ElevenLabs Speech-to-Text V2
Audio
ElevenLabs Speech-to-Text V2
POST
ElevenLabs Speech-to-Text V2
Transcribe audio or video files. When use_multi_channel is true and the uploaded audio has multiple channels, a ‘transcripts’ object is returned, with one transcription per channel. Otherwise, a single transcription result is returned.
Request Headers
string
required
Enum value:
application/jsonstring
required
Bearer authentication format: Bearer {{API Key}}.
Request Body
integer
If specified, the system will do its best to sample deterministically. Requests with the same seed and parameters should return the same result, but absolute determinism is not guaranteed. Must be an integer between 0 and 2147483647.Range: [0, 2147483647]
boolean
default:false
Whether to label the current speaker in the uploaded file.
string
default:"other"
Input audio format. Options are ‘pcm_s16le_16’ or ‘other’. pcm_s16le_16 requires the audio to be 16 kHz sample rate, 16-bit integer, mono, little-endian format, and has lower latency than encoded waveforms.Allowed values:
pcm_s16le_16, othernumber
Controls the randomness of the transcription output. The range is 0.0 to 2.0; higher values produce more diverse and less deterministic results. If omitted, the default temperature of the selected model will be used (usually 0).Range: [0, 2]
integer
The maximum number of speakers in the uploaded file. This can be used to help distinguish speakers, with support for up to 32 speakers.Range: [1, 32]
string
Specify the ISO-639-1 or ISO-639-3 language code of the audio file. Providing it in advance can sometimes improve transcription performance. The default is null, which automatically detects the language.
boolean
default:true
Whether to mark audio events such as (laughter) and (footsteps) in the transcription.
string
required
HTTPS link to the file to be transcribed. Exactly one of file and cloud_storage_url must be provided. The file must be accessible via HTTPS and smaller than 2 GB. Any valid HTTPS address is supported, including cloud storage (AWS S3, GCS, Cloudflare R2, etc.), CDNs, or other HTTPS sources. Presigned links with tokens or URL query parameter authentication are supported.
boolean
default:false
Whether the audio file is multi-channel and each channel contains only a single speaker. When enabled, each channel will be transcribed independently and the results will be merged. Each word in the output contains a channel_index field. Up to 5 channels are supported.
number
Diarization threshold. With a higher value, one person is less likely to be split into multiple speakers, but different people are more likely to be merged into one speaker (fewer identified speakers). With a lower value, one person is more likely to be split into multiple speakers, but different people are less likely to be merged into one speaker (more speakers). Can only be set when diarize=True and num_speakers=None. The default is None, which selects a threshold based on the model id (usually 0.22).Range: [0.1, 0.4]
string
default:"word"
The granularity of timestamps in the transcription. ‘word’ provides word-level timestamps, while ‘character’ provides timestamps for each character.Allowed values:
none, word, characterResponse Information
The response may be one of the following response types:
Response Type 1
Response Type 1
string
required
The raw transcribed text.
object[]
required
A list of words and their timing information.
integer
The channel index corresponding to this transcription (valid for multi-channel audio).
string
required
The detected language code (for example, ‘eng’ for English).
string
The unique transcription ID for this response.
number
required
The confidence of language detection (between 0 and 1).