@elevenlabs/client 0.15.1 → 1.0.0-rc.0

package/dist/platform/VoiceSessionSetup.d.ts

@@ -0,0 +1,35 @@
+import type { Options } from "../BaseConversation";
+import type { BaseConnection } from "../utils/BaseConnection";
+import type { InputController } from "../InputController";
+import type { OutputController } from "../OutputController";
+import { type PlaybackEventTarget } from "../utils/output";
+export type VoiceSessionSetupResult = {
+    connection: BaseConnection;
+    input: InputController;
+    output: OutputController;
+    playbackEventTarget: PlaybackEventTarget | null;
+    detach: () => void;
+};
+export type VoiceSessionSetupStrategy = (options: Options) => Promise<VoiceSessionSetupResult>;
+/**
+ * Sets up input and output controllers for an existing connection.
+ * Shared helper used by platform-specific setup strategies.
+ */
+export declare function setupInputOutput(options: Options, connection: BaseConnection): Promise<Omit<VoiceSessionSetupResult, "connection">>;
+/**
+ * Web platform session setup strategy.
+ * Creates a connection and sets up input/output based on the connection type.
+ */
+export declare function webSessionSetup(options: Options): Promise<VoiceSessionSetupResult>;
+/**
+ * The active session setup strategy.
+ * Defaults to web platform strategy.
+ * Can be overridden by platform-specific entrypoints (e.g. React Native).
+ */
+export declare let setupStrategy: VoiceSessionSetupStrategy;
+/**
+ * Override the voice session setup strategy.
+ * Called by platform-specific entrypoints to inject their own setup handling.
+ */
+export declare function setSetupStrategy(strategy: VoiceSessionSetupStrategy): void;
+//# sourceMappingURL=VoiceSessionSetup.d.ts.map

package/dist/platform/VoiceSessionSetup.d.ts.map

@@ -0,0 +1 @@
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package/dist/platform/VoiceSessionSetup.js

@@ -0,0 +1,74 @@
+import { MediaDeviceOutput } from "../utils/output";
+import { MediaDeviceInput } from "../utils/input";
+import { WebSocketConnection } from "../utils/WebSocketConnection";
+import { WebRTCConnection } from "../utils/WebRTCConnection";
+import { attachInputToConnection } from "../utils/attachInputToConnection";
+import { attachConnectionToOutput } from "../utils/attachConnectionToOutput";
+import { createConnection } from "../utils/ConnectionFactory";
+/**
+ * Sets up input and output controllers for an existing connection.
+ * Shared helper used by platform-specific setup strategies.
+ */
+export async function setupInputOutput(options, connection) {
+    if (connection instanceof WebRTCConnection) {
+        return {
+            input: connection.input,
+            output: connection.output,
+            playbackEventTarget: null,
+            detach: () => { },
+        };
+    }
+    else if (connection instanceof WebSocketConnection) {
+        const [input, output] = await Promise.all([
+            MediaDeviceInput.create({
+                ...connection.inputFormat,
+                preferHeadphonesForIosDevices: options.preferHeadphonesForIosDevices,
+                inputDeviceId: options.inputDeviceId,
+                workletPaths: options.workletPaths,
+                libsampleratePath: options.libsampleratePath,
+            }),
+            MediaDeviceOutput.create({
+                ...connection.outputFormat,
+                outputDeviceId: options.outputDeviceId,
+                workletPaths: options.workletPaths,
+            }),
+        ]);
+        const detachInput = attachInputToConnection(input, connection);
+        const detachOutput = attachConnectionToOutput(connection, output);
+        return {
+            input,
+            output,
+            playbackEventTarget: output,
+            detach: () => {
+                detachInput();
+                detachOutput();
+            },
+        };
+    }
+    else {
+        throw new Error(`Unsupported connection type: ${connection.constructor.name}`);
+    }
+}
+/**
+ * Web platform session setup strategy.
+ * Creates a connection and sets up input/output based on the connection type.
+ */
+export async function webSessionSetup(options) {
+    const connection = await createConnection(options);
+    const io = await setupInputOutput(options, connection);
+    return { connection, ...io };
+}
+/**
+ * The active session setup strategy.
+ * Defaults to web platform strategy.
+ * Can be overridden by platform-specific entrypoints (e.g. React Native).
+ */
+export let setupStrategy = webSessionSetup;
+/**
+ * Override the voice session setup strategy.
+ * Called by platform-specific entrypoints to inject their own setup handling.
+ */
+export function setSetupStrategy(strategy) {
+    setupStrategy = strategy;
+}
+//# sourceMappingURL=VoiceSessionSetup.js.map

