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Agent Skill · AI 工具

voice-ai-development(语音 AI 开发)

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。

voice-ai-development 是AI 工具类 Agent Skill,可用宿主为 Codex、Claude、Cursor、Gemini CLI,通过GitHub 仓库获取,权限为需要联网;许可证为 MIT,第三方扫描有未通过项,按安装量在AI 工具类的 2,536 个 Skill 里排第 348。

总安装 6,772周安装 285GitHub Stars 26,369下载 2,371核验 2026-05-01

安装说明

本站只整理中文说明,不托管安装包,也不代用户安装。
GitHub
许可证
MIT

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:voice-ai-development(语音 AI 开发)
来源仓库:https://github.com/davila7/claude-code-templates
仓库路径:skills/voice-ai-development
安装命令:
npx skills add https://github.com/davila7/claude-code-templates --skill voice-ai-development
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。不同来源提供的安装方式可能略有差异;本站展示可直接复制的安装命令,安装前请核对来源页面。

skills.shnpx skills
npx skills add https://github.com/davila7/claude-code-templates --skill voice-ai-development

这个 Skill 是做什么的

AI 工具类 Skill 用来扩展助手本身的能力,例如接模型、接检索、接外部服务。

用于辅助音频、音乐、语音转写、语音合成或声音素材处理,适合生成配乐说明或整理音频流程。适用于调用语音工具、处理播客和视频配音素材等场景。通过 npx skills add 命令从指定 GitHub 仓库安装并使用。使用时需确认输入音频来源、输出格式及时长限制,注意版权合规边界。

voice-ai-development 属于AI 工具类 Skill,可作为该场景下的辅助能力补充。

来源仓库内路径:skills/voice-ai-development

上手步骤

4 步
  1. 1

    确认自己的宿主工具

    公开资料标注支持 Codex、Claude、Cursor、Gemini CLI;先确认自己用的是其中之一,再按对应工具的 Skills 目录放置。

  2. 2

    拿到安装命令

    可以直接复制的命令是 npx skills add https://github.com/davila7/claude-code-templates --skill voice-ai…;命令里的占位符换成自己的路径或密钥再执行。

  3. 3

    安装后重启宿主

    命令执行完要完全退出宿主工具再打开,让新装的 Skill 被重新扫描到,否则列表里可能看不到。

  4. 4

    按最小权限试用

    它需要需要联网,先在测试目录或测试账号里跑,确认没问题再放到正式项目。

常见问题

6 条

voice-ai-development 怎么安装?

在终端执行 npx skills add https://github.com/davila7/claude-code-templates --skill voice-ai-developme…。装完完全退出宿主工具再打开,让它重新扫描本地 Skill。

voice-ai-development 支持哪些 AI 工具?

公开资料标注支持 Codex、Claude、Cursor、Gemini CLI。其他工具只要能读取本地 Skill 目录,一般也能手动放进去,但要自己确认触发方式是否一致。

voice-ai-development 需要什么权限?

权限级别是需要联网。该 Skill 可能需要联网访问来源站点、仓库或外部 API;具体网络访问范围需要结合源码和 README 复核。

voice-ai-development 是免费的吗?

来源标注的许可证是 MIT。本站不代收任何费用,也不托管安装包;能否商用、要不要署名,以许可证原文为准。

voice-ai-development 用的人多吗?

公开数据是累计安装 6,772 次,近一周安装 285 次,下载 2,371 次,GitHub 26,369 星。这些数字来自第三方目录统计,只能当作热度参考,不代表质量或安全性。

voice-ai-development 安全吗?

第三方扫描有未通过项,风险较高。除非你确认过源码,否则不建议直接安装到常用环境。

值得选的地方和要先想清楚的地方

值得选的地方

  • 同时适配 4 种工具,换客户端不用重新找替代品。
  • 3 个目录同时收录,说明这个 Skill 流传比较广。
  • 许可证明确为 MIT,商用和分发边界好判断。
  • 来源仓库有 26,369 星,维护活跃度相对可查。

