voice-changer
Skill527 starsChanged 14 days agoMITView on GitHub
Transform the voice in an audio recording into a different target voice while preserving emotion, timing, and delivery using the ElevenLabs Voice Changer (speech-to-speech) API. Use when converting one voice to another, changing the speaker/narrator of an existing recording, dubbing a voice-over in a different voice, creating character voices from a scratch performance, anonymizing a speaker, or any "voice conversion / voice transfer / speech-to-speech" task. Make sure to use this skill whenever the user mentions voice changing, voice conversion, speech-to-speech, swapping a voice in audio, re-voicing a clip, or applying a different voice to an existing recording — even if they don't explicitly say "voice changer". Do not use this skill to create or clone a new voice from a voice sample — that is voice cloning (IVC/PVC), a separate feature; this skill only converts an existing recording into an existing voice_id.
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- Common
- Distinctive
- Very specific
- Very specific: line 94
Only models whose can_do_voice_conversion property is true can be used here.
- Distinctive: line 102
Use any voice ID from pre-made voices, your cloned voices, or the voice library.
- Very specific: line 168
If the input recording is noisy, either pre-process with the voice-isolator skill or pass remove_background_noise=True to do it in a single …
- Very specific: line 183
If you already have raw 16-bit PCM mono @ 16kHz, passing file_format="pcm_s16le_16" skips decoding and reduces latency:
≈ “Common” is an estimate: at least 1 in 5 of the 308,232 skill files we read have a rule of the same kind, not the same words. “Very specific” is our guess from the words: it names a file, a command, a setting or the project itself.
Files
ElevenLabs Voice Changer
Transform the voice in an audio recording into a different target voice. Voice Changer (previously called Speech-to-Speech — the REST endpoints are POST /v1/voice-changer/{voice_id} and /v1/voice-changer/{voice_id}/stream, while the SDK methods and CLI command still use the speech_to_speech / speechToSpeech / speech-to-speech name) keeps the original performance — emotion, pacing, intonation, breaths, whispers, laughs, cries — and only swaps who is speaking.
Setup, before writing any code: if
ELEVENLABS_API_KEYisn't set for this project, runnpx -y @elevenlabs/cli@latest onboard initfrom the app's folder first and follow what it prints: it signs the user in and stores a key for the project, so nobody copies one by hand. Otherwise see the Installation Guide. For JavaScript, use@elevenlabs/*packages only.
Key Facts
- Maximum input length: 5 minutes per request — split longer recordings into chunks and stitch the outputs.
- Maximum file size: 50 MB per request — compress to MP3 if your source is larger.
- Pricing: 1,000 characters per minute of audio processed (duration-based, not text-based).
- Recommended model:
eleven_multilingual_sts_v2— often outperformseleven_english_sts_v2even for English-only content.
Quick Start
Python
from elevenlabs import ElevenLabs
client = ElevenLabs()
with open("source.mp3", "rb") as audio_file:
audio_stream = client.speech_to_speech.convert(
voice_id="JBFqnCBsd6RMkjVDRZzb", # George
audio=audio_file,
model_id="eleven_multilingual_sts_v2",
output_format="mp3_44100_128",
)
with open("converted.mp3", "wb") as f:
for chunk in audio_stream:
f.write(chunk)JavaScript
import { ElevenLabsClient } from "@elevenlabs/elevenlabs-js";
import { createReadStream, createWriteStream } from "fs";
const client = new ElevenLabsClient();
const audioStream = await client.speechToSpeech.convert("JBFqnCBsd6RMkjVDRZzb", {
audio: createReadStream("source.mp3"),
modelId: "eleven_multilingual_sts_v2",
outputFormat: "mp3_44100_128",
});
audioStream.pipe(createWriteStream("converted.mp3"));CLI
elevenlabs speech-to-speech convert \
--voice-id JBFqnCBsd6RMkjVDRZzb \
--audio source.mp3 \
--model-id eleven_multilingual_sts_v2 \
--output-format mp3_44100_128 \
--output converted.mp3Parameters
| Parameter | Type | Default | Description |
voice_id | string (required) | — | Target voice to speak in. Use a pre-made voice ID, a cloned voice, or a voice from the library |
