fix(#388): address PR review findings — fix WebSocket/REST bugs, improve error handling, fix types and comments
Critical fixes: - Fix FormData field name mismatch (audio -> file) to match backend FileInterceptor - Add /speech namespace to WebSocket connection URL - Pass auth token in WebSocket handshake options - Wrap audio.play() in try-catch for NotAllowedError and DOMException handling - Replace bare catch block with named error parameter and descriptive message - Add connect_error and disconnect event handlers to WebSocket - Update JSDoc to accurately describe batch transcription (not real-time partial) Important fixes: - Emit transcription-error before disconnect in gateway auth failures - Capture MediaRecorder error details and clean up media tracks on error - Change TtsDefaultConfig.format type from string to AudioFormat - Define canonical SPEECH_TIERS and AUDIO_FORMATS arrays as single source of truth - Fix voice count from 54 to 53 in provider, AGENTS.md, and docs - Fix inaccurate comments (Piper formats, tier prop, SpeachesProvider, TextValidationPipe) Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
This commit is contained in:
@@ -34,7 +34,7 @@ speech/
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└── providers/
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├── base-tts.provider.ts # Abstract base class (OpenAI SDK + common logic)
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├── base-tts.provider.spec.ts
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├── kokoro-tts.provider.ts # Default tier (CPU, 54 voices, 8 languages)
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├── kokoro-tts.provider.ts # Default tier (CPU, 53 voices, 8 languages)
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├── kokoro-tts.provider.spec.ts
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├── chatterbox-tts.provider.ts # Premium tier (GPU, voice cloning, emotion control)
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├── chatterbox-tts.provider.spec.ts
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@@ -2,7 +2,7 @@
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* SynthesizeDto
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*
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* DTO for text-to-speech synthesis requests.
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* The text field is validated by TextValidationPipe for length/emptiness.
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* Text and option fields are validated by class-validator decorators.
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* Additional options control voice, speed, format, and tier selection.
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*
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* Issue #398
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@@ -10,29 +10,13 @@
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import { IsString, IsOptional, IsNumber, IsIn, Min, Max, MaxLength } from "class-validator";
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import { Type } from "class-transformer";
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import { AUDIO_FORMATS, SPEECH_TIERS } from "../interfaces/speech-types";
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import type { AudioFormat, SpeechTier } from "../interfaces/speech-types";
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/**
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* Valid audio output formats for TTS synthesis.
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*/
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const VALID_AUDIO_FORMATS: readonly AudioFormat[] = [
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"mp3",
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"wav",
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"opus",
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"flac",
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"aac",
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"pcm",
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] as const;
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/**
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* Valid TTS tiers for provider selection.
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*/
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const VALID_SPEECH_TIERS: readonly SpeechTier[] = ["default", "premium", "fallback"] as const;
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export class SynthesizeDto {
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/**
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* Text to convert to speech.
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* Validated separately by TextValidationPipe for length and emptiness.
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* Validated by class-validator decorators for type and maximum length.
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*/
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@IsString({ message: "text must be a string" })
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@MaxLength(4096, { message: "text must not exceed 4096 characters" })
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@@ -66,8 +50,8 @@ export class SynthesizeDto {
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*/
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@IsOptional()
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@IsString({ message: "format must be a string" })
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@IsIn(VALID_AUDIO_FORMATS, {
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message: `format must be one of: ${VALID_AUDIO_FORMATS.join(", ")}`,
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@IsIn(AUDIO_FORMATS, {
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message: `format must be one of: ${AUDIO_FORMATS.join(", ")}`,
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})
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format?: AudioFormat;
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@@ -78,8 +62,8 @@ export class SynthesizeDto {
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*/
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@IsOptional()
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@IsString({ message: "tier must be a string" })
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@IsIn(VALID_SPEECH_TIERS, {
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message: `tier must be one of: ${VALID_SPEECH_TIERS.join(", ")}`,
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@IsIn(SPEECH_TIERS, {
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message: `tier must be one of: ${SPEECH_TIERS.join(", ")}`,
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})
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tier?: SpeechTier;
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}
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@@ -6,6 +6,7 @@
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export type { ISTTProvider } from "./stt-provider.interface";
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export type { ITTSProvider } from "./tts-provider.interface";
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export { SPEECH_TIERS, AUDIO_FORMATS } from "./speech-types";
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export type {
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SpeechTier,
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AudioFormat,
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@@ -12,19 +12,21 @@
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// ==========================================
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/**
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* TTS provider tier.
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* Canonical array of TTS provider tiers.
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* Determines which TTS engine is used for synthesis.
