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text-to-speech文字转语音

Agent Skill

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。

总安装

1,327

周安装

57

GitHub Stars

14

下载量

465
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安装说明

本站只整理中文说明和来源信息,不托管安装包,也不代用户安装。

GitHub

来源数

2

许可证

unknown

最后核验

2026-05-01

来源状态

来源可访问

安装方式

通过对话安装

复制提示词发给支持本地命令或 Skills 的 AI 助手,先确认命令和权限,再让它执行。

请帮我安装这个 Agent Skill:text-to-speech(文字转语音)
来源仓库:https://github.com/connorads/dotfiles
仓库路径:skills/text-to-speech
安装命令:
npx skills add https://github.com/connorads/dotfiles --skill text-to-speech
安装前请先检查当前环境是否支持对应 CLI,并向我确认将要执行的命令、安装目录、联网范围和文件读写权限;确认后再执行。

命令行安装

复制命令到本机终端执行。该命令会通过 npx skills 从第三方来源获取 Skill;本站只展示命令,不托管安装包,也不自动执行。

skills.shnpx skills
npx skills add https://github.com/connorads/dotfiles --skill text-to-speech

简介

text-to-speech 用于生成自然语音,支持 70 多种语言和不同质量/延迟模型选择。

  • 它基于 ElevenLabs API,适用于旁白、有声书或交互式语音产品。
  • 使用时需提供文本、音色 ID 和模型参数,注意字符长度限制。
  • 涉及多语言或特定口音时应验证模型可用性。
  • 适用宿主包括 Codex、Claude、Cursor、Gemini CLI,接入前应确认版本、权限和运行环境要求。

SKILL.md

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 IDLanguagesLatencyBest For
eleven_v370+StandardHighest quality, emotional range
eleven_multilingual_v229StandardHigh quality, long-form content
eleven_flash_v2_532~75msUltra-low latency, real-time
eleven_flash_v2English~75msEnglish-only, fastest
eleven_turbo_v2_532~250-300msBalanced quality/speed
eleven_turbo_v2English~250-300msEnglish-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

FormatDescription
mp3_44100_128MP3 44.1kHz 128kbps (default) - compressed, good for web/apps
mp3_44100_192MP3 44.1kHz 192kbps (Creator+) - higher quality compressed
mp3_44100_64MP3 44.1kHz 64kbps - lower quality, smaller files
mp3_22050_32MP3 22.05kHz 32kbps - smallest MP3 files
pcm_16000Raw PCM 16kHz - use for real-time processing
pcm_22050Raw PCM 22.05kHz
pcm_24000Raw PCM 24kHz - good balance for streaming
pcm_44100Raw PCM 44.1kHz (Pro+) - CD quality
pcm_48000Raw PCM 48kHz (Pro+) - highest quality
ulaw_8000μ-law 8kHz - standard for phone systems (Twilio, telephony)
alaw_8000A-law 8kHz - telephony (alternative to μ-law)
opus_48000_64Opus 48kHz 64kbps - efficient streaming codec
wav_44100WAV 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

适合场景

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02

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安装后应在对应宿主中按原始 README 的触发条件使用;具体调用方式请以来源页面和 README 为准。

平台分布

Codex

35.37%
按下载量换算164

Claude

28.39%
按下载量换算132

Cursor

18.54%
按下载量换算86

Gemini CLI

9.55%
按下载量换算44

安全审计

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通过

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通过

Snyk

通过

权限和风险

敏感数据

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安装前确认

本站仅展示第三方公开信息,不托管安装包,不提供自动安装或运行环境。安装前应自行审查源码、依赖和命令行为。当前只有一个来源,正式发布前建议补源仓库或其他目录站核验。

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