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Blue","email":"support@breezeblue.ai"},"license":"MIT","homepage":"https://breezeblue.ai","keywords":["audio","breeze","breeze-blue","sdk","text-to-speech","tts","voice"],"repository":{"url":"git+https://github.com/breezebluestudio/BreezeBlueStudio.git","type":"git","directory":"packages/breeze-typescript"},"description":"ESM-first TypeScript SDK for the Breeze Blue Developer API.","maintainers":[{"name":"breezeblue","email":"infra@breeze.blue"}],"readme":"# @breeze.blue/sdk\n\nESM-first TypeScript SDK for the Breeze Blue Developer API. Covers\ntext-to-speech, voice management, voice preview generation, history audio,\nmodels, and account usage.\n\nThe root entrypoint is safe to import in Node, browser, and edge runtimes. Local\naudio playback helpers live in the Node-only `@breeze.blue/sdk/node` subpath.\n\n## Install\n\n```sh\npnpm add @breeze.blue/sdk\n```\n\nor:\n\n```sh\nnpm install @breeze.blue/sdk\n```\n\n## API Key\n\nCreate an API key in the Breeze Blue Developer Console, then export it:\n\n```sh\nexport BREEZE_API_KEY=brz_...\n```\n\nThe SDK sends the key with the `xi-api-key` header. It also sends\n`x-breeze-sdk` for server-side observability.\n\nAn API key is required for every REST method. The one exception is\n`textToSpeech.realtime.connect(...)` with a `clientSecret`, which lets a\nbrowser client run without a key (see the realtime section below).\n\n## Quickstart\n\n```ts\nimport { BreezeBlueClient } from \"@breeze.blue/sdk\";\nimport { save } from \"@breeze.blue/sdk/node\";\n\nconst client = new BreezeBlueClient();\n\nconst audio = await client.textToSpeech.convert(\n \"voc_...\",\n { text: \"Hello from Breeze Blue.\" },\n { outputFormat: \"mp3\" },\n);\n\nawait save(audio, \"hello.mp3\");\nconsole.log(audio.contentType);\nconsole.log(audio.historyItemId);\n```\n\n`new BreezeBlueClient()` reads `BREEZE_API_KEY` by default and sends requests to\n`https://api.breeze.blue`. To point at another environment, pass `baseUrl` or\nset `BREEZE_BASE_URL`.\n\n```ts\nconst client = new BreezeBlueClient({\n apiKey: \"brz_...\",\n baseUrl: \"https://api.breeze.blue\",\n timeout: 120_000,\n});\n```\n\nPer-request options accept `timeout`, `signal`, and extra `headers`.\n\n## Text to Speech\n\n```ts\nconst audio = await client.textToSpeech.convert(\"voc_...\", {\n text: \"Render this line.\",\n});\n\nconst audioStream = await client.textToSpeech.stream(\"voc_...\", {\n text: \"Stream this line.\",\n});\n\nconst enhanced = await client.textToSpeech.enhance({\n instruction: \"Calm, warm, bedtime narration.\",\n languageCode: \"en\",\n});\n```\n\nUse async text-to-speech for long text, reference-heavy voices, or batch\nproduction where the caller should not hold an HTTP connection open:\n\n```ts\nconst job = await client.textToSpeech.createJob(\n \"voc_...\",\n { text: \"Render this longer script.\" },\n { outputFormat: \"mp3\" },\n);\n\nconst status = await client.generationJobs.get(job.generationJobId);\nif (status.status === \"ready\") {\n const audio = await client.generationJobs.downloadAudio(job.generationJobId);\n await save(audio, \"async.mp3\");\n}\n```\n\nIf the job is still active, `downloadAudio(...)` rejects with\n`BreezeBlueGenerationNotReadyError`; read `error.retryAfter` before retrying.\n\nUse realtime text-to-speech when one WebSocket connection should handle multiple\nconversation turns. Realtime audio is fixed to raw `pcm_s16le`, 24000 Hz, mono,\n16-bit frames.\n\nRealtime needs a `WebSocket` implementation. Browsers, edge runtimes, and\nNode 22+ provide a global one; on Node 20, pass a constructor (for example the\n`ws` package's `WebSocket`) via `new BreezeBlueClient({ webSocket })`.\n\nStart consuming before appending text. Audio can arrive after `flush()` and may\ncontinue after `endTurn()`; keep the single consumer running until `turn.done`.\n\n```ts\nconst connection = await client.textToSpeech.realtime.connect(\"voc_...