先想清楚的地方

  • 第三方扫描存在提醒项或未通过项,需要人工逐条确认。
  • 第三方目录整理,版本更新和行为细节以来源仓库为准。

适不适合你

更适合

  • 已经在用 Codex,想直接扩展它能力的人。
  • 需要处理AI 工具类任务、又不想每次都手写提示词的人。

不太适合

  • 需要厂商级技术支持和 SLA 的生产系统。

AI 工具类 Skill 对比

6 个同类
Skill总安装GitHub 星权限许可证扫描
executing-plans

执行计划 · Codex、Claude

1,515,336173,110只读MIT通过
verification-before-completion

完成前的验证 · Codex、Claude

1,319,472173,121权限未标注MIT通过
subagent-driven-development

子 Agent 驱动开发 · Codex、Claude

1,312,416173,138只读MIT通过
copy-editing

复制编辑 · Codex、Claude

1,209,57625,632只读MIT通过
paywall-upgrade-cro

付费专区升级 cro · Codex、Claude

1,030,17625,617只读MIT通过
voice-ai-development

语音 AI 开发 · 本页

6,77226,369需要联网MIT未通过

按安装量取同分类前几名;数字来自第三方目录统计,只作热度参考,不代表质量结论。

SKILL.md 原文

Voice AI Development

Role: Voice AI Architect

You are an expert in building real-time voice applications. You think in terms of latency budgets, audio quality, and user experience. You know that voice apps feel magical when fast and broken when slow. You choose the right combination of providers for each use case and optimize relentlessly for perceived responsiveness.

Capabilities

  • OpenAI Realtime API
  • Vapi voice agents
  • Deepgram STT/TTS
  • ElevenLabs voice synthesis
  • LiveKit real-time infrastructure
  • WebRTC audio handling
  • Voice agent design
  • Latency optimization

Requirements

  • Python or Node.js
  • API keys for providers
  • Audio handling knowledge

Patterns

OpenAI Realtime API

Native voice-to-voice with GPT-4o

When to use: When you want integrated voice AI without separate STT/TTS

import asyncio
import websockets
import json
import base64

OPENAI_API_KEY = "sk-..."

async def voice_session():
    url = "wss://api.openai.com/v1/realtime?model=gpt-4o-realtime-preview"
    headers = {
        "Authorization": f"Bearer {OPENAI_API_KEY}",
        "OpenAI-Beta": "realtime=v1"
    }

    async with websockets.connect(url, extra_headers=headers) as ws:
        # Configure session
        await ws.send(json.dumps({
            "type": "session.update",
            "session": {
                "modalities": ["text", "audio"],
                "voice": "alloy",  # alloy, echo, fable, onyx, nova, shimmer
                "input_audio_format": "pcm16",
                "output_audio_format": "pcm16",
                "input_audio_transcription": {
                    "model": "whisper-1"
                },
                "turn_detection": {
                    "type": "server_vad",  # Voice activity detection
                    "threshold": 0.5,
                    "prefix_padding_ms": 300,
                    "silence_duration_ms": 500
                },
                "tools": [
                    {
                        "type": "function",
                        "name": "get_weather",
                        "description": "Get weather for a location",
                        "parameters": {
                            "type": "object",
                            "properties": {
                                "location": {"type": "string"}
                            }
                        }
                    }
                ]
            }
        }))

        # Send audio (PCM16, 24kHz, mono)
        async def send_audio(audio_bytes):
            await ws.send(json.dumps({
                "type": "input_audio_buffer.append",
                "audio": base64.b64encode(audio_bytes).decode()
            }))

        # Receive events
        async for message in ws:
            event = json.loads(message)

            if event["type"] == "resp

Vapi Voice Agent

Build voice agents with Vapi platform

When to use: Phone-based agents, quick deployment

# Vapi provides hosted voice agents with webhooks

from flask import Flask, request, jsonify
import vapi

app = Flask(__name__)
client = vapi.Vapi(api_key="...")

# Create an assistant
assistant = client.assistants.create(
    name="Support Agent",
    model={
        "provider": "openai",
        "model": "gpt-4o",
        "messages": [
            {
                "role": "system",
                "content": "You are a helpful support agent..."
            }
        ]
    },
    voice={
        "provider": "11labs",
        "voiceId": "21m00Tcm4TlvDq8ikWAM"  # Rachel
    },
    firstMessage="Hi! How can I help you today?",
    transcriber={
        "provider": "deepgram",
        "model": "nova-2"
    }
)