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*
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* - default: Primary TTS engine (e.g., Kokoro)
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* - premium: Higher quality TTS engine (e.g., Chatterbox)
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* - fallback: Backup TTS engine (e.g., Piper/OpenedAI)
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*/
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export type SpeechTier = "default" | "premium" | "fallback";
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export const SPEECH_TIERS = ["default", "premium", "fallback"] as const;
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export type SpeechTier = (typeof SPEECH_TIERS)[number];
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/**
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* Audio output format for TTS synthesis.
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* Canonical array of audio output formats for TTS synthesis.
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*/
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export type AudioFormat = "mp3" | "wav" | "opus" | "flac" | "aac" | "pcm";
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export const AUDIO_FORMATS = ["mp3", "wav", "opus", "flac", "aac", "pcm"] as const;
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export type AudioFormat = (typeof AUDIO_FORMATS)[number];
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// ==========================================
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// STT Types
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@@ -16,7 +16,7 @@ import type { TranscribeOptions, TranscriptionResult } from "./speech-types";
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*
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* @example
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* ```typescript
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* class SpeachesProvider implements ISTTProvider {
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* class SpeachesSttProvider implements ISTTProvider {
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* readonly name = "speaches";
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*
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* async transcribe(audio: Buffer, options?: TranscribeOptions): Promise<TranscriptionResult> {
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@@ -5,7 +5,7 @@
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* CPU-based, always available, Apache 2.0 license.
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*
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* Features:
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* - 54 built-in voices across 8 languages
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* - 53 built-in voices across 8 languages
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* - Speed control: 0.25x to 4.0x
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* - Output formats: mp3, wav, opus, flac
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* - Voice metadata derived from ID prefix (language, gender, accent)
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@@ -222,7 +222,7 @@ export function parseVoicePrefix(voiceId: string): VoicePrefixMetadata {
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/**
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* Kokoro-FastAPI TTS provider (default tier).
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*
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* CPU-based text-to-speech engine with 54 built-in voices across 8 languages.
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* CPU-based text-to-speech engine with 53 built-in voices across 8 languages.
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* Uses the OpenAI-compatible API exposed by Kokoro-FastAPI.
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*
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* @example
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@@ -254,7 +254,7 @@ export class KokoroTtsProvider extends BaseTTSProvider {
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/**
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* List all available Kokoro voices with metadata.
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*
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* Returns the full catalog of 54 built-in voices with language, gender,
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* Returns the full catalog of 53 built-in voices with language, gender,
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* and accent information derived from voice ID prefixes.
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*
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* @returns Array of VoiceInfo objects for all Kokoro voices
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@@ -9,7 +9,7 @@
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* - OpenAI-compatible API via OpenedAI Speech server
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* - 100+ Piper voices across 40+ languages
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* - 6 standard OpenAI voice names mapped to Piper voices
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* - Output formats: mp3, wav, opus, flac, aac, pcm
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* - Output formats: mp3, wav, opus, flac
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* - CPU-only, no GPU required
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* - GPL license (via OpenedAI Speech)
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*
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@@ -18,7 +18,7 @@ import { ChatterboxTTSProvider } from "./chatterbox-tts.provider";
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import { KokoroTtsProvider } from "./kokoro-tts.provider";
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import { PiperTtsProvider } from "./piper-tts.provider";
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import type { ITTSProvider } from "../interfaces/tts-provider.interface";
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import type { SpeechTier, AudioFormat } from "../interfaces/speech-types";
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import type { SpeechTier } from "../interfaces/speech-types";
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import type { SpeechConfig } from "../speech.config";
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// ==========================================
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@@ -44,7 +44,7 @@ export function createTTSProviders(config: SpeechConfig): Map<SpeechTier, ITTSPr
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const provider = new KokoroTtsProvider(
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config.tts.default.url,
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config.tts.default.voice,
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config.tts.default.format as AudioFormat
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config.tts.default.format
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);
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providers.set("default", provider);
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logger.log(`Registered default TTS provider: kokoro at ${config.tts.default.url}`);
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@@ -12,6 +12,7 @@
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*/
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import { registerAs } from "@nestjs/config";
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import type { AudioFormat } from "./interfaces/speech-types";
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// ==========================================
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// Default values
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@@ -58,7 +59,7 @@ export interface TtsDefaultConfig {
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enabled: boolean;
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url: string;
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voice: string;
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format: string;
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format: AudioFormat;
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}
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export interface TtsPremiumConfig {
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@@ -247,7 +248,7 @@ export function getSpeechConfig(): SpeechConfig {
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enabled: isTtsEnabled(),
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url: process.env.TTS_DEFAULT_URL ?? TTS_DEFAULT_DEFAULTS.url,
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voice: process.env.TTS_DEFAULT_VOICE ?? TTS_DEFAULT_DEFAULTS.voice,
