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liber-speechapi言论自由 API

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。

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OpenClaw

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

MIT-0

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:liber-speechapi(言论自由 API)
来源仓库:https://github.com/liberalchang/liber-speechapi
安装命令:
openclaw skills install liber-speechapi
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 OpenClaw 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。

ClawHubOpenClaw
openclaw skills install liber-speechapi

简介

liber-speechapi 集成 ASR 与 TTS 能力,支持语音消息转写与文本合成输出。

  • 适用于播客剪辑、视频配音素材生成或会议记录自动化处理的场景。
  • 可处理 Telegram 语音消息,并支持自定义音色与环境参数调整。
  • 涉及人声克隆或公开发布内容时,必须获得授权并遵守版权法规。
  • 音频处理受模型限制,长段语音可能分段处理,需预留足够时间与资源。

SKILL.md

name
liber-speechapi
description
Use Liber SpeechAPI for three speech workflows: (1) Telegram/openclaw voice-message handling with ASR, concise reply summarization, and Telegram-compatible OGG/Opus TTS, (2) direct text-to-speech generation when the user explicitly asks to synthesize text into audio, and (3) direct speech-to-text transcription when the user explicitly asks to convert audio into text or structured JSON. Support environment-based configuration, optional voice cloning from a reference audio file, and fallback between shared python-env skill and the local Python environment.

Liber SpeechAPI

Use this skill for three related tasks:

  • handle Telegram/openclaw voice-message workflows end to end
  • convert user-provided text to speech on demand
  • convert user-provided audio to text on demand

Follow this workflow

  1. Read references/config.md to resolve configuration from .env and config.json.
  2. Read references/workflow.md for Telegram/openclaw voice-message handling.
  3. Read references/api.md when you need endpoint and payload details.
  4. Read references/parameters.md for detailed ASR/TTS parameter meanings and defaults.
  5. Use scripts/summarize_for_voice.py only when a reply must be shortened for voice playback.
  6. Use scripts/liber_speech_client.py for deterministic ASR/TTS calls instead of rewriting HTTP request logic.

Environment selection

Prefer a shared python-env skill if it is available in the current environment.

If python-env is not available, use the local Python environment for this skill.

When running local Python commands:

  • use Python 3.11 if available
  • allow Python 3.10 when 3.11 is unavailable
  • install only the minimal dependencies required by the bundled scripts
  • do not hardcode secrets; read them from .env

Configuration model

.env

Load core service settings from .env in priority order:

  1. environment variables (LIBER_API_BASE_URL and LIBER_API_KEY)
  2. ~/.openclaw/.env file (for global configuration)
  3. the skill directory's .env file
  4. the current working directory's .env file

Environment variables take the highest priority, followed by the global config file ~/.openclaw/.env, then local skill directory, and finally the current working directory.

Required settings:

  • LIBER_API_BASE_URL
  • LIBER_API_KEY

config.json

Load detailed defaults from speechapi_config.json in ~/.openclaw/workspace/config/ to prevent overwrites during skill updates. Fallback to local config.json if the external config doesn't exist.

Key behavior:

  • values of "default" or null are omitted from API requests
  • Telegram-specific voice replies use global.telegram_tts_format
  • direct text-to-speech uses tts.format as its default output format
  • direct speech-to-text uses global.asr_output as its default output mode

Direct text-to-speech

When the user explicitly asks to convert text to speech:

  1. use scripts/liber_speech_client.py tts
  2. default to wav unless the caller explicitly requests another format
  3. include audio_prompt only when clone audio is enabled and the file exists
  4. return the TTS result URL or saved output path to the caller

Direct speech-to-text

When the user explicitly asks to convert audio to text:

  1. use scripts/liber_speech_client.py asr
  2. default to structured json output
  3. return plain text only when the caller explicitly wants transcript text only

Telegram/openclaw workflow

For incoming Telegram voice/audio:

  1. download or access the local audio file
  2. send it to ASR and extract the recognized text
  3. send the transcript to openclaw
  4. if the final reply is too long for voice, shorten it to within the configured summary limit
  5. synthesize the final spoken reply with Telegram-compatible ogg_opus
  6. return the resulting audio URL or saved output path to the caller

Telegram-specific guidance

For Telegram voice replies:

  • force ogg_opus output
  • keep spoken output concise and natural
  • if the original answer is verbose, preserve intent and key facts but compress aggressively
  • avoid reading markdown, code blocks, tables, or long lists verbatim

Safety and robustness

  • never print or log API keys
  • validate input file existence before ASR
  • validate text is non-empty before TTS
  • use request timeouts
  • handle HTTP failures with clear error messages
  • if TTS clone audio is configured but missing, continue without cloning instead of failing
  • if summarization fails, fall back to conservative truncation rather than blocking the reply
  • default direct ASR output to JSON and default direct TTS output to WAV unless the caller requests otherwise

Expected outputs

Depending on the task, return one of:

  • structured ASR JSON
  • plain transcript text
  • concise voice-ready text
  • TTS result URL
  • saved audio file path
  • a structured JSON object containing transcript, summary, and synthesis result

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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

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