\", {\n modelId: \"breeze-tts-2\",\n});\n\nconst pcmChunks: Uint8Array[] = [];\nconst consumer = (async () => {\n for await (const message of connection) {\n if (message.type === \"audio\") {\n // Forward this chunk to your playback or transport layer immediately.\n pcmChunks.push(message.audio);\n } else if (message.type === \"error\") {\n throw new Error(`Realtime TTS failed: ${message.code}: ${message.message}`);\n } else if (message.type === \"turn.done\") {\n return;\n }\n }\n throw new Error(\"Realtime session closed before turn.done\");\n})();\n\nconnection.startTurn(\"turn_1\");\nconnection.appendText(\"Hello from Breeze.\");\nconnection.flush();\nconnection.endTurn();\n\nawait consumer;\nconnection.close();\n```\n\nAlways consume the connection (or its `audio()` / `events()` iterators) while a\nturn is active, and call `close()` when you are done. When the SDK detects a\nconnection failure — an invalid server frame, or more than 16 MiB of messages\npiling up unconsumed — it closes the connection and the iterator rejects with\na `BreezeBlueRealtimeError`. The `audio()` helper also throws\n`BreezeBlueRealtimeError` for server `error` events and abnormal WebSocket\ncloses instead of silently ending.\n\nWhen `BreezeBlueRealtimeError.recoverable` is `true`, the same connection\nremains usable. Most recoverable turn-command errors cancel the active turn.\nAn active-turn `session.update` rejection is the exception: the turn keeps\nrunning. If `connection.audio()` throws that error, start a fresh `audio()`\niterator immediately on the same connection and continue collecting the\ncurrent turn; do not start a replacement turn.\n\nEvents are a typed discriminated union (`session.ready`, `session.updated`, `turn.started`,\n`audio.started`, `turn.done`, `turn.cancelled`, `usage.committed`,\n`session.expiring`, `session.closed`, `error`, `pong`), with camelCase fields\nsuch as `turnId`, `historyItemId`, `expiresAt`, and `ttfaMs`. Treat the union as\nnon-exhaustive: the server may add event types, and the SDK delivers unknown\nJSON events unchanged — ignore event types you do not recognize instead of\nswitching exhaustively.\n\nBetween turns, update the synthesis instructions without replacing the\nWebSocket. `updateInstructions(...)` only sends the request; the matching\n`session.updated` acknowledgement remains on the same ordered iterator. Wait\nfor it before starting the next turn:\n\n```ts\nconnection.updateInstructions(\"Speak faster and with more energy.\");\nfor await (const message of connection) {\n if (message.type === \"session.updated\") {\n if (message.instructions !== \"Speak faster and with more energy.\") {\n throw new Error(\"Unexpected instructions acknowledgement\");\n }\n break;\n }\n}\n\nconnection.startTurn(\"turn_2\");\n```\n\nInstructions must be a non-empty string of at most 1,000 characters. Only one\nupdate may await acknowledgement, and updates are accepted only while no turn\nis active. An update attempted during a turn is rejected without interrupting\nthat turn. Keep one stream consumer: if a long-lived consumer owns the\niterator, have it signal your turn producer when `session.updated` arrives\ninstead of starting a second iterator.\n\nSend a keepalive ping inside the session's `inactivityTimeoutSeconds` window;\nthe server answers with a `pong` event. Keep it running during active turns too:\na long TTFA or upstream stall with no audio does not pause the idle deadline.\n\n```ts\nconst keepalive = setInterval(() => connection.ping(), 10_000);\n// ... run turns ...\nclearInterval(keepalive);\nconnection.close();\n```\n\nFor a long logical conversation, use the opt-in managed connection. It derives\na safe keepalive interval from `session.ready` and requires a server response\nafter each ping. A missing acknowledgement replaces an idle half-open socket;\nif a turn is active, it reports `TURN_INTERRUPTED` and never replays commands.\nThe manager also marks a physical WebSocket for turn-boundary rotation after\n10 minutes by default (or earlier when the server deadline requires it). An\nactive turn can delay the switch beyond that threshold; the server-advertised\nhard lifetime still applies. The manager\nperforms up to three unexpected idle reconnect attempts per interruption with\nfresh sessions and jittered exponential backoff:\n\n```ts\nconst connection = await client.textToSpeech.realtime.connectManaged(\"voc_...