package/dist/platform/VoiceSessionSetup.js.map

@@ -0,0 +1 @@
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package/dist/scribe/connection.d.ts

@@ -221,3 +221,4 @@ export declare class RealtimeConnection {
      */
     close(): void;
 }
+//# sourceMappingURL=connection.d.ts.map

package/dist/scribe/connection.d.ts.map

@@ -0,0 +1 @@
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package/dist/scribe/connection.js

@@ -0,0 +1,363 @@
+/**
+ * Simple EventEmitter implementation for browser compatibility.
+ */
+class EventEmitter {
+    listeners = new Map();
+    on(event, listener) {
+        if (!this.listeners.has(event)) {
+            this.listeners.set(event, new Set());
+        }
+        const eventListeners = this.listeners.get(event);
+        if (eventListeners) {
+            eventListeners.add(listener);
+        }
+    }
+    off(event, listener) {
+        const eventListeners = this.listeners.get(event);
+        if (eventListeners) {
+            eventListeners.delete(listener);
+        }
+    }
+    emit(event, ...args) {
+        const eventListeners = this.listeners.get(event);
+        if (eventListeners) {
+            eventListeners.forEach(listener => {
+                listener(...args);
+            });
+        }
+    }
+}
+/**
+ * Events emitted by the RealtimeConnection.
+ */
+export var RealtimeEvents;
+(function (RealtimeEvents) {
+    /** Emitted when the session is successfully started */
+    RealtimeEvents["SESSION_STARTED"] = "session_started";
+    /** Emitted when a partial (interim) transcript is available */
+    RealtimeEvents["PARTIAL_TRANSCRIPT"] = "partial_transcript";
+    /** Emitted when a final transcript is available */
+    RealtimeEvents["COMMITTED_TRANSCRIPT"] = "committed_transcript";
+    /** Emitted when a final transcript with timestamps is available */
+    RealtimeEvents["COMMITTED_TRANSCRIPT_WITH_TIMESTAMPS"] = "committed_transcript_with_timestamps";
+    /** Emitted when an authentication error occurs */
+    RealtimeEvents["AUTH_ERROR"] = "auth_error";
+    /** Emitted when an error occurs (also emitted for all specific error types) */
+    RealtimeEvents["ERROR"] = "error";
+    /** Emitted when the WebSocket connection is opened */
+    RealtimeEvents["OPEN"] = "open";
+    /** Emitted when the WebSocket connection is closed */
+    RealtimeEvents["CLOSE"] = "close";
+    /** Emitted when a quota exceeded error occurs */
+    RealtimeEvents["QUOTA_EXCEEDED"] = "quota_exceeded";
+    /** Emitted when commit is throttled */
+    RealtimeEvents["COMMIT_THROTTLED"] = "commit_throttled";
+    /** Emitted when a transcriber error occurs */
+    RealtimeEvents["TRANSCRIBER_ERROR"] = "transcriber_error";
+    /** Emitted when terms have not been accepted */
+    RealtimeEvents["UNACCEPTED_TERMS"] = "unaccepted_terms";
+    /** Emitted when rate limited */
+    RealtimeEvents["RATE_LIMITED"] = "rate_limited";
+    /** Emitted when there's an input error */
+    RealtimeEvents["INPUT_ERROR"] = "input_error";
+    /** Emitted when the queue overflows */
+    RealtimeEvents["QUEUE_OVERFLOW"] = "queue_overflow";
+    /** Emitted when resources are exhausted */
+    RealtimeEvents["RESOURCE_EXHAUSTED"] = "resource_exhausted";
+    /** Emitted when session time limit is exceeded */
+    RealtimeEvents["SESSION_TIME_LIMIT_EXCEEDED"] = "session_time_limit_exceeded";
+    /** Emitted when chunk size is exceeded */
+    RealtimeEvents["CHUNK_SIZE_EXCEEDED"] = "chunk_size_exceeded";
+    /** Emitted when there's insufficient audio activity */
+    RealtimeEvents["INSUFFICIENT_AUDIO_ACTIVITY"] = "insufficient_audio_activity";
+})(RealtimeEvents || (RealtimeEvents = {}));
+/**
