Files
gradio-webrtc/README_FASTRTC.md
neil.xh f476f9cf29 gs对话接入
本次代码评审新增并完善了gs视频聊天功能,包括前后端接口定义、状态管理及UI组件实现,并引入了新的依赖库以支持更多互动特性。
Link: https://code.alibaba-inc.com/xr-paas/gradio_webrtc/codereview/21273476
* 更新python 部分

* 合并videochat前端部分

* Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 替换audiowave

* 导入路径修改

* 合并websocket mode逻辑

* feat: gaussian avatar chat

* 增加其他渲染的入参

* feat: ws连接和使用

* Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 右边距离超出容器宽度,则向左移动

* 配置传递

* Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 高斯包异常

* 同步webrtc_utils

* 更新webrtc_utils

* 兼容on_chat_datachannel

* 修复设备名称列表没有正常显示的问题

* copy 传递 webrtc_id

* Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 保证webrtc 完成后再进行websocket连接

* feat: 音频表情数据接入

* dist 上传

* canvas 隐藏

* feat: 高斯文件下载进度透出

* Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 修改无法获取权限问题

* Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 先获取权限再获取设备

* fix: gs资源下载完成前不处理ws数据

* fix: merge

* 话术调整

* Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 修复设备切换后重新对话,又切换回默认设备的问题

* Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 更新localvideo 尺寸

* Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 不能默认default

* 修改音频权限问题

* 更新打包结果

* fix: 对话按钮状态跟gs资源挂钩,删除无用代码

* fix: merge

* feat: gs渲染模块从npm包引入

* fix

* 新增对话记录

* Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 样式修改

* 更新包

* fix: gs数字人初始化位置和静音

* 对话记录滚到底部

* 至少100%高度

* Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 略微上移文本框

* 开始连接时清空对话记录

* fix: update gs render npm

* Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 逻辑保证

* Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* feat: 音频初始化配置是否静音

* actionsbar在有字幕时调整位置

* Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 样式优化

* feat: 增加readme

* fix: 资源图片

* fix: docs

* fix: update gs render sdk

* fix: gs模式下画面位置计算

* fix: update readme

* 设备判断,太窄处理

* Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* 是否有权限和是否有设备分开

* feat: gs 下载和加载钩子函数分离

* Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19

* fix: update gs render sdk

* 替换

* dist

* 上传文件

* del
2025-04-16 19:09:04 +08:00

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FastRTC

FastRTC Logo
Static Badge Static Badge

The Real-Time Communication Library for Python.

Turn any python function into a real-time audio and video stream over WebRTC or WebSockets.

Installation

pip install fastrtc

to use built-in pause detection (see ReplyOnPause), and text to speech (see Text To Speech), install the vad and tts extras:

pip install "fastrtc[vad, tts]"

Key Features

  • 🗣️ Automatic Voice Detection and Turn Taking built-in, only worry about the logic for responding to the user.
  • 💻 Automatic UI - Use the .ui.launch() method to launch the webRTC-enabled built-in Gradio UI.
  • 🔌 Automatic WebRTC Support - Use the .mount(app) method to mount the stream on a FastAPI app and get a webRTC endpoint for your own frontend!
  • Websocket Support - Use the .mount(app) method to mount the stream on a FastAPI app and get a websocket endpoint for your own frontend!
  • 📞 Automatic Telephone Support - Use the fastphone() method of the stream to launch the application and get a free temporary phone number!
  • 🤖 Completely customizable backend - A Stream can easily be mounted on a FastAPI app so you can easily extend it to fit your production application. See the Talk To Claude demo for an example on how to serve a custom JS frontend.

Docs

https://fastrtc.org

Examples

See the Cookbook for examples of how to use the library.

🗣️👀 Gemini Audio Video Chat

Stream BOTH your webcam video and audio feeds to Google Gemini. You can also upload images to augment your conversation!

Demo | Code

🗣️ Google Gemini Real Time Voice API

Talk to Gemini in real time using Google's voice API.

Demo | Code

🗣️ OpenAI Real Time Voice API

Talk to ChatGPT in real time using OpenAI's voice API.

Demo | Code

🤖 Hello Computer

Say computer before asking your question!

Demo | Code

🤖 Llama Code Editor

Create and edit HTML pages with just your voice! Powered by SambaNova systems.

