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POST
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:s2.1-pro

Specify which TTS model to use. Use s2.1-pro-free for the free developer tier. If omitted or set to an unrecognized value, the request falls back to s2.1-pro.

Available options:
s1,
s2-pro,
s2.1-pro,
s2.1-pro-free,
drama-3

Body

Request body for text-to-speech synthesis. Supports single-speaker synthesis on all compatible TTS models. Multi-speaker dialogue synthesis is available with the S2 family (s2-pro, s2.1-pro, s2.1-pro-free) and drama-3, not s1.

Single Speaker

Provide either reference_id (string) pointing to a voice model, or references (array of ReferenceAudio) for zero-shot cloning.

Multiple Speakers (Dialogue — S2 family and drama-3)

For multi-speaker synthesis, provide:

  • reference_id: array of voice model IDs, e.g., ["speaker-0-id", "speaker-1-id"]
  • text: use speaker tags <|speaker:0|>, <|speaker:1|>, etc. to indicate speaker changes, e.g., "<|speaker:0|>Hello!<|speaker:1|>Hi there!"

Alternatively, for zero-shot multi-speaker:

  • references: 2D array where each inner array contains references for one speaker
  • reference_id: array of identifiers (can be arbitrary strings for zero-shot)

Example (Multi-Speaker with Model IDs)

​
text
string
required

Text to convert to speech.

​
temperature
number
default:0.7

Controls expressiveness. Higher is more varied, lower is more consistent.

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

Controls diversity via nucleus sampling.

Required range: 0 <= x <= 1
​
references

Single speaker: array of reference audio samples

​
reference_id

Single speaker: voice model ID string

​
prosody
ProsodyControl · object | null

Speed and volume adjustments for the output.

​
chunk_length
integer
default:300

Text segment size for processing.

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

Rewrites numbers, dates, units and symbols as spoken words before synthesis, improving stability. Uses language when it is set and otherwise detects the language of each text chunk.

​
language
string | null

BCP-47 language tag of the text, for example en, es-MX or zh-CN. Case-insensitive, and _ is read as -. Used only by text normalization (how numbers, dates, units and symbols are spelled out before synthesis) and by word-timestamp alignment. It is not passed to the speech model and does not change the voice, accent or audio quality; a wrong tag only changes how those tokens are spelled out. Omit it, or send auto or an empty string, to detect the language per chunk. A tag without dedicated normalization rules skips normalization instead of failing. Without a region, Spanish follows Spain conventions; send es-MX for Mexican readings. When normalize is false it only affects word timestamps. For text that mixes languages, leave it unset.

Maximum string length: 35
Pattern: ^[A-Za-z0-9_-]*$
Example:

"en"

​
format
enum<string>
default:mp3

Output audio format.

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

Audio sample rate in Hz. When null, uses the format's default (44100 Hz for most formats, 48000 Hz for opus).

​
mp3_bitrate
enum<integer>
default:128

MP3 bitrate in kbps. Only applies when format is mp3.

Available options:
64,
128,
192
​
opus_bitrate
enum<integer>
default:-1000

Opus bitrate in bps. -1000 for automatic. Only applies when format is opus.

Available options:
-1000,
24000,
32000,
48000,
64000
​
latency
enum<string>
default:normal

Latency-quality trade-off. normal: best quality, balanced: reduced latency, low: lowest latency.

Available options:
low,
normal,
balanced
​
max_new_tokens
integer
default:1024

Maximum audio tokens to generate per text chunk.

​
repetition_penalty
number
default:1.2

Penalty for repeating audio patterns. Values above 1.0 reduce repetition.

​
min_chunk_length
integer
default:50

Minimum characters before splitting into a new chunk.

Required range: 0 <= x <= 100
​
condition_on_previous_chunks
boolean
default:true

Use previous audio as context for voice consistency.

​
early_stop_threshold
number
default:1

Early stopping threshold for batch processing.

Required range: 0 <= x <= 1
​
features
string[]

Optional request-scoped TTS feature flags forwarded verbatim to the inference backend. Use ["quality-guard"] to enable the quality guard for this synthesis request. Feature availability is determined by the inference backend.

​
pronunciation_dictionary
PronunciationDictionaryRef · object

Reference to a server-managed pronunciation dictionary. The server resolves contents via the platform pronunciation-dictionaries API and forwards the full merged dictionary to the inference engine, which performs the substitution.

Response

Request fulfilled, document follows