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format: process.env.TTS_DEFAULT_FORMAT ?? TTS_DEFAULT_DEFAULTS.format,
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format: (process.env.TTS_DEFAULT_FORMAT ?? TTS_DEFAULT_DEFAULTS.format) as AudioFormat,
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},
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premium: {
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enabled: isTtsPremiumEnabled(),
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@@ -100,6 +100,9 @@ export class SpeechGateway implements OnGatewayConnection, OnGatewayDisconnect {
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const timeoutId = setTimeout(() => {
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if (!authenticatedClient.data.userId) {
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this.logger.warn(`Client ${authenticatedClient.id} timed out during authentication`);
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authenticatedClient.emit("transcription-error", {
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message: "Authentication timed out.",
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});
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authenticatedClient.disconnect();
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}
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}, this.CONNECTION_TIMEOUT_MS);
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@@ -109,6 +112,9 @@ export class SpeechGateway implements OnGatewayConnection, OnGatewayDisconnect {
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if (!token) {
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this.logger.warn(`Client ${authenticatedClient.id} connected without token`);
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authenticatedClient.emit("transcription-error", {
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message: "Authentication failed: no token provided.",
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});
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authenticatedClient.disconnect();
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clearTimeout(timeoutId);
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return;
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@@ -118,6 +124,9 @@ export class SpeechGateway implements OnGatewayConnection, OnGatewayDisconnect {
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if (!sessionData) {
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this.logger.warn(`Client ${authenticatedClient.id} has invalid token`);
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authenticatedClient.emit("transcription-error", {
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message: "Authentication failed: invalid or expired token.",
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});
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authenticatedClient.disconnect();
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clearTimeout(timeoutId);
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return;
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@@ -133,6 +142,9 @@ export class SpeechGateway implements OnGatewayConnection, OnGatewayDisconnect {
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if (!workspaceMembership) {
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this.logger.warn(`User ${userId} has no workspace access`);
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authenticatedClient.emit("transcription-error", {
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message: "Authentication failed: no workspace access.",
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});
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authenticatedClient.disconnect();
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clearTimeout(timeoutId);
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return;
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@@ -151,6 +163,9 @@ export class SpeechGateway implements OnGatewayConnection, OnGatewayDisconnect {
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`Authentication failed for speech client ${authenticatedClient.id}:`,
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error instanceof Error ? error.message : "Unknown error"
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);
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authenticatedClient.emit("transcription-error", {
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message: "Authentication failed: an unexpected error occurred.",
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});
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authenticatedClient.disconnect();
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}
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}
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@@ -19,7 +19,7 @@ export interface TextToSpeechButtonProps {
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text: string;
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/** Optional voice ID to use */
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voice?: string;
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/** Optional tier (e.g. "standard", "premium") */
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/** Optional tier (e.g. "default", "premium", "fallback") */
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tier?: string;
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/** Optional className for the container */
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className?: string;
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@@ -173,8 +173,17 @@ export function useTextToSpeech(): UseTextToSpeechReturn {
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const play = useCallback(async (): Promise<void> => {
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const audio = audioRef.current;
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if (audio) {
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await audio.play();
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setIsPlaying(true);
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try {
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await audio.play();
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setIsPlaying(true);
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} catch (err) {
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const message =
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err instanceof DOMException && err.name === "NotAllowedError"
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? "Playback was blocked by the browser. Try interacting with the page first."
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: "Unable to play audio. The format may not be supported.";
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setError(message);
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setIsPlaying(false);
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}
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}
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}, []);
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@@ -1,8 +1,8 @@
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/**
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* useVoiceInput hook
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*
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* Custom hook for microphone capture and real-time transcription.
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* Supports WebSocket streaming for real-time partial transcriptions
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* Custom hook for microphone capture and speech-to-text transcription.
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* Supports WebSocket streaming with batch transcription on stop,
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* with REST upload fallback when WebSocket is unavailable.
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*/
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@@ -20,6 +20,8 @@ export interface UseVoiceInputOptions {
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useWebSocket?: boolean;
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/** Audio sample rate in Hz (default: 16000) */
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sampleRate?: number;
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/** Authentication token for WebSocket connection */
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token?: string;
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}
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/** Return type for the useVoiceInput hook */
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@@ -75,14 +77,14 @@ function getAudioMimeType(): string {
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}
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/**
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* Hook for microphone capture and real-time speech-to-text transcription.
