text-to-speech
ElevenLabs Text-to-Speech
Generate natural speech from text - supports 74+ languages, multiple models for quality vs latency tradeoffs.
Setup: See Installation Guide. For JavaScript, use
@elevenlabs/*packages only.
Quick Start
Python
from elevenlabs.client import ElevenLabs
client = ElevenLabs()
audio = client.text_to_speech.convert(
text="Hello, welcome to ElevenLabs!",
voice_id="JBFqnCBsd6RMkjVDRZzb", # George
model_id="eleven_multilingual_v2"
)
with open("output.mp3", "wb") as f:
for chunk in audio:
f.write(chunk)
JavaScript
import { ElevenLabsClient } from "@elevenlabs/elevenlabs-js";
import { createWriteStream } from "fs";
const client = new ElevenLabsClient();
const audio = await client.textToSpeech.convert("JBFqnCBsd6RMkjVDRZzb", {
text: "Hello, welcome to ElevenLabs!",
modelId: "eleven_multilingual_v2",
});
audio.pipe(createWriteStream("output.mp3"));
cURL
curl -X POST "https://api.elevenlabs.io/v1/text-to-speech/JBFqnCBsd6RMkjVDRZzb" \
-H "xi-api-key: $ELEVENLABS_API_KEY" -H "Content-Type: application/json" \
-d '{"text": "Hello!", "model_id": "eleven_multilingual_v2"}' --output output.mp3
Models
| Model ID | Languages | Latency | Best For |
|---|---|---|---|
eleven_v3 |
74 | Standard | Highest quality, emotional range |
eleven_multilingual_v2 |
29 | Standard | High quality, most use cases |
eleven_flash_v2_5 |
32 | ~75ms | Ultra-low latency, real-time |
eleven_flash_v2 |
English | ~75ms | English-only, fastest |
eleven_turbo_v2_5 |
32 | Low | Balanced quality/speed |
Voice IDs
Use pre-made voices or create custom voices in the dashboard.
Popular voices:
JBFqnCBsd6RMkjVDRZzb- George (male, narrative)EXAVITQu4vr4xnSDxMaL- Sarah (female, soft)onwK4e9ZLuTAKqWW03F9- Daniel (male, authoritative)XB0fDUnXU5powFXDhCwa- Charlotte (female, conversational)
voices = client.voices.get_all()
for voice in voices.voices:
print(f"{voice.voice_id}: {voice.name}")
Voice Settings
Fine-tune how the voice sounds:
- Stability: How consistent the voice stays. Lower values = more emotional range and variation, but can sound unstable. Higher = steady, predictable delivery.
- Similarity boost: How closely to match the original voice sample. Higher values sound more like the original but may amplify audio artifacts.
- Style: Exaggerates the voice's unique style characteristics (only works with v2+ models).
- Speaker boost: Post-processing that enhances clarity and voice similarity.
from elevenlabs import VoiceSettings
audio = client.text_to_speech.convert(
text="Customize my voice settings.",
voice_id="JBFqnCBsd6RMkjVDRZzb",
voice_settings=VoiceSettings(
stability=0.5,
similarity_boost=0.75,
style=0.5,
use_speaker_boost=True
)
)
Language Enforcement
Force specific language for pronunciation:
audio = client.text_to_speech.convert(
text="Bonjour, comment allez-vous?",
voice_id="JBFqnCBsd6RMkjVDRZzb",
model_id="eleven_multilingual_v2",
language_code="fr" # ISO 639-1 code
)
Text Normalization
Controls how numbers, dates, and abbreviations are converted to spoken words. For example, "01/15/2026" becomes "January fifteenth, twenty twenty-six":
"auto"(default): Model decides based on context"on": Always normalize (use when you want natural speech)"off": Speak literally (use when you want "zero one slash one five...")
audio = client.text_to_speech.convert(
text="Call 1-800-555-0123 on 01/15/2026",
voice_id="JBFqnCBsd6RMkjVDRZzb",
apply_text_normalization="on"
)
Request Stitching
When generating long audio in multiple requests, the audio can have pops, unnatural pauses, or tone shifts at the boundaries. Request stitching solves this by letting each request know what comes before/after it:
# First request
audio1 = client.text_to_speech.convert(
text="This is the first part.",
voice_id="JBFqnCBsd6RMkjVDRZzb",
next_text="And this continues the story."
)
# Second request using previous context
audio2 = client.text_to_speech.convert(
text="And this continues the story.",
voice_id="JBFqnCBsd6RMkjVDRZzb",
previous_text="This is the first part."
)
Output Formats
| Format | Description |
|---|---|
mp3_44100_128 |
MP3 44.1kHz 128kbps (default) - compressed, good for web/apps |
mp3_44100_192 |
MP3 44.1kHz 192kbps (Creator+) - higher quality compressed |
pcm_16000 |
Raw uncompressed audio at 16kHz - use for real-time processing |
pcm_22050 |
Raw uncompressed audio at 22.05kHz |
pcm_24000 |
Raw uncompressed audio at 24kHz - good balance for streaming |
pcm_44100 |
Raw uncompressed audio at 44.1kHz (Pro+) - CD quality |
ulaw_8000 |
μ-law compressed 8kHz - standard for phone systems (Twilio, telephony) |
Streaming
For real-time applications:
audio_stream = client.text_to_speech.convert(
text="This text will be streamed as audio.",
voice_id="JBFqnCBsd6RMkjVDRZzb",
model_id="eleven_flash_v2_5" # Ultra-low latency
)
for chunk in audio_stream:
play_audio(chunk)
See references/streaming.md for WebSocket streaming.
Error Handling
try:
audio = client.text_to_speech.convert(
text="Generate speech",
voice_id="invalid-voice-id"
)
except Exception as e:
print(f"API error: {e}")
Common errors:
- 401: Invalid API key
- 422: Invalid parameters (check voice_id, model_id)
- 429: Rate limit exceeded
Tracking Costs
Monitor character usage via response headers (x-character-count, request-id):
response = client.text_to_speech.convert.with_raw_response(
text="Hello!", voice_id="JBFqnCBsd6RMkjVDRZzb", model_id="eleven_multilingual_v2"
)
audio = response.parse()
print(f"Characters used: {response.headers.get('x-character-count')}")