\", {\n modelId: \"breeze-tts-2\",\n});\n\nconst consumer = (async () => {\n for await (const message of connection) {\n if (message.type === \"audio\") {\n // Play or forward this PCM chunk immediately.\n } else if (message.type === \"session.ready\") {\n // A new physical epoch is ready; the logical connection remains the same.\n } else if (message.type === \"turn.done\") {\n return;\n }\n }\n})();\n\nawait connection.startTurnWhenReady(\"turn_1\");\nconnection.appendText(\"Hello from a long-running conversation.\");\nconnection.endTurn();\nawait consumer;\nconnection.close();\n```\n\nUse `heartbeatTimeoutMs` to tune the acknowledgement deadline and\n`maxPhysicalSessionMs` to tune the client-side epoch cap. Set\n`maxPhysicalSessionMs: 0` only when you intentionally want to rely solely on\nthe server-advertised deadline.\n\nWhen an idle reconnect is handled successfully, the logical iterator stays\nopen, suppresses the replaced physical socket's terminal event, and emits the\nnew epoch's `session.ready`. The replacement sends its first heartbeat\nimmediately and resumes the derived cadence after an inbound acknowledgement.\nReconnect attempts are bounded per interruption. `startTurnWhenReady(...)`\nwaits only when one of these idle replacements is already in progress and\n`turn.start` has not been sent; await it before sending any other turn command.\nAfter a planned rotation, an old epoch with outstanding `usage.committed`\nevents drains in parallel for up to five seconds and closes as soon as all\nknown completed turns settle. This event is best-effort; use history and usage\nAPIs as the durable source of truth.\n\nThe managed connection never buffers turn content or replays a command. The\nbounded `startTurnWhenReady(...)` wait happens before its first WebSocket\nwrite. If a WebSocket is interrupted while a turn is active, its iterator rejects with\n`BreezeBlueRealtimeError` and `error.code === \"TURN_INTERRUPTED\"`; decide from\nyour application conversation state whether and how to start a new turn.\nTreat the manager as an active realtime call, not a presence channel. Always\ncall `connection.close()` when the call ends, the user leaves, or the page\nenters a long-lived background state; otherwise its heartbeat intentionally\nkeeps a server WebSocket slot occupied.\n\nTo cut time to first audio, create the session ahead of time (for example while\nyour app is still preparing the turn) and connect with its `clientSecret` when\nthe first text is ready — only the WebSocket handshake remains:\n\n```ts\nconst session = await client.textToSpeech.realtime.createSession(\"voc_...\", {\n modelId: \"breeze-tts-2\",\n});\n\n// Later, when the first text is ready:\nconst connection = await client.textToSpeech.realtime.connect(\"voc_...\", {\n clientSecret: session.clientSecret,\n websocketUrl: session.websocketUrl,\n directWebsocketUrl: session.directWebsocketUrl,\n});\n```\n\nWhen the session response includes `directWebsocketUrl`, the SDK prefers its\nquery-free origin and authenticates with the WebSocket subprotocol. It falls\nback to `websocketUrl` for services that have not enabled the direct transport.\n\nThe same `clientSecret` handoff lets a browser connect without ever seeing your\nAPI key: create the session on your server, hand `session.clientSecret` to the\npage, and build a key-less client there:\n\n```ts\n// Browser — no API key required for clientSecret connections.\nconst browserClient = new BreezeBlueClient();\nconst connection = await browserClient.textToSpeech.realtime.connect(\"voc_...\", {\n clientSecret, // received from your server\n});\n```\n\nFor a managed browser connection, provide a `sessionFactory`. The SDK invokes it\nfor every physical epoch, so the callback must fetch a newly minted secret from\nyour backend rather than cache the first response. Session configuration belongs\nin that backend request when a custom factory is used:\n\n```ts\nimport { BreezeBlueRealtimeError } from \"@breeze.blue/sdk\";\n\nconst connection = await browserClient.textToSpeech.realtime.connectManaged(\"voc_...