+ * Manages a real-time transcription WebSocket connection.
+ *
+ * @example
+ * ```typescript
+ * const connection = await Scribe.connect({
+ *     token: "...",
+ *     modelId: "scribe_v2_realtime",
+ *     audioFormat: AudioFormat.PCM_16000,
+ *     sampleRate: 16000,
+ * });
+ *
+ * connection.on(RealtimeEvents.SESSION_STARTED, (data) => {
+ *     console.log("Session started");
+ * });
+ *
+ * connection.on(RealtimeEvents.PARTIAL_TRANSCRIPT, (data) => {
+ *     console.log("Partial:", data.transcript);
+ * });
+ *
+ * connection.on(RealtimeEvents.COMMITTED_TRANSCRIPT, (data) => {
+ *     console.log("Final:", data.transcript);
+ *     connection.close();
+ * });
+ *
+ * // Send audio data
+ * connection.send({ audioBase64: base64String });
+ *
+ * // Commit and close
+ * connection.commit();
+ * ```
+ */
+export class RealtimeConnection {
+    websocket = null;
+    eventEmitter = new EventEmitter();
+    currentSampleRate = 16000;
+    _audioCleanup;
+    constructor(sampleRate) {
+        this.currentSampleRate = sampleRate;
+    }
+    /**
+     * @internal
+     * Used internally by ScribeRealtime to attach the WebSocket after connection is created.
+     */
+    setWebSocket(websocket) {
+        this.websocket = websocket;
+        // If WebSocket is already open, emit OPEN event immediately
+        if (this.websocket.readyState === WebSocket.OPEN) {
+            this.eventEmitter.emit(RealtimeEvents.OPEN);
+        }
+        else {
+            // Otherwise, wait for the open event
+            this.websocket.addEventListener("open", () => {
+                this.eventEmitter.emit(RealtimeEvents.OPEN);
+            });
+        }
+        this.websocket.addEventListener("message", (event) => {
+            try {
+                const data = JSON.parse(event.data);
+                switch (data.message_type) {
+                    case "session_started":
+                        this.eventEmitter.emit(RealtimeEvents.SESSION_STARTED, data);
+                        break;
+                    case "partial_transcript":
+                        this.eventEmitter.emit(RealtimeEvents.PARTIAL_TRANSCRIPT, data);
+                        break;
+                    case "committed_transcript":
+                        this.eventEmitter.emit(RealtimeEvents.COMMITTED_TRANSCRIPT, data);
+                        break;
+                    case "committed_transcript_with_timestamps":
+                        this.eventEmitter.emit(RealtimeEvents.COMMITTED_TRANSCRIPT_WITH_TIMESTAMPS, data);
+                        break;
+                    // Error cases - emit both specific event and generic ERROR
+                    case "auth_error":
+                        this.eventEmitter.emit(RealtimeEvents.AUTH_ERROR, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "quota_exceeded":
+                        this.eventEmitter.emit(RealtimeEvents.QUOTA_EXCEEDED, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "commit_throttled":
+                        this.eventEmitter.emit(RealtimeEvents.COMMIT_THROTTLED, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "transcriber_error":
+                        this.eventEmitter.emit(RealtimeEvents.TRANSCRIBER_ERROR, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "unaccepted_terms":
+                        this.eventEmitter.emit(RealtimeEvents.UNACCEPTED_TERMS, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "rate_limited":
+                        this.eventEmitter.emit(RealtimeEvents.RATE_LIMITED, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "input_error":
+                        this.eventEmitter.emit(RealtimeEvents.INPUT_ERROR, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "queue_overflow":