audio | file (required) | — | Source audio whose performance (emotion, timing, delivery) will be preserved |
model_id | string | eleven_english_sts_v2 | eleven_multilingual_sts_v2 for 29 languages, eleven_english_sts_v2 for English-only |
output_format | string | mp3_44100_128 | See output formats table below |
voice_settings | JSON string | — | Override stored voice settings for this request only |
seed | integer | — | Best-effort deterministic sampling (0 – 4294967295) |
remove_background_noise | boolean | false | Run the isolation model on the input before conversion |
file_format | string | other | other for any encoded audio, or pcm_s16le_16 for 16-bit PCM mono @ 16kHz little-endian (lower latency) |
optimize_streaming_latency | int (query) | — | 0–4. Trade quality for latency. 4 is fastest but disables the text normalizer |
enable_logging | boolean (query) | true | Set to false for zero-retention mode (enterprise only — disables history/stitching) |
Models
| Model ID | Languages | Best For |
eleven_multilingual_sts_v2 | 29 | Recommended for everything — often outperforms the English model even on English audio |
eleven_english_sts_v2 | English | API default — English-only fallback |
Only models whose can_do_voice_conversion property is true can be used here. Voice Changer does not currently have a low-latency "flash/turbo" tier — if you need one, keep pcm_s16le_16 input, an opus_* / low-bitrate mp3_* output, and raise optimize_streaming_latency.
Languages (eleven_multilingual_sts_v2)
English (US, UK, AU, CA), Japanese, Chinese, German, Hindi, French (FR, CA), Korean, Portuguese (BR, PT), Italian, Spanish (ES, MX), Indonesian, Dutch, Turkish, Filipino, Polish, Swedish, Bulgarian, Romanian, Arabic (SA, AE), Czech, Greek, Finnish, Croatian, Malay, Slovak, Danish, Tamil, Ukrainian, Russian.
Target Voices
Use any voice ID from pre-made voices, your cloned voices, or the voice library.
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}")Converting from a URL
import requests
from io import BytesIO
from elevenlabs import ElevenLabs
client = ElevenLabs()
audio_url = "https://storage.googleapis.com/eleven-public-cdn/audio/marketing/nicole.mp3"
response = requests.get(audio_url)
audio_data = BytesIO(response.content)
audio_stream = client.speech_to_speech.convert(
voice_id="JBFqnCBsd6RMkjVDRZzb",
audio=audio_data,
model_id="eleven_multilingual_sts_v2",
output_format="mp3_44100_128",
)
with open("converted.mp3", "wb") as f:
for chunk in audio_stream:
f.write(chunk)Voice Settings Override
Fine-tune the target voice for a single request without changing its stored defaults:
from elevenlabs import VoiceSettings
audio_stream = client.speech_to_speech.convert(
voice_id="JBFqnCBsd6RMkjVDRZzb",
audio=audio_file,
model_id="eleven_multilingual_sts_v2",
voice_settings=VoiceSettings(
stability=0.5,
similarity_boost=0.75,
style=0.0,
use_speaker_boost=True,
),
)- Stability: lower = more emotional range (follows the source more freely), higher = steadier delivery.
- Similarity boost: higher = closer to the target voice's timbre, may amplify source artifacts.
- Style: exaggerates the target voice's unique characteristics (v2+ models).
- Speaker boost: post-processing to sharpen clarity of the target voice.
Cleaning Up Noisy Source Audio
If the input recording is noisy, either pre-process with the voice-isolator skill or pass remove_background_noise=True to do it in a single call:
audio_stream = client.speech_to_speech.convert(
voice_id="JBFqnCBsd6RMkjVDRZzb",
audio=audio_file,
model_id="eleven_multilingual_sts_v2",
remove_background_noise=True,
)Cleaner input almost always produces better conversion — the model is trying to match phonemes and prosody, and background noise gets in the way.
Low-Latency PCM Input
If you already have raw 16-bit PCM mono @ 16kHz, passing file_format="pcm_s16le_16" skips decoding and reduces latency:
audio_stream = client.speech_to_speech.convert(
voice_id="JBFqnCBsd6RMkjVDRZzb",
audio=pcm_bytes,
model_id="eleven_multilingual_sts_v2",
file_format="pcm_s16le_16",
)Pair this with optimize_streaming_latency (0–4) as a query param for further latency reductions at some quality cost.
