Realtime Text-to-Speech (TTS) API
Overview
The Realtime TTS API is a WebSocket endpoint for real-time text-to-speech synthesis. It accepts text incrementally and returns synthesized audio as base64-encoded chunks with minimal latency.
Connection
Connect to the endpoint below, passing your API Key as the token query parameter (or in the Authorization header):
wss://stream.palabra.ai/tts-api/v1/text-to-speech/stream?token={API_KEY}(Europe)wss://stream.us.palabra.ai/tts-api/v1/text-to-speech/stream?token={API_KEY}(United States)
TTS Session Lifecycle
- Connect to the WebSocket endpoint.
- Send an
initmessage with voice and output settings. - Send
textmessages (up to 1024 characters each); mark final chunks withis_eos: true. - Receive
audio_chunkmessages with base64-encoded audio. - Send
cancelanytime to stop synthesis.
The session persists until you disconnect. Settings from init apply to the entire session and can be overridden in individual text messages.
Supported Languages
| Code | Language |
|---|---|
en | English (generic) |
en-us | English (US) |
en-gb | English (UK) |
de | German |
es | Spanish |
es-eu | Spanish (European) |
es-la | Spanish (Latin American) |
fr | French |
fr-eu | French (European) |
fr-ca | French (Canadian) |
hi | Hindi |
it | Italian |
nl | Dutch |
pt | Portuguese |
pt-eu | Portuguese (European) |
pt-la | Portuguese (Latin American) |
ru | Russian |
ko | Korean |
cs | Czech |
fi | Finnish |
id | Indonesian |
pl | Polish |
ro | Romanian |
sk | Slovak |
sv | Swedish |
tr | Turkish |
Client → Server Messages
init
Must be sent once, immediately after connecting.
{
"type": "init",
"language": "en",
"model": "auto",
"voice_options": {
"voice_id": "default_low",
"speed": 1.0,
"deaccent_strength": 1.0
},
"output": {
"format": "pcm",
"sample_rate": 24000
}
}
| Field | Type | Required | Default | Description |
|---|---|---|---|---|
language | string | ✓ | — | BCP-47 language code, e.g. en, ru, de, fr. |
model | string | — | auto | TTS model ID. Use auto to select the best model for the language. |
voice_options | object | ✓ | — | Voice configuration. See fields below. |
output | object | — | — | Output audio settings. See fields below. |
voice_options
| Field | Type | Required | Default | Description |
|---|---|---|---|---|
voice_id | string | ✓ | — | Voice identifier. Use "default_low" or "default_high" values for the language's default voice, or the ID of any built-in voice or a voice you've cloned. |
speed | float | — | 1.0 | Speed multiplier. Range: 0.0–2.0. Value 1.0 is normal speed and gives the best quality — increase to speak faster, decrease to speak slower. |
deaccent_strength | float | — | 1.0 | Range: 0.0–1.0. Сloned voices can carry over an accent from the sample's original language and speak language with it — most commonly an English accent, regardless of what language the sample was cloned from. This shifts the voice's embedding toward the native voice cluster of chosen language to reduce that accent. 0.0 disables the effect; 1.0 applies it fully. A value of 1.0 is recommended. Lowering it slightly preserves some of the accent, if needed. |
output
| Field | Type | Required | Default | Description |
|---|---|---|---|---|
format | string | — | mp3 | Output audio format. One of pcm, mp3, wav. |
sample_rate | integer | — | 24000 | Output sample rate in Hz. Range: 8000–48000. |
text
Send a chunk of text to synthesize. Each message must be 1024 characters or fewer. Send multiple messages to stream longer text — mark the last chunk of each sentence with is_eos: true.
Voice options can be overridden per message — only the fields you supply are changed.
{
"type": "text",
"text": "Hello, how can I help you today?",
"is_eos": true
}
{
"type": "text",
"text": "This part is spoken faster.",
"is_eos": false,
"voice_options": {
"speed": 1.2
}
}
| Field | Type | Required | Default | Description |
|---|---|---|---|---|
text | string | ✓ | — | Text to synthesize. Max 1024 characters per message. |
generation_id | string | — | — | Optional client-supplied ID (4–256 chars). If set, it's returned as generation_id on the resulting audio_chunk messages so you can correlate output with this request. If omitted, the server assigns one. |
is_eos | boolean | — | false | End of sentence. When true, finalizes speech synthesis for the provided sentence. |
voice_options | object | — | — | Override of session voice options for this and subsequent messages. |
cancel
Stop the current synthesis immediately. The session stays open — you can send new text messages right after.
{
"type": "cancel"
}
Server → Client Messages
audio_chunk
Sent as audio is generated. Each message contains a base64-encoded audio chunk in the format specified in init.output.format.