+                        this.eventEmitter.emit(RealtimeEvents.QUEUE_OVERFLOW, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "resource_exhausted":
+                        this.eventEmitter.emit(RealtimeEvents.RESOURCE_EXHAUSTED, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "session_time_limit_exceeded":
+                        this.eventEmitter.emit(RealtimeEvents.SESSION_TIME_LIMIT_EXCEEDED, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "chunk_size_exceeded":
+                        this.eventEmitter.emit(RealtimeEvents.CHUNK_SIZE_EXCEEDED, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "insufficient_audio_activity":
+                        this.eventEmitter.emit(RealtimeEvents.INSUFFICIENT_AUDIO_ACTIVITY, data);
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    case "error":
+                        this.eventEmitter.emit(RealtimeEvents.ERROR, data);
+                        break;
+                    default:
+                        console.warn("Unknown message type:", data);
+                }
+            }
+            catch (error) {
+                console.error("Failed to parse WebSocket message:", error, event.data);
+                this.eventEmitter.emit(RealtimeEvents.ERROR, new Error(`Failed to parse message: ${error}`));
+            }
+        });
+        this.websocket.addEventListener("error", (error) => {
+            console.error("WebSocket error:", error);
+            this.eventEmitter.emit(RealtimeEvents.ERROR, error);
+        });
+        this.websocket.addEventListener("close", (event) => {
+            console.log(`WebSocket closed: code=${event.code}, reason="${event.reason}", wasClean=${event.wasClean}`);
+            // Emit error if close was not clean or had an error code
+            if (!event.wasClean || (event.code !== 1000 && event.code !== 1005)) {
+                const errorMessage = `WebSocket closed unexpectedly: ${event.code} - ${event.reason || "No reason provided"}`;
+                console.error(errorMessage);
+                this.eventEmitter.emit(RealtimeEvents.ERROR, new Error(errorMessage));
+            }
+            this.eventEmitter.emit(RealtimeEvents.CLOSE, event);
+        });
+    }
+    /**
+     * Attaches an event listener for the specified event.
+     *
+     * @param event - The event to listen for (use RealtimeEvents enum)
+     * @param listener - The callback function to execute when the event fires
+     *
+     * @example
+     * ```typescript
+     * connection.on(RealtimeEvents.SESSION_STARTED, (data) => {
+     *     console.log("Session started", data.session_id);
+     * });
+     *
+     * connection.on(RealtimeEvents.PARTIAL_TRANSCRIPT, (data) => {
+     *     console.log("Partial:", data.text);
+     * });
+     *
+     * connection.on(RealtimeEvents.COMMITTED_TRANSCRIPT, (data) => {
+     *     console.log("Final:", data.text);
+     * });
+     * ```
+     */
+    on(event, listener) {
+        this.eventEmitter.on(event, listener);
+    }
+    /**
+     * Removes an event listener for the specified event.
+     *
+     * @param event - The event to stop listening for
+     * @param listener - The callback function to remove
+     *
+     * @example
+     * ```typescript
+     * const handler = (data: PartialTranscriptMessage) => console.log(data.text);
+     * connection.on(RealtimeEvents.PARTIAL_TRANSCRIPT, handler);
+     *
+     * // Later, remove the listener
+     * connection.off(RealtimeEvents.PARTIAL_TRANSCRIPT, handler);
+     * ```
+     */
+    off(event, listener) {
+        this.eventEmitter.off(event, listener);
+    }
+    /**
+     * Sends audio data to the transcription service.
+     *
+     * @param data - Audio data configuration
