文字转语音

SKILL开发

原名 text-to-speech

开发 接触敏感数据 许可证不明确 CodexClaudeCursorGemini CLI
安装这个技能

用于辅助音频、音乐、语音转写、语音合成或声音素材处理。它适合让 Agent 生成配乐说明、整理音频流程、调用语音工具或处理播客和视频配音素材。使用时需要确认输入音频来源、输出格式、时长和模型限制;涉及人声克隆、版权音乐或公开发布时,应先核对授权和合规边界。

累计安装 97,920 本周 4,012
GitHub 星标 194 仓库 skills
下载量 31,680 技能市场下载数
权限级别 接触敏感数据 安装前请确认风险范围
许可证 许可证不明确 unknown

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
npx skills add https://github.com/elevenlabs/skills --skill text-to-speech

技能档案

开发
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Codex、Claude、Cursor、Gemini CLI技能可被这些客户端加载
安装方式GitHub 仓库安装包 elevenlabs/skills
权限级别接触敏感数据安装前请确认风险范围
许可证unknown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2026-05-01上游最后核验时间 · 数据更新 未标注
正文体量6.9 KBSKILL.md 原文大小,越大说明说明书写得越细

能做什么

4 项能力
上游给出的能力概览与适用场景
  • 按任务关键词查找相关 Skills
  • 展示可复制的安装命令
  • 保留来源站点、仓库和原始说明,方便继续核验
  • 展示第三方安全扫描或审计结果
用户想查找某类 Agent Skill 时需要根据任务场景推荐可安装能力包时需要对比不同来源的安装命令和来源信息时

权限与风险

接触敏感数据
装之前先读完这段

安装与使用指南

安装步骤

本站只展示命令,不托管安装包,也不自动执行。

怎么用

安装后应在对应宿主中按原始 README 的触发条件使用;当前草稿只做目录说明,具体调用方式由规则抽取,异常条目再复核。

SKILL.md 全文

6.9 KB
模型实际读到的说明书原文,来自技能仓库

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:

  • 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,
        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":

  • "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
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:

  • 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')}")

References

安全信息

审计结论
未标注不代表已通过审计
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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 2026-05-01。