvider, RecognitionResultTypeV1, RecognitionVGFStateSchema, RecordingStatus, STAGES, SampleRate, SelfServeVllmModel, SimplifiedVGFRecognitionClient, TimeoutError, TranscriptionStatus, ValidationError, createClient, createClientWithBuilder, createDefaultASRConfig, createInitialRecognitionState, createSimplifiedVGFClient, getRecognitionConductorBase, getRecognitionConductorHost, getRecognitionConductorHttpBase, getRecognitionConductorWsBase, getRecognitionServiceBase, getRecognitionServiceHost, getRecognitionServiceHttpBase, getRecognitionServiceWsBase, getUserFriendlyMessage, isExceptionImmediatelyAvailable, isNormalDisconnection, isValidRecordingStatusTransition, normalizeStage, resetRecognitionVGFState };
 export type { ASRRequestConfig, ASRRequestV1, AudioMetricsResultV1, AuthenticationException, ConnectionException, ErrorResultV1, FunctionCallResultV1, GameContextV1, IRecognitionClient, IRecognitionClientConfig, IRecognitionClientStats, ISimplifiedVGFRecognitionClient, MetadataResultV1, ProviderException, QuotaExceededException, RealTimeTwoWayWebSocketRecognitionClientConfig, RecognitionCallbackUrl, RecognitionException, RecognitionState, RecordingStatusType, SimplifiedVGFClientConfig, SlotMap, Stage, TimeoutException, TranscriptionResult, TranscriptionResultV1, TranscriptionStatusType, UnknownException, ValidationException };

package/dist/index.d.ts

@@ -11,6 +11,6 @@ export { type RecognitionState, RecognitionVGFStateSchema, RecordingStatus, Tran
 export { resetRecognitionVGFState } from './vgf-recognition-mapper.js';
 export { AudioEncoding } from '@recog/websocket';
 export { type GameContextV1, type SlotMap, RecognitionContextTypeV1, ControlSignalTypeV1, ControlSignalTypeV1 as ControlSignal, // Alias for backward compatibility
-type TranscriptionResultV1, type FunctionCallResultV1, type MetadataResultV1, type AudioMetricsResultV1, type ErrorResultV1, RecognitionResultTypeV1, ClientControlActionV1, type ASRRequestConfig, type ASRRequestV1, FinalTranscriptStability, createDefaultASRConfig, RecognitionProvider, DeepgramModel, ElevenLabsModel, FireworksModel, GladiaModel, GoogleModel, GeminiModel, OpenAIModel, SelfServeVllmModel, OpenAIRealtimeModel, MistralVoxtralModel, CartesiaModel, DashScopeModel, BedrockModel, AwsTranscribeModel, Language, SampleRate, STAGES, type Stage } from '@recog/shared-types';
+type TranscriptionResultV1, type FunctionCallResultV1, type MetadataResultV1, type AudioMetricsResultV1, type ErrorResultV1, RecognitionResultTypeV1, ClientControlActionV1, type ASRRequestConfig, type ASRRequestV1, FinalTranscriptStability, createDefaultASRConfig, RecognitionProvider, DeepgramModel, ElevenLabsModel, FireworksModel, GladiaModel, GoogleModel, GeminiModel, OpenAIModel, SelfServeVllmModel, OpenAIRealtimeModel, MistralVoxtralModel, CartesiaModel, DashScopeModel, BedrockModel, AwsTranscribeModel, AmazonNovaSonicModel, Language, SampleRate, STAGES, type Stage } from '@recog/shared-types';
 export { getRecognitionServiceBase, getRecognitionServiceHttpBase, getRecognitionServiceWsBase, getRecognitionServiceHost, getRecognitionConductorBase, getRecognitionConductorHttpBase, getRecognitionConductorWsBase, getRecognitionConductorHost, normalizeStage, RECOGNITION_SERVICE_BASES, RECOGNITION_CONDUCTOR_BASES } from '@recog/shared-config';
 //# sourceMappingURL=index.d.ts.map

package/dist/index.d.ts.map

@@ -1 +1 @@
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