+     * @param data.audioBase64 - Base64-encoded audio data
+     * @param data.commit - Whether to commit the transcription after this chunk. You likely want to use connection.commit() instead (default: false)
+     * @param data.sampleRate - Sample rate of the audio (default: configured sample rate)
+     * @param data.previousText - Send context to the model via base64 encoded audio or text from a previous transcription. Can only be sent alongside the first audio chunk. If sent in a subsequent chunk, an error will be returned.
+     *
+     * @throws {Error} If the WebSocket connection is not open
+     *
+     * @example
+     * ```typescript
+     * // Send audio chunk without committing
+     * connection.send({
+     *     audioBase64: base64EncodedAudio,
+     * });
+     *
+     * // Send audio chunk with custom sample rate and previous text
+     * connection.send({
+     *     audioBase64: base64EncodedAudio,
+     *     sampleRate: 16000,
+     *     previousText: "Previous transcription text",
+     * });
+     * ```
+     */
+    send(data) {
+        if (!this.websocket || this.websocket.readyState !== WebSocket.OPEN) {
+            throw new Error("WebSocket is not connected");
+        }
+        const message = {
+            message_type: "input_audio_chunk",
+            audio_base_64: data.audioBase64,
+            commit: data.commit ?? false,
+            sample_rate: data.sampleRate ?? this.currentSampleRate,
+            previous_text: data.previousText,
+        };
+        this.websocket.send(JSON.stringify(message));
+    }
+    /**
+     * Commits the transcription, signaling that a segment of audio has been sent. This clears the buffer and triggers a COMMITTED_TRANSCRIPT event. Context from previous segments is kept.
+     * Committing a segment triggers a COMMITTED_TRANSCRIPT event.
+     *
+     * @throws {Error} If the WebSocket connection is not open
+     *
+     * @remarks
+     * Only needed when using CommitStrategy.MANUAL.
+     * When using CommitStrategy.VAD, commits are handled automatically by the server.
+     *
+     * @example
+     * ```typescript
+     * // Send all audio chunks
+     * for (const chunk of audioChunks) {
+     *     connection.send({ audioBase64: chunk });
+     * }
+     *
+     * // Finalize the transcription
+     * connection.commit();
+     * ```
+     */
+    commit() {
+        if (!this.websocket || this.websocket.readyState !== WebSocket.OPEN) {
+            throw new Error("WebSocket is not connected");
+        }
+        const message = {
+            message_type: "input_audio_chunk",
+            audio_base_64: "",
+            commit: true,
+            sample_rate: this.currentSampleRate,
+        };
+        this.websocket.send(JSON.stringify(message));
+    }
+    /**
+     * Closes the WebSocket connection and cleans up resources.
+     * This will terminate any ongoing transcription and stop microphone streaming if active.
+     *
+     * @remarks
+     * After calling close(), this connection cannot be reused.
+     * Create a new connection if you need to start transcribing again.
+     *
+     * @example
+     * ```typescript
+     * connection.on(RealtimeEvents.COMMITTED_TRANSCRIPT, (data) => {
+     *     console.log("Segment committed:", data.transcript);
+     *     connection.close();
+     * });
+     * ```
+     */
+    close() {
+        // Cleanup audio resources (microphone stream, audio context)
+        if (this._audioCleanup) {
+            this._audioCleanup();
+        }
+        // Close WebSocket connection
+        if (this.websocket) {
+            this.websocket.close(1000, "User ended session");
+        }
+    }
+}
+//# sourceMappingURL=connection.js.map

