ElevenLabs Provider
The ElevenLabs provider contains language model support for the ElevenLabs transcription and speech generation APIs.
Setup
The ElevenLabs provider is available in the @ai-sdk/elevenlabs
module. You can install it with
pnpm add @ai-sdk/elevenlabs
Provider Instance
You can import the default provider instance elevenlabs
from @ai-sdk/elevenlabs
:
import { elevenlabs } from '@ai-sdk/elevenlabs';
If you need a customized setup, you can import createElevenLabs
from @ai-sdk/elevenlabs
and create a provider instance with your settings:
import { createElevenLabs } from '@ai-sdk/elevenlabs';
const elevenlabs = createElevenLabs({ // custom settings, e.g. fetch: customFetch,});
You can use the following optional settings to customize the ElevenLabs provider instance:
-
apiKey string
API key that is being sent using the
Authorization
header. It defaults to theELEVENLABS_API_KEY
environment variable. -
headers Record<string,string>
Custom headers to include in the requests.
-
fetch (input: RequestInfo, init?: RequestInit) => Promise<Response>
Custom fetch implementation. Defaults to the global
fetch
function. You can use it as a middleware to intercept requests, or to provide a custom fetch implementation for e.g. testing.
Speech Models
You can create models that call the ElevenLabs speech API
using the .speech()
factory method.
The first argument is the model id e.g. eleven_multilingual_v2
.
const model = elevenlabs.speech('eleven_multilingual_v2');
You can also pass additional provider-specific options using the providerOptions
argument. For example, supplying a voice to use for the generated audio.
import { experimental_generateSpeech as generateSpeech } from 'ai';import { elevenlabs } from '@ai-sdk/elevenlabs';
const result = await generateSpeech({ model: elevenlabs.speech('eleven_multilingual_v2'), text: 'Hello, world!', providerOptions: { elevenlabs: {} },});
-
language_code string or null
Optional. Language code (ISO 639-1) used to enforce a language for the model. Currently, only Turbo v2.5 and Flash v2.5 support language enforcement. For other models, providing a language code will result in an error. -
voice_settings object or null
Optional. Voice settings that override stored settings for the given voice. These are applied only to the current request.- stability double or null
Optional. Determines how stable the voice is and the randomness between each generation. Lower values introduce broader emotional range; higher values result in a more monotonous voice. - use_speaker_boost boolean or null
Optional. Boosts similarity to the original speaker. Increases computational load and latency. - similarity_boost double or null
Optional. Controls how closely the AI should adhere to the original voice. - style double or null
Optional. Amplifies the style of the original speaker. May increase latency if set above 0.
- stability double or null
-
pronunciation_dictionary_locators array of objects or null
Optional. A list of pronunciation dictionary locators to apply to the text, in order. Up to 3 locators per request.
Each locator object:- pronunciation_dictionary_id string (required)
The ID of the pronunciation dictionary. - version_id string or null (optional)
The version ID of the dictionary. If not provided, the latest version is used.
- pronunciation_dictionary_id string (required)
-
seed integer or null
Optional. If specified, the system will attempt to sample deterministically. Must be between 0 and 4294967295. Determinism is not guaranteed. -
previous_text string or null
Optional. The text that came before the current request's text. Can improve continuity when concatenating generations or influence current generation continuity. -
next_text string or null
Optional. The text that comes after the current request's text. Can improve continuity when concatenating generations or influence current generation continuity. -
previous_request_ids array of strings or null
Optional. List of request IDs for samples generated before this one. Improves continuity when splitting large tasks. Max 3 IDs. If bothprevious_text
andprevious_request_ids
are sent,previous_text
is ignored. -
next_request_ids array of strings or null
Optional. List of request IDs for samples generated after this one. Useful for maintaining continuity when regenerating a sample. Max 3 IDs. If bothnext_text
andnext_request_ids
are sent,next_text
is ignored. -
apply_text_normalization enum
Optional. Controls text normalization.
Allowed values:'auto'
(default),'on'
,'off'
.'auto'
: System decides whether to apply normalization (e.g., spelling out numbers).'on'
: Always apply normalization.'off'
: Never apply normalization.
Foreleven_turbo_v2_5
andeleven_flash_v2_5
, can only be enabled with Enterprise plans.
-
apply_language_text_normalization boolean
Optional. Defaults tofalse
. Controls language text normalization, which helps with proper pronunciation in some supported languages (currently only Japanese). May significantly increase latency.
Model Capabilities
Model | Instructions |
---|---|
eleven_v3 | |
eleven_multilingual_v2 | |
eleven_flash_v2_5 | |
eleven_flash_v2 | |
eleven_turbo_v2_5 | |
eleven_turbo_v2 | |
eleven_monolingual_v1 | |
eleven_multilingual_v1 |
Transcription Models
You can create models that call the ElevenLabs transcription API
using the .transcription()
factory method.
The first argument is the model id e.g. scribe_v1
.
const model = elevenlabs.transcription('scribe_v1');
You can also pass additional provider-specific options using the providerOptions
argument. For example, supplying the input language in ISO-639-1 (e.g. en
) format can sometimes improve transcription performance if known beforehand.
import { experimental_transcribe as transcribe } from 'ai';import { elevenlabs } from '@ai-sdk/elevenlabs';
const result = await transcribe({ model: elevenlabs.transcription('scribe_v1'), audio: new Uint8Array([1, 2, 3, 4]), providerOptions: { elevenlabs: { languageCode: 'en' } },});
The following provider options are available:
-
languageCode string
An ISO-639-1 or ISO-639-3 language code corresponding to the language of the audio file. Can sometimes improve transcription performance if known beforehand. Defaults to
null
, in which case the language is predicted automatically. -
tagAudioEvents boolean
Whether to tag audio events like (laughter), (footsteps), etc. in the transcription. Defaults to
true
. -
numSpeakers integer
The maximum amount of speakers talking in the uploaded file. Can help with predicting who speaks when. The maximum amount of speakers that can be predicted is 32. Defaults to
null
, in which case the amount of speakers is set to the maximum value the model supports. -
timestampsGranularity enum
The granularity of the timestamps in the transcription. Defaults to
'word'
. Allowed values:'none'
,'word'
,'character'
. -
diarize boolean
Whether to annotate which speaker is currently talking in the uploaded file. Defaults to
true
. -
fileFormat enum
The format of input audio. Defaults to
'other'
. Allowed values:'pcm_s16le_16'
,'other'
. For'pcm_s16le_16'
, the input audio must be 16-bit PCM at a 16kHz sample rate, single channel (mono), and little-endian byte order. Latency will be lower than with passing an encoded waveform.
Model Capabilities
Model | Transcription | Duration | Segments | Language |
---|---|---|---|---|
scribe_v1 | ||||
scribe_v1_experimental |