ElevenLabs Speech to Text V1
Audio
ElevenLabs Speech to Text V1
POST
ElevenLabs Speech to Text V1
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 transcript per channel. Otherwise, a single transcription result is returned.
Request Headers
Enum value:
application/jsonBearer authentication format: Bearer {{API key}}.
Request Body
If specified, the system will make a best effort 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.Value range: [0, 2147483647]
Whether to annotate the current speaker in the uploaded file.
The input audio format. Options are ‘pcm_s16le_16’ or ‘other’. pcm_s16le_16 requires the audio to be 16kHz sample rate, 16-bit integer, mono, little-endian format, and has lower latency compared to encoded waveforms.Allowed values:
pcm_s16le_16, otherControls the randomness of the transcription output. The value ranges from 0.0 to 2.0; higher values make results more diverse and less deterministic. If omitted, the selected model’s default temperature is used (usually 0).Value range: [0, 2]
The maximum number of speakers in the uploaded file. This can be used to help distinguish speakers, with support for up to 32 speakers.Value range: [1, 32]
Specify the ISO-639-1 or ISO-639-3 language code of the audio file. Providing this in advance can sometimes improve transcription performance. The default is null, which automatically detects the language.
Whether to tag audio events such as (laughter) and (footsteps) in the transcription.
The HTTPS link to the file to be transcribed. Exactly one of file and cloud_storage_url must be provided. The file must be accessible over HTTPS and smaller than 2GB. 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 authentication via URL query parameters are supported.
Whether the audio file is multi-channel and each channel contains only a single speaker. When enabled, each channel is transcribed independently and the results are combined. Each word in the output content contains a channel_index field. Up to 5 channels are supported.
The diarization threshold. A higher value lowers the probability that one person is split into multiple speakers, but increases the probability that different people are merged into one speaker (fewer speakers detected). A lower value increases the probability that one person is split into multiple speakers, but reduces the probability that different people are merged into one speaker (more speakers detected). Can only be set when diarize=True and num_speakers=None. The default is None, and the threshold is selected based on the model id (usually 0.22).Value range: [0.1, 0.4]
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
The raw transcribed text.
A list of words and their timing information.
The channel index corresponding to this transcript (valid for multi-channel audio).
The detected language code (for example, ‘eng’ for English).
The unique transcription ID for this response.
The confidence of language detection (between 0 and 1).