\", {\n sessionFactory: async ({ signal }) => {\n const response = await fetch(\"/api/breeze-realtime-session\", {\n method: \"POST\",\n signal,\n });\n if (!response.ok) {\n const reconnect =\n [408, 425, 429].includes(response.status) || response.status >= 500;\n throw new BreezeBlueRealtimeError(\"Could not create realtime session\", {\n code: \"SESSION_FACTORY_ERROR\",\n reconnect,\n });\n }\n return response.json(); // { clientSecret, websocketUrl?, directWebsocketUrl? }\n },\n});\n```\n\nHonor the factory `signal`: the SDK aborts it when the logical connection closes\nor the physical-epoch timeout expires. A custom callback that ignores the\nsignal must still enforce its own bounded request timeout.\n\nAfter a dynamic instruction update receives `session.updated`, the managed SDK\ncarries the confirmed value to later physical epochs. SDK-created replacements\ninclude it in their session request; factory-created replacements receive an\nautomatic `session.update` before their next `session.ready` is exposed to the\nlogical iterator.\n\n`modelId`, `languageCode`, `voiceSettings`, `inactivityTimeoutSeconds`, and\n`enableLogging` are fixed when the session is created. The initial\n`instructions` value is also supplied at creation, but it can later be\nreplaced between turns with `updateInstructions(...)`. `connect` ignores\ninitial session options when `clientSecret` or `websocketUrl` is provided and\nlogs a warning.\n\nIf a realtime WebSocket is interrupted by a network change, service deployment,\nor upstream realtime worker restart, `connectManaged(...)` handles bounded\nreconnects only while there is no active turn. With the lower-level\n`connect(...)`, handle `error.meta.reconnect === true` or a `session.closed`\nevent with `reconnect === true` by creating a new session and starting a new\nturn from your own conversation state. Active turns are never resumed in place.\nDuring a managed reconnect, transient session-creation responses (408, 425, 429,\nand 5xx) share the same bounded reconnect budget; other 4xx responses fail the\nlogical connection immediately. A custom `sessionFactory` should preserve that\ndistinction with `new BreezeBlueRealtimeError(message, { reconnect })`; a plain\n`Error` cannot communicate a terminal policy response and is treated as\ntransient inside the same bounded budget. `GENERATION_CAPACITY_EXCEEDED` is\nturn-scoped: wait for the following `turn.cancelled`, back off using\n`meta.retryAfterSeconds`, and start a new turn on the same managed logical\nconnection without replaying text.\n\nThe API uses the default text-to-speech model when `modelId` is omitted. If\nyou need to select a model explicitly, call `client.models.list()` and pass one\nof the returned `modelId` values.\n\nAll field names follow TypeScript conventions (`modelId`, `voiceSettings`,\n`outputFormat`, `historyItemId`, ...). The SDK translates them to the\nsnake_case HTTP wire format on send and translates JSON responses back to\ncamelCase on receive.\n\nAudio responses expose convenient helpers:\n\n```ts\nawait audio.arrayBuffer();\nawait audio.bytes();\nawait audio.blob();\n\naudio.contentType;\naudio.historyItemId;\n```\n\nNode playback helpers:\n\n```ts\nimport { play, save, stream } from \"@breeze.blue/sdk/node\";\n\nawait play(audio); // ffplay, with macOS afplay fallback\nawait save(audio, \"x.mp3\");\nawait stream(audioStream); // mpv\n```\n\nStreaming text-to-speech defaults to `pcm` to reduce time to first audio. Pass\n`{ outputFormat: \"wav\" }` or `{ outputFormat: \"mp3\" }` when you need that wire\nformat explicitly.\n\nSync and async text-to-speech also take a sample rate and, for the lossy encodings, a bitrate, such as\n`{ outputFormat: \"wav_48000\" }`. Breeze resamples its native 24000 Hz output for\ndownstream compatibility; profiles carrying a sample rate are not available for streaming.