ElevenLabs Text-to-Speech
Generate natural speech from text - supports 70+ languages, multiple models for quality vs latency tradeoffs.
Setup: See Installation Guide. For JavaScript, use @elevenlabs/* packages only.
Quick Start
Python
from elevenlabs 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
70+ Standard Highest quality, emotional range
eleven_multilingual_v2
29 Standard High quality, long-form content
eleven_flash_v2_5
32 ~75ms Ultra-low latency, real-time
eleven_flash_v2
English ~75ms English-only, fastest
eleven_turbo_v2_5
32 ~250-300ms Balanced quality/speed
eleven_turbo_v2
English ~250-300ms English-only, balanced
Voice IDs
Use pre-made voices or create custom voices in the dashboard.
Popular voices:
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JBFqnCBsd6RMkjVDRZzb
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George (male, narrative)
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EXAVITQu4vr4xnSDxMaL
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Sarah (female, soft)
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onwK4e9ZLuTAKqWW03F9
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Daniel (male, authoritative)
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XB0fDUnXU5powFXDhCwa
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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:
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Stability: How consistent the voice stays. Lower values = more emotional range and variation, but can sound unstable. Higher = steady, predictable delivery.
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Similarity boost: How closely to match the original voice sample. Higher values sound more like the original but may amplify audio artifacts.
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Style: Exaggerates the voice's unique style characteristics (only works with v2+ models).
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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, speed=1.0, # 0.25 to 4.0 (default 1.0) 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":
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"auto" (default): Model decides based on context
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"on" : Always normalize (use when you want natural speech)
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"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
mp3_44100_64
MP3 44.1kHz 64kbps - lower quality, smaller files
mp3_22050_32
MP3 22.05kHz 32kbps - smallest MP3 files
pcm_16000
Raw PCM 16kHz - use for real-time processing
pcm_22050
Raw PCM 22.05kHz
pcm_24000
Raw PCM 24kHz - good balance for streaming
pcm_44100
Raw PCM 44.1kHz (Pro+) - CD quality
pcm_48000
Raw PCM 48kHz (Pro+) - highest quality
ulaw_8000
μ-law 8kHz - standard for phone systems (Twilio, telephony)
alaw_8000
A-law 8kHz - telephony (alternative to μ-law)
opus_48000_64
Opus 48kHz 64kbps - efficient streaming codec
wav_44100
WAV 44.1kHz - uncompressed with headers
Streaming
For real-time applications, use the stream method (returns audio chunks as they're generated):
audio_stream = client.text_to_speech.stream( 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:
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401: Invalid API key
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422: Invalid parameters (check voice_id, model_id)
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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')}")
References
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Installation Guide
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Streaming Audio
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Voice Settings