{
"message_type": "audio_chunk",
"data": {
"audio": "Y3VyaW91cyBtaW5kcyB0aGluayBhbGlrZQ==",
"size": 9600,
"generation_id": "1a2b3c4d",
"last_chunk": false,
"chunk_generation_delta": 120,
"audio_len": 0.2
}
}
| Field | Type | Description |
|---|---|---|
data.audio | string | Base64-encoded audio data. |
data.size | integer | Size of the decoded audio in bytes. |
data.generation_id | string | ID of the generation this chunk belongs to. Echoes the generation_id you sent in the text message, or a server-assigned value if you didn't. Use it to correlate chunks with a specific request. |
data.last_chunk | boolean | true on the final chunk of a generation — synthesis for that sentence is complete. Lets you detect the end without guessing. |
data.chunk_generation_delta | integer | null | Latency metric: milliseconds elapsed from the start of the generation to the moment this chunk was produced. null when the backend reports no timing for the chunk — in particular on the final end-of-generation marker (see last_chunk). Informational only. |
data.audio_len | number | null | Duration of this chunk's audio in seconds. Present on every chunk that carries audio; null on the final end-of-generation marker, which carries no audio. |
error
Sent on session-level or synthesis errors.
{
"message_type": "error",
"data": {
"code": "SERVICE_UNAVAILABLE",
"desc": "Speech synthesis timed out. Please try again."
}
}
| Code | Retryable | Description |
|---|---|---|
SERVICE_UNAVAILABLE | ✓ | Synthesis service issue. Wait and retry. |
SERVER_ERROR | ✗ | Server-side synthesis failure that is not retryable. |
UNKNOWN_ERROR | ✗ | Unexpected, unhandled internal error. Contact support if it persists. |
BAD_REQUEST | ✗ | Incoming message is not valid JSON or not valid UTF-8. The session stays open. |
VALIDATION_ERROR | ✗ | Invalid message field. See desc for details. |
CONFLICT | ✗ | Session already initialised. Reconnect to change settings. |
SESSION_NOT_FOUND | ✗ | Send init before sending text. |
UNAUTHORIZED | ✗ | API Key or access token missing, invalid, or expired. |
RATE_LIMIT_EXCEEDED | ✓ | Text-message rate limit exceeded (max 50 messages per second). The session stays open — wait briefly and retry. |
- Connections — max 20 new connections per minute per token. Exceeding this does not produce a
RATE_LIMIT_EXCEEDEDmessage: the server closes the WebSocket with code1008(policy violation) and reasonConnection rate limit exceeded. Wait before reconnecting. - Text messages — max 50
textmessages per second. Exceeding this returns anerrorwith codeRATE_LIMIT_EXCEEDEDand keeps the session open.
Output formats
| Format | Description |
|---|---|
pcm | Raw 16-bit signed PCM, little-endian. No container. Best for real-time playback — schedule each chunk immediately as it arrives. |
mp3 | MPEG Layer 3 compressed audio. Collect all chunks and decode together. |
wav | PCM with RIFF/WAVE container. Collect all chunks and decode together. |
data.audio as base64 → Uint8Array → Int16Array (little-endian) → Float32Array, create an AudioBuffer and schedule with AudioBufferSourceNode.start(nextTime) — accumulating nextTime += audioBuffer.duration after each chunk for gapless playback.Example — streaming from an LLM
This example shows how sentences split into chunks can be streamed to the TTS API. Voice speed is increased mid-stream to demonstrate per-message overrides.
import asyncio, json, base64, websockets
WS_TTS_URL = "wss://stream.palabra.ai/tts-api/v1/text-to-speech/stream"
API_KEY = "<your-api-key>" # create one at https://platform.palabra.ai/api-keys
SENTENCES = [
(
"The sun was setting over the mountains,",
"casting long golden shadows across the valley below.",
),
(
"Birds were returning to their nests,",
"filling the air with their evening songs.",
),
(
"A gentle breeze moved through the tall grass,",
"creating waves that rippled toward the horizon.",
),
]
async def main():
url = f"{WS_TTS_URL}?token={API_KEY}"
async with websockets.connect(url) as ws:
# 1. Initialise the session
await ws.send(json.dumps({
"type": "init",
"language": "en",
"model": "auto",
"voice_options": {
"voice_id": "default_low",
"speed": 1.0,
"deaccent_strength": 1.0,
},
"output": {
"format": "pcm",
"sample_rate": 24000,
}
}))
# 2. Send each sentence chunk by chunk, is_eos=True on the last chunk of each sentence
sent = 0
for i, sentence in enumerate(SENTENCES):
for k, chunk in enumerate(sentence):
msg = {
"type": "text",
"text": chunk,
"is_eos": k == len(sentence) - 1,
}
if k == 0: # override speed only at the start of each sentence
msg["voice_options"] = {"speed": min(0.5 + i / 10, 1.0)}
await ws.send(json.dumps(msg))
sent += 1
# 3. Collect audio until every generation has signalled last_chunk
audio = bytearray()
done = 0
async for raw in ws:
data = json.loads(raw)
if data.get("message_type") == "audio_chunk":
chunk = data["data"]
audio.extend(base64.b64decode(chunk["audio"]))
print(f"Chunk: {chunk['size']} bytes, total: {len(audio)}")
if chunk.get("last_chunk"):
done += 1
if done == sent: # all sentences finished
break
elif data.get("message_type") == "error":
print(f"Error: {data['data']['code']} — {data['data']['desc']}")
break
print(f"Total: {len(audio)} bytes")
with open("output.pcm", "wb") as f:
f.write(audio)
# Play: ffplay -f s16le -ar 24000 -ac 1 output.pcm
asyncio.run(main())