package/dist/scribe/connection.js.map

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package/dist/scribe/index.d.ts

@@ -5,3 +5,4 @@ export { RealtimeEvents } from "./connection";
 export type { RealtimeEventMap } from "./connection";
 export type { AudioOptions, MicrophoneOptions } from "./scribe";
 export type { WebSocketMessage, SessionStartedMessage, PartialTranscriptMessage, CommittedTranscriptMessage, CommittedTranscriptWithTimestampsMessage, ScribeErrorMessage, ScribeAuthErrorMessage, ScribeQuotaExceededErrorMessage, ScribeCommitThrottledErrorMessage, ScribeTranscriberErrorMessage, ScribeUnacceptedTermsErrorMessage, ScribeRateLimitedErrorMessage, ScribeInputErrorMessage, ScribeQueueOverflowErrorMessage, ScribeResourceExhaustedErrorMessage, ScribeSessionTimeLimitExceededErrorMessage, ScribeChunkSizeExceededErrorMessage, ScribeInsufficientAudioActivityErrorMessage, } from "./connection";
+//# sourceMappingURL=index.d.ts.map

package/dist/scribe/index.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"index.d.ts","sourceRoot":"","sources":["../../src/scribe/index.ts"],"names":[],"mappings":"AACA,OAAO,EAAE,cAAc,IAAI,MAAM,EAAE,MAAM,UAAU,CAAC;AAGpD,OAAO,EAAE,kBAAkB,EAAE,MAAM,cAAc,CAAC;AAGlD,OAAO,EAAE,WAAW,EAAE,cAAc,EAAE,MAAM,UAAU,CAAC;AACvD,OAAO,EAAE,cAAc,EAAE,MAAM,cAAc,CAAC;AAC9C,YAAY,EAAE,gBAAgB,EAAE,MAAM,cAAc,CAAC;AAGrD,YAAY,EAAE,YAAY,EAAE,iBAAiB,EAAE,MAAM,UAAU,CAAC;AAChE,YAAY,EACV,gBAAgB,EAChB,qBAAqB,EACrB,wBAAwB,EACxB,0BAA0B,EAC1B,wCAAwC,EACxC,kBAAkB,EAClB,sBAAsB,EACtB,+BAA+B,EAC/B,iCAAiC,EACjC,6BAA6B,EAC7B,iCAAiC,EACjC,6BAA6B,EAC7B,uBAAuB,EACvB,+BAA+B,EAC/B,mCAAmC,EACnC,0CAA0C,EAC1C,mCAAmC,EACnC,2CAA2C,GAC5C,MAAM,cAAc,CAAC"}

package/dist/scribe/index.js

@@ -0,0 +1,8 @@
+// Export the main Scribe class (renamed from ScribeRealtime)
+export { ScribeRealtime as Scribe } from "./scribe";
+// Export connection class
+export { RealtimeConnection } from "./connection";
+// Export enums
+export { AudioFormat, CommitStrategy } from "./scribe";
+export { RealtimeEvents } from "./connection";
+//# sourceMappingURL=index.js.map

package/dist/scribe/index.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../src/scribe/index.ts"],"names":[],"mappings":"AAAA,6DAA6D;AAC7D,OAAO,EAAE,cAAc,IAAI,MAAM,EAAE,MAAM,UAAU,CAAC;AAEpD,0BAA0B;AAC1B,OAAO,EAAE,kBAAkB,EAAE,MAAM,cAAc,CAAC;AAElD,eAAe;AACf,OAAO,EAAE,WAAW,EAAE,cAAc,EAAE,MAAM,UAAU,CAAC;AACvD,OAAO,EAAE,cAAc,EAAE,MAAM,cAAc,CAAC"}

package/dist/scribe/scribe.d.ts

@@ -121,3 +121,4 @@ export declare class ScribeRealtime {
     private static streamFromMicrophone;
 }
 export {};
+//# sourceMappingURL=scribe.d.ts.map

package/dist/scribe/scribe.d.ts.map

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