POST
/
v1
/
tts
This response does not have an example.

This endpoint only accepts application/json and application/msgpack.

For best results, upload reference audio using the create model before using this one. This improves speech quality and reduces latency.

To upload audio clips directly, without pre-uploading, serialize the request body with MessagePack as per the instructions.

Audio formats supported:

  • WAV / PCM
    • Sample Rate: 8kHz, 16kHz, 24kHz, 32kHz, 44.1kHz
    • Default Sample Rate: 44.1kHz
    • 16-bit, mono
  • MP3
    • Sample Rate: 32kHz, 44.1kHz
    • Default Sample Rate: 44.1kHz
    • mono
    • Bitrate: 64kbps, 128kbps (default), 192kbps
  • Opus
    • Sample Rate: 48kHz
    • Default Sample Rate: 48kHz
    • mono
    • Bitrate: -1000 (auto), 24kbps, 32kbps (default), 48kbps, 64kbps

Authorizations

​
Authorization
string
header
required

Bearer authentication header of the form Bearer <token>, where <token> is your auth token.

Headers

​
model
enum<string>
default:speech-1.5

Specify which TTS model to use

Available options:
speech-1.5,
speech-1.6,
agent-x0

Body

​
text
string
required

Text to be converted to speech

​
temperature
number
default:0.7

Controls randomness in the speech generation. Higher values (e.g., 1.0) make the output more random, while lower values (e.g., 0.1) make it more deterministic

Required range: 0 <= x <= 1
​
top_p
number
default:0.7

Controls diversity via nucleus sampling. Lower values (e.g., 0.1) make the output more focused, while higher values (e.g., 1.0) allow more diversity

Required range: 0 <= x <= 1
​
references
object[] | null

References to be used for the speech, this requires MessagePack serialization, this will override reference_voices and reference_texts

​
reference_id
string | null

ID of the reference model o be used for the speech

​
prosody
object | null

Prosody to be used for the speech

​
chunk_length
integer
default:200

Chunk length to be used for the speech

Required range: 100 <= x <= 300
​
normalize
boolean
default:true

Whether to normalize the speech, this will reduce the latency but may reduce performance on numbers and dates

​
format
enum<string>
default:mp3

Format to be used for the speech

Available options:
wav,
pcm,
mp3,
opus
​
sample_rate
integer | null

Sample rate to be used for the speech

​
mp3_bitrate
enum<integer>
default:128

MP3 Bitrate to be used for the speech

Available options:
64,
128,
192
​
opus_bitrate
enum<integer>
default:32

Opus Bitrate to be used for the speech

Available options:
-1000,
24,
32,
48,
64
​
latency
enum<string>
default:normal

Latency to be used for the speech, balanced will reduce the latency but may lead to performance degradation

Available options:
normal,
balanced

Response

200
_mintlify/placeholder

Request fulfilled, document follows