\n\n## Voices\n\nSearch every voice available to the account and inspect a single voice. Public\ncatalog results combine semantic similarity with exact, prefix, and substring\nname matching; saved personal voices remain searchable by name, description,\nor voice ID.\n\n```ts\nconst voices = await client.voices.search({ search: \"calm documentary narrator\" });\nconst newestFirst = await client.voices.search();\nconst dailyTrend = await client.voices.search({\n voiceType: \"default\",\n sort: \"trend\",\n languageCode: \"zh\",\n gender: [\"female\", \"neutral\"],\n tone: [\"warm\"],\n});\nconst taggedFavorites = await client.voices.search({\n favoritesOnly: true,\n tags: [\"narration\", \"novel=三体\"],\n});\nconst firstVoiceId = voices.voices[0].voiceId;\n\nconst voice = await client.voices.get(firstVoiceId);\nconst settings = await client.voices.getSettings(firstVoiceId);\n\nconst randomVoice = await client.voices.random();\nconsole.log(randomVoice.voiceId, randomVoice.name);\n// Narrow the random pool by language and/or voice source.\nconst randomCatalogVoice = await client.voices.random({ languageCode: \"zh\", voiceType: \"default\" });\n\n// Discover the current code-only Voice Metadata contract before building a form.\nconst metadataOptions = await client.voices.metadataOptions();\nconsole.log(metadataOptions.languageCodes);\nconsole.log(metadataOptions.accentCodesByLanguage.en);\n```\n\nVoice listings keep the published default of stable creation-time order from\nnewest to oldest. Use `{ sort: \"trend\", voiceType: \"default\" }` for the shared\nDaily Trend order. Eligible voices not yet ranked follow the snapshot members;\nreturned page tokens pin the initial supplemental candidates. Search queries remain relevance-ranked; Trend is only a\nweak prior within the same relevance tier.\n\nBreeze voice creation is always two steps: produce a **preview**, let the\nuser accept it, then **save the preview** as a real voice. Three ways to\nproduce a preview:\n\n```ts\nimport { readFile } from \"node:fs/promises\";\n\n// Option A — clone preview from an audio sample\nconst clonePreview = await client.voices.createClonePreview({\n name: \"Demo voice\",\n file: {\n data: await readFile(\"sample.wav\"),\n filename: \"sample.wav\",\n contentType: \"audio/wav\",\n },\n});\nlet generatedVoiceId = clonePreview.generatedVoiceId;\n\n// Option B — design preview from a text description (no audio)\nconst design = await client.voices.createDesignPreview({\n voiceDescription: \"Warm documentary narrator with clear articulation.\",\n});\ngeneratedVoiceId = design.previews[0].generatedVoiceId;\n```\n\nOption C keeps an existing voice and makes it speak another language.\nLocalization runs as a background job: start it, then poll until the status is\n`ready`. `name` defaults to the source voice name.\n\n```ts\nconst job = await client.voices.createLocalizePreview({\n voiceId: firstVoiceId,\n languageCode: \"es\",\n name: \"Documentary narrator (Spanish)\",\n});\n\nlet localized = await client.voices.getLocalizePreview(job.generationJobId);\nwhile (localized.status !== \"ready\") {\n if (localized.status === \"failed\" || localized.status === \"cancelled\") {\n throw new Error(localized.error?.detail ?? localized.status);\n }\n await new Promise((resolve) => setTimeout(resolve, 2000));\n localized = await client.voices.getLocalizePreview(job.generationJobId);\n}\n\ngeneratedVoiceId = localized.generatedVoiceId!;\n```\n\nVoice Remix creates a same-language candidate from an owned or official voice\nand a custom description or preset. Each successful candidate uses the Voice Design candidate\nprice. The source remains unchanged.\n\n```ts\nconst remixJob = await client.voices.createRemixPreview({\n voiceId: firstVoiceId,\n prompt: \"Make the voice warmer and more expressive.\",\n guidanceScale: 4,\n});\nlet remix = await client.voices.getRemixPreview(remixJob.generationJobId);\nwhile (remix.status !