# Webhook for conversation events
@app.route("/vapi/webhook", methods=["POST"])
def vapi_webhook():
    event = request.json

    if event["type"] == "function-call":
        # Handle tool call
        name = event["functionCall"]["name"]
        args = event["functionCall"]["parameters"]

        if name == "check_order":
            result = check_order(args["order_id"])
            return jsonify({"result": result})

    elif event["type"] == "end-of-call-report":
        # Call ended - save transcript
        transcript = event["transcript"]
        save_transcript(event["call"]["id"], transcript)

    return jsonify({"ok": True})

# Start outbound call
call = client.calls.create(
    assistant_id=assistant.id,
    customer={
        "number": "+1234567890"
    },
    phoneNumber={
        "twilioPhoneNumber": "+0987654321"
    }
)

# Or create web call
web_call = client.calls.create(
    assistant_id=assistant.id,
    type="web"
)
# Returns URL for WebRTC connection

Deepgram STT + ElevenLabs TTS

Best-in-class transcription and synthesis

When to use: High quality voice, custom pipeline

import asyncio
from deepgram import DeepgramClient, LiveTranscriptionEvents
from elevenlabs import ElevenLabs

# Deepgram real-time transcription
deepgram = DeepgramClient(api_key="...")

async def transcribe_stream(audio_stream):
    connection = deepgram.listen.live.v("1")

    async def on_transcript(result):
        transcript = result.channel.alternatives[0].transcript
        if transcript:
            print(f"Heard: {transcript}")
            if result.is_final:
                # Process final transcript
                await handle_user_input(transcript)

    connection.on(LiveTranscriptionEvents.Transcript, on_transcript)

    await connection.start({
        "model": "nova-2",  # Best quality
        "language": "en",
        "smart_format": True,
        "interim_results": True,  # Get partial results
        "utterance_end_ms": 1000,
        "vad_events": True,  # Voice activity detection
        "encoding": "linear16",
        "sample_rate": 16000
    })

    # Stream audio
    async for chunk in audio_stream:
        await connection.send(chunk)

    await connection.finish()

# ElevenLabs streaming synthesis
eleven = ElevenLabs(api_key="...")

def text_to_speech_stream(text: str):
    """Stream TTS audio chunks."""
    audio_stream = eleven.text_to_speech.convert_as_stream(
        voice_id="21m00Tcm4TlvDq8ikWAM",  # Rachel
        model_id="eleven_turbo_v2_5",  # Fastest
        text=text,
        output_format="pcm_24000"  # Raw PCM for low latency
    )

    for chunk in audio_stream:
        yield chunk

# Or with WebSocket for lowest latency
async def tts_websocket(text_stream):
    async with eleven.text_to_speech.stream_async(
        voice_id="21m00Tcm4TlvDq8ikWAM",
        model_id="eleven_turbo_v2_5"
    ) as tts:
        async for text_chunk in text_stream:
            audio = await tts.send(text_chunk)
            yield audio

        # Flush remaining audio
        final_audio = await tts.flush()
        yield final_audio

Anti-Patterns

❌ Non-streaming Pipeline

Why bad: Adds seconds of latency. User perceives as slow. Loses conversation flow.

Instead: Stream everything:

  • STT: interim results
  • LLM: token streaming
  • TTS: chunk streaming Start TTS before LLM finishes.

❌ Ignoring Interruptions

Why bad: Frustrating user experience. Feels like talking to a machine. Wastes time.

Instead: Implement barge-in detection. Use VAD to detect user speech. Stop TTS immediately. Clear audio queue.

❌ Single Provider Lock-in

Why bad: May not be best quality. Single point of failure. Harder to optimize.

Instead: Mix best providers:

  • Deepgram for STT (speed + accuracy)
  • ElevenLabs for TTS (voice quality)
  • OpenAI/Anthropic for LLM

Limitations

  • Latency varies by provider
  • Cost per minute adds up
  • Quality depends on network
  • Complex debugging

Related Skills

Works well with: langgraph, structured-output, langfuse

平台分布

Claude Code

29.78%
按下载量换算706

OpenCode

25.05%
按下载量换算594

Cursor

16.63%
按下载量换算394

Antigravity

13.13%
按下载量换算311

Gemini CLI

7.17%
按下载量换算170

Codex

3.39%
按下载量换算80

安全审计

Gen Agent Trust Hub

通过

Socket

通过

Snyk

未通过

权限和风险

需要联网

该 Skill 可能需要联网访问来源站点、仓库或外部 API;具体网络访问范围需要结合源码和 README 复核。

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。来源安全扫描存在提醒或未通过项,安装前请自行复核。

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