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* Hook for microphone capture and speech-to-text transcription.
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*
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* Uses WebSocket streaming by default for real-time partial transcriptions.
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* Uses WebSocket streaming by default with batch transcription on stop.
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* Falls back to REST upload (POST /api/speech/transcribe) if WebSocket
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* is disabled or unavailable.
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*/
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export function useVoiceInput(options: UseVoiceInputOptions = {}): UseVoiceInputReturn {
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const { onTranscript, useWebSocket: useWs = true, sampleRate = 16000 } = options;
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const { onTranscript, useWebSocket: useWs = true, sampleRate = 16000, token } = options;
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const [isRecording, setIsRecording] = useState(false);
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const [transcript, setTranscript] = useState("");
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@@ -143,9 +145,12 @@ export function useVoiceInput(options: UseVoiceInputOptions = {}): UseVoiceInput
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};
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animationFrameRef.current = requestAnimationFrame(updateLevel);
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} catch {
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} catch (err) {
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// Audio analysis is non-critical; continue without it
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console.warn("Audio analysis not available");
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console.warn(
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"Audio level visualization unavailable:",
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err instanceof Error ? err.message : String(err)
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);
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}
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}, []);
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@@ -169,11 +174,14 @@ export function useVoiceInput(options: UseVoiceInputOptions = {}): UseVoiceInput
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* Connect to the speech WebSocket namespace
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*/
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const connectSocket = useCallback((): Socket => {
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const socket = io(API_BASE_URL, {
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const socket = io(`${API_BASE_URL}/speech`, {
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path: "/socket.io",
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transports: ["websocket", "polling"],
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...(token ? { auth: { token } } : {}),
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});
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// Future use: the gateway does not currently emit transcription-partial,
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// but the listener is registered for when real-time partial transcription is added.
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socket.on("transcription-partial", (data: TranscriptionPartialPayload) => {
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setPartialTranscript(data.text);
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});
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@@ -188,9 +196,19 @@ export function useVoiceInput(options: UseVoiceInputOptions = {}): UseVoiceInput
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setError(data.message);
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});
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socket.on("connect_error", (err: Error) => {
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setError(`WebSocket connection failed: ${err.message}`);
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});
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socket.on("disconnect", (reason: string) => {
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if (reason !== "io client disconnect") {
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setError(`WebSocket disconnected unexpectedly: ${reason}`);
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}
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});
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socketRef.current = socket;
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return socket;
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}, []);
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}, [token]);
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/**
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* Disconnect the WebSocket
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@@ -200,6 +218,8 @@ export function useVoiceInput(options: UseVoiceInputOptions = {}): UseVoiceInput
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socketRef.current.off("transcription-partial");
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socketRef.current.off("transcription-final");
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socketRef.current.off("transcription-error");
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socketRef.current.off("connect_error");
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socketRef.current.off("disconnect");
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socketRef.current.disconnect();
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socketRef.current = null;
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}
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@@ -211,7 +231,7 @@ export function useVoiceInput(options: UseVoiceInputOptions = {}): UseVoiceInput
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const sendAudioViaRest = useCallback(async (audioBlob: Blob): Promise<void> => {
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try {
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const formData = new FormData();
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formData.append("audio", audioBlob, "recording.webm");
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formData.append("file", audioBlob, "recording.webm");
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const response = await apiPostFormData<TranscribeResponse>(
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"/api/speech/transcribe",
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@@ -315,10 +335,16 @@ export function useVoiceInput(options: UseVoiceInputOptions = {}): UseVoiceInput
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});
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// Handle errors
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mediaRecorder.addEventListener("error", () => {
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setError("Recording encountered an issue. Please try again.");
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mediaRecorder.addEventListener("error", (event: Event) => {
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let errorMessage = "Recording encountered an issue. Please try again.";
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if ("error" in event && event.error instanceof DOMException) {
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errorMessage = `Recording error: ${event.error.name} - ${event.error.message}`;
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}
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setError(errorMessage);
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setIsRecording(false);
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isRecordingRef.current = false;
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stopMediaTracks();
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cleanupAudioAnalysis();
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});
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// Start recording with timeslice for streaming chunks (250ms intervals)
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@@ -494,7 +494,7 @@ Boolean parsing: `value === "true"` or `value === "1"`. Unset or empty values de
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**Capabilities:**
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- 54 built-in voices across 8 languages
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- 53 built-in voices across 8 languages
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- Speed control: 0.25x to 4.0x
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- Output formats: mp3, wav, opus, flac
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- Voice metadata derived from ID prefix (language, gender, accent)
|
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|
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