== \"ready\") {\n if (remix.status === \"failed\" || remix.status === \"cancelled\") {\n throw new Error(remix.error?.detail ?? remix.status);\n }\n await new Promise((resolve) => setTimeout(resolve, 2000));\n remix = await client.voices.getRemixPreview(remixJob.generationJobId);\n}\ngeneratedVoiceId = remix.generatedVoiceId!;\n```\n\nUse `streamPreview` and `savePreview` with the generated ID to audition and save\na private, independent voice. A client timeout leaves the server job running.\n\nCustom input is enhanced into an instruction and matching script in the source\nlanguage. List templates with `client.voices.listRemixPresets()` and pass\n`presetId` instead of `prompt` to use one. Edited templates are enhanced;\nunchanged templates are translated only when the source language differs.\n\nTo compare strengths, call `client.voices.prepareRemixPreview(...)` first and\npass its `preparationToken` to each create call with the same input. The token\nlasts 24 hours. Values 2 / 4 / 6 / 8 mean Stable / Expressive / Creative / Intense;\n4 is the default. Web Auto compares separate candidates at 4, 6, and 8.\nUse `options.headers[\"Idempotency-Key\"]` to retry uncertain submissions safely,\nwith a distinct key per candidate. Status includes `comparisonAudioUrl`,\n`comparisonText`, and `sourceVoice`. Saving without `voiceName` or\n`voiceDescription` uses the source values.\nSee the [Voice Remix guide](https://docs.breezeblue.ai/guides/voice-remix).\n\nFor the fastest first audio, generate one design preview as a live 24 kHz mono\nPCM response. The Node `stream()` helper consumes the response body directly;\na clean end means `generatedVoiceId` is ready to save:\n\n```ts\nimport { stream } from \"@breeze.blue/sdk/node\";\n\nconst livePreview = await client.voices.streamDesignPreview({\n voiceDescription: \"Warm documentary narrator with clear articulation.\",\n text: \"This is a short preview script.\",\n});\nawait stream(livePreview);\ngeneratedVoiceId = livePreview.generatedVoiceId!;\n```\n\n`files` is also accepted with exactly one item.\n\nDownload a completed preview so the user can audition it, then save the one they\npick:\n\n```ts\nconst audio = await client.voices.streamPreview(generatedVoiceId);\nawait save(audio, \"preview.mp3\");\n\nconst saved = await client.voices.savePreview({\n generatedVoiceId,\n voiceName: \"Documentary narrator\",\n languageCode: \"en\",\n gender: \"neutral\",\n age: \"middle_aged\",\n tone: [\"warm\", \"articulate\"],\n accent: \"american\",\n tags: [\"narration\", \"novel=三体\"],\n});\n```\n\nEdit, tune settings, or delete a saved voice:\n\n```ts\nawait client.voices.edit(saved.voiceId, {\n name: \"Renamed narrator\",\n languageCode: \"en\",\n tone: [\"calm\", \"measured\"],\n});\n// null clears nullable metadata; [] clears tone. Omitted fields remain unchanged.\nawait client.voices.edit(saved.voiceId, { accent: null, tone: [] });\nawait client.voices.editSettings(saved.voiceId, { guidanceScale: 1.2 });\nawait client.voices.favorite(firstVoiceId);\nawait client.voices.updateTags(firstVoiceId, [\"narration\", \"novel=三体\"]);\nawait client.voices.updateTags(firstVoiceId, []); // Resume owner tag inheritance.\nawait client.voices.unfavorite(firstVoiceId);\nawait client.voices.delete(saved.voiceId);\n```\n\nVoice Metadata uses stable code values. `gender`, `age`, and `accent` may be\n`null`; `tone` contains up to three distinct codes. English and Chinese use\ndifferent accent code sets, and other languages require `accent: null`.\nCall `client.voices.metadataOptions()` to discover every accepted code and the\nlanguage-to-accent cascade at runtime.