Voice Changer

SKILL音频生成
音频生成 未标注权限 许可证不明确

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.

一句话选型结论

按资料字段整理

Voice Changer是音频生成类的 Agent Skill,安装方式:GitHub 仓库,权限级别未标注,许可证不明确,没有现成安装命令,需按说明手动放置 SKILL.md。技能本质是给模型读的说明书,效果取决于模型与客户端;装完先用一个小任务验证,再放进正式流程。

分类音频生成 安装方式GitHub 仓库 权限级别未标注权限 许可证许可证不明确
累计安装 未标注 本周 未标注
GitHub 星标 未标注 仓库 skills
下载量 未标注 技能市场下载数
权限级别 未标注权限 安装前请确认风险范围
许可证 许可证不明确 未说明

安装

GitHub 仓库
在支持 Skills 的客户端里执行,装完问一句「你有哪些技能」验证
  1. 1
    确认客户端支持

    确认你的客户端支持 Skills(Claude、Codex、Cursor 等),并升级到较新的版本。

  2. 2
    拿到技能文件

    按安装说明或仓库 README,把 SKILL.md 放到客户端的 skills 目录(一般一个技能一个文件夹)。

  3. 3
    刷新技能列表

    重新打开客户端,或刷新技能列表,让新放进来的技能被加载。

  4. 4
    验证是否生效

    问一句「你有哪些技能」,确认列表里出现它,再拿一个小任务试跑。

技能档案

音频生成
字段来自技能仓库与公开索引,中文说明由本站标注
作者 / 维护方elevenlabs仓库 skills · 版本 未标注
宿主客户端未标注技能可被这些客户端加载
安装方式GitHub 仓库安装包 elevenlabs/skills
权限级别未标注权限安装前请确认风险范围
许可证未说明许可证不明确,商用前请自行确认条款
仓库可用性available技能仓库当前是否可访问
最近核验未标注上游最后核验时间 · 数据更新 2026-08-28
正文体量11.6 KBSKILL.md 原文大小,越大说明说明书写得越细

典型使用场景

3 类
按分类整理的常见用法,具体以 SKILL.md 为准
生成配音与音频

按文本产出配音或音效。

转写与整理

把音频转成文字并整理重点。

做音频脚本

按场景写口播与分镜脚本。

选择建议

装之前先看这几条
按条目字段给出的判断,不替你决定用不用

值得选的地方

  • 基础信息齐全,可以先按说明装一个试试。

先想清楚的地方

  • 没有标注权限级别,实际能做什么要装好后确认。
  • 许可证不明确,商用或二次分发前先到仓库确认条款。
  • 没有现成安装命令,要按安装说明或仓库 README 手动放置。
  • 没有做站内核验,内容以作者仓库为准。
  • 技能只是说明书,实际效果取决于模型与客户端,装完先用小任务试一遍。

更适合

  • 需要在音频生成场景里给模型补一套做法的人。
  • 愿意先读一遍 SKILL.md、确认它会让模型做什么的人。

不太适合

  • 对许可证要求很严、必须明确可商用的人。
  • 只想装完就自动变强、不打算看说明书的人。

同类对比

6 个同类
同分类里挑最接近的几条,字段以各自详情页为准
技能权限级别安装方式累计安装宿主客户端安装命令
Voice Changer当前 未标注权限 GitHub 仓库 未标注 未标注 无
下垂 接触敏感数据 ClawHub 市场 209.9 万 OpenClaw 有
新闻摘要 联网访问 ClawHub 市场 136.5 万 OpenClaw 有
OpenAI TTS 搜索 接触敏感数据 ClawHub 市场 22.3 万 OpenClaw 有
心 tts 接触敏感数据 ClawHub 市场 20.2 万 OpenClaw 有
语音转录 接触敏感数据 ClawHub 市场 19.6 万 OpenClaw 有

SKILL.md 全文

11.6 KB
模型实际读到的说明书原文,来自技能仓库

ElevenLabs Voice Changer

Transform the voice in an audio recording into a different target voice. Voice Changer (previously called Speech-to-Speech — the API endpoint and SDK methods still use the speech_to_speech / speechToSpeech name) keeps the original performance — emotion, pacing, intonation, breaths, whispers, laughs, cries — and only swaps who is speaking.

> Setup: See 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 outperforms eleven_english_sts_v2 even 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.mp3

Parameters

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=True or 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_id here 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, or file_format vs the supplied audio)
- 429: Rate limit exceeded

References

  • Installation Guide

常见问题

5 问
答案由本站依据条目字段整理
Voice Changer怎么安装?

没有现成安装命令:按安装说明或仓库 README,把 SKILL.md 放进客户端的 skills 目录(一般一个技能一个文件夹)。

Voice Changer支持哪些客户端?

资料没有标注宿主客户端,但只要客户端支持 Skills(Claude、Codex、Cursor 等),一般都能加载。

Voice Changer权限高吗,安全吗?

权限级别为未标注权限。安装前先看它会读取或写入哪些内容。

Voice Changer收费吗?

本站不代收费用,也没有标注具体价格。是否收费取决于技能本身和所用的模型或客户端,正式使用前按作者说明确认。

Voice Changer用什么许可证?

许可证标注为 未说明(许可证不明确)。商用或二次分发前请到仓库确认条款原文。

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技能内容以作者仓库为准;分类、权限级别与许可证状态的中文说明为中文整理,最近核验 未标注。