Output Formats
| Format | Description |
mp3_44100_128 | MP3 44.1kHz 128kbps (default) — good for web/apps |
mp3_44100_192 | MP3 44.1kHz 192kbps (Creator+) — higher quality |
mp3_44100_64 | MP3 44.1kHz 64kbps — smaller files |
mp3_22050_32 | MP3 22.05kHz 32kbps — smallest MP3 |
pcm_16000 | Raw PCM 16kHz — real-time pipelines |
pcm_24000 | Raw PCM 24kHz — good streaming balance |
pcm_44100 | Raw PCM 44.1kHz (Pro+) — CD quality |
pcm_48000 | Raw PCM 48kHz (Pro+) — highest quality |
ulaw_8000 | μ-law 8kHz — Twilio / telephony |
alaw_8000 | A-law 8kHz — telephony |
opus_48000_64 | Opus 48kHz 64kbps — efficient streaming |
Deterministic Output
Pass a seed to make repeated conversions of the same input return (best-effort) identical audio — useful for testing and A/B comparisons.
audio_stream = client.speech_to_speech.convert(
voice_id="JBFqnCBsd6RMkjVDRZzb",
audio=audio_file,
model_id="eleven_multilingual_sts_v2",
seed=12345,
)Input Audio Best Practices
The conversion quality is bounded by the input recording — the model can only swap the timbre, not rescue a bad source. A few practical rules:
- Be expressive. Whisper, shout, laugh, cry — the model preserves all of it. Flat input gives you flat output.
- Watch microphone gain. Too quiet and the model under-detects phonemes; too loud and clipping bleeds into the conversion. Aim for healthy peaks, no clipping.
- Accent and cadence transfer from the source, not the target. If you read in an American accent and target the British "George" voice, you get George's timbre with an American accent. To dub into a different accent or language, record someone speaking in that target accent/language and convert into a cloned/library voice.
- Clean up noise first. Either pass
remove_background_noise=Trueor run the source through the voice-isolator skill before conversion. Noise hurts more here than in TTS. - Split long recordings. Anything over 5 minutes must be chunked. Cut at natural pauses, convert each piece, and concatenate the resulting audio.
Common Workflows
- Re-voice a narration — keep the performance of a scratch recording, swap in a different narrator voice.
- Localize / dub — convert a voice-over into the same speaker's cloned voice in another language (using
eleven_multilingual_sts_v2). - Create character voices — act out a line yourself, convert into a distinctive character voice for games or animation.
- Anonymize a speaker — replace a recognizable voice with a neutral pre-made voice while preserving what was said and how.
- Pair with voice-isolator — isolate the source voice first (or set
remove_background_noise=True) for noisy field recordings before conversion. - Pair with voice cloning — clone a target voice from a short sample, then use its
voice_idhere as the conversion target.
Error Handling
try:
audio_stream = client.speech_to_speech.convert(
voice_id="JBFqnCBsd6RMkjVDRZzb",
audio=audio_file,
model_id="eleven_multilingual_sts_v2",
)
except Exception as e:
print(f"Voice changer failed: {e}")Common errors:
- 401: Invalid API key
- 422: Invalid parameters (check
voice_id,model_id, orfile_formatvs the supplied audio) - 429: Rate limit exceeded
References
- Installation Guide
Other versions3 projects use a version of this.
Word for word · 1
- RobStrayer/voice-agent-skills★ 1100% same
Edited · 1
- elevenlabs/plugin★ 399% same
Wrote their own version · 1
- wlzh/skills★ 694same name
How matches are found. Word for word: the exact same text. Edited: shares passages with this file (“99% same” means 99% of its five-word phrases also appear here; 10% or more counts). Their own version: similar meaning, or, weakest, the same name. We can’t tell which project had it first. Projects are counted among the 200 most-starred candidates, and mentions are links to the repository we found, up to 50 per site, so both counts are a floor.
In this repository12 other files it gives its agents.
- agentsSkill
- update-skills-from-changelogSkill
- dubbingSkill
- musicSkill
- onboardingSkill
- realtimeSkill
- setup-api-keySkill
- sound-effectsSkill
- speech-engineSkill
- speech-to-textSkill
- text-to-speechSkill
- voice-isolatorSkill
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