\n\n## History, Models, and Account\n\n```ts\nconst models = await client.models.list();\nconst balance = await client.account.balance();\nconst currentKey = (await client.account.currentApiKey()).apiKey;\nconsole.log(currentKey.status, currentKey.creditRemaining);\nconst usage = await client.account.usage({ days: 7 });\nconst keyUsage = await client.account.usage({\n apiKeyId: \"key_01hprod\",\n clientType: \"sdk\",\n});\n\nconst history = await client.history.list({ pageSize: 10 });\nconst item = await client.history.get(history.history[0].historyItemId);\nconst audio = await client.history.downloadAudio(item.historyItemId);\n```\n\n## Errors\n\nAPI failures throw `BreezeBlueAPIError` subclasses:\n\n- `BreezeBlueAuthenticationError`\n- `BreezeBlueBadRequestError`\n- `BreezeBlueConflictError`\n- `BreezeBlueForbiddenError`\n- `BreezeBlueNotFoundError`\n- `BreezeBlueValidationError`\n- `BreezeBlueRateLimitError`\n- `BreezeBlueInsufficientCreditsError`\n- `BreezeBlueUpstreamError` for upstream generation/storage/model failures,\n commonly 502/504\n- `BreezeBlueServiceUnavailableError` for capacity or service-configuration\n failures, commonly 503\n\n```ts\nimport { BreezeBlueClient, BreezeBlueRateLimitError } from \"@breeze.blue/sdk\";\n\ntry {\n await new BreezeBlueClient().models.list();\n} catch (error) {\n if (error instanceof BreezeBlueRateLimitError) {\n console.log(error.headers.get(\"retry-after\"));\n }\n}\n```\n\nEach API error exposes `status`, `code`, `detail`, `meta`, and `headers`.\n\nClone previews automatically generate a short script in the detected reference audio language. They accept a reference sample, name, and optional description. Save the preview as a voice, then use Text to Speech for custom scripts, instructions, or target languages.\n\n### HTTP TTS speed\n\nPass `voiceSettings: { speed: 1.25 }` to control speech speed without changing pitch, from 0.5 to 2.0. Omission always means 1.0; saved voice speed is not inherited. Sync, async and HTTP streaming support this parameter; Realtime does not.\n\nPass `voiceSettings: { volume: 1.5 }` to apply a request-level linear amplitude multiplier. Values range from 0.01 to 2.0; omission means 1.0. Values above 1.0 may clip peaks. Sync, async and HTTP streaming support this parameter; it does not change saved voice settings or apply to Realtime.\n\n## Word timing\n\nStream audio and word/token timestamps with `client.textToSpeech.streamWithTimestamps(...)`. Use `for await` to consume chunks containing `audioBase64` and `wordTimestamps`; breaking iteration closes the stream. Decode the audio separately and merge repeated word indices. See [Speech timing](https://docs.breezeblue.ai/guides/speech-timing) for complete examples and timing semantics.\n\nFor a complete response, use `client.textToSpeech.convertWithTimestamps(...)`. It returns base64 audio, its content type, and the full word/token timestamp list. Decode the audio before saving or playback. See [Speech timing](https://docs.breezeblue.ai/guides/speech-timing) and [Convert with timestamps](https://docs.breezeblue.ai/api-reference/text-to-speech/convert-with-timestamps).\n\nFor background generation with timing, use `client.textToSpeech.createJobWithTimestamps(...)`. Poll with `client.generationJobs.get(jobId)`; a ready job includes `wordTimestamps` and the audio download URL.\n\nClone preview creation accepts an omitted name. The server uses the filename stem or `Cloned Voice`; saving a Clone preview without a name keeps its preview name. Other preview types still require a save name. Explicit names longer than 80 Unicode characters are rejected.\n\n## Studio management\n\n`client.studio` exposes `projects`, `chapters`, `clips`, `tracks`, `instances`, and `files`. Import scripts with existing voice IDs, edit inherited instructions and silence, upload MP3/WAV materials, and arrange background audio. Project writes require the latest edit token and a UUID operation ID. Conflicts preserve error metadata and are not retried automatically.\n\nSee the [Studio initialization guide](https://docs.breezeblue.ai/guides/studio-projects) for a complete workflow. Signed uploads use a separate request without the API key. Management operations do not generate speech or export projects.\n\n`client.studio.files.upload(blob, { operationId, name, projectId })` creates the file record, uploads to the signed URL, completes the upload and waits for `ready`; failures reject with `BreezeBlueStudioFileError`. Timeline positions, trims and lengths are 48 kHz frames; use `secondsToFrames` and `framesToSeconds` to convert.\n","readmeFilename":"README.md"}