Demo | Code

🗣️ Talk to Claude

Use the Anthropic and Play.Ht APIs to have an audio conversation with Claude.

Demo | Code

🎵 Whisper Transcription

Have whisper transcribe your speech in real time!

Demo | Code

📷 Yolov10 Object Detection

Run the Yolov10 model on a user webcam stream in real time!

Demo | Code

🗣️ Kyutai Moshi

Kyutai's moshi is a novel speech-to-speech model for modeling human conversations.

Demo | Code

🗣️ Hello Llama: Stop Word Detection

A code editor built with Llama 3.3 70b that is triggered by the phrase "Hello Llama". Build a Siri-like coding assistant in 100 lines of code!

Demo | Code

Usage

This is an shortened version of the official usage guide.

  • .ui.launch(): Launch a built-in UI for easily testing and sharing your stream. Built with Gradio.
  • .fastphone(): Get a free temporary phone number to call into your stream. Hugging Face token required.
  • .mount(app): Mount the stream on a FastAPI app. Perfect for integrating with your already existing production system.

Quickstart

Echo Audio

from fastrtc import Stream, ReplyOnPause
import numpy as np

def echo(audio: tuple[int, np.ndarray]):
    # The function will be passed the audio until the user pauses
    # Implement any iterator that yields audio
    # See "LLM Voice Chat" for a more complete example
    yield audio

stream = Stream(
    handler=ReplyOnPause(echo),
    modality="audio", 
    mode="send-receive",
)

LLM Voice Chat

from fastrtc import (
    ReplyOnPause, AdditionalOutputs, Stream,
    audio_to_bytes, aggregate_bytes_to_16bit
)
import gradio as gr
from groq import Groq
import anthropic
from elevenlabs import ElevenLabs

groq_client = Groq()
claude_client = anthropic.Anthropic()
tts_client = ElevenLabs()


# See "Talk to Claude" in Cookbook for an example of how to keep 
# track of the chat history.
def response(
    audio: tuple[int, np.ndarray],
):
    prompt = groq_client.audio.transcriptions.create(
        file=("audio-file.mp3", audio_to_bytes(audio)),
        model="whisper-large-v3-turbo",
        response_format="verbose_json",
    ).text
    response = claude_client.messages.create(
        model="claude-3-5-haiku-20241022",
        max_tokens=512,
        messages=[{"role": "user", "content": prompt}],
    )
    response_text = " ".join(
        block.text
        for block in response.content
        if getattr(block, "type", None) == "text"
    )
    iterator = tts_client.text_to_speech.convert_as_stream(
        text=response_text,
        voice_id="JBFqnCBsd6RMkjVDRZzb",
        model_id="eleven_multilingual_v2",
        output_format="pcm_24000"
        
    )
    for chunk in aggregate_bytes_to_16bit(iterator):
        audio_array = np.frombuffer(chunk, dtype=np.int16).reshape(1, -1)
        yield (24000, audio_array)

stream = Stream(
    modality="audio",
    mode="send-receive",
    handler=ReplyOnPause(response),
)

Webcam Stream

from fastrtc import Stream
import numpy as np


def flip_vertically(image):
    return np.flip(image, axis=0)


stream = Stream(
    handler=flip_vertically,
    modality="video",
    mode="send-receive",
)

Object Detection

from fastrtc import Stream
import gradio as gr
import cv2
from huggingface_hub import hf_hub_download
from .inference import YOLOv10

model_file = hf_hub_download(
    repo_id="onnx-community/yolov10n", filename="onnx/model.onnx"
)

# git clone https://huggingface.co/spaces/fastrtc/object-detection
# for YOLOv10 implementation
model = YOLOv10(model_file)

def detection(image, conf_threshold=0.3):
    image = cv2.resize(image, (model.input_width, model.input_height))
    new_image = model.detect_objects(image, conf_threshold)
    return cv2.resize(new_image, (500, 500))

stream = Stream(
    handler=detection,
    modality="video", 
    mode="send-receive",
    additional_inputs=[
        gr.Slider(minimum=0, maximum=1, step=0.01, value=0.3)
    ]
)

Running the Stream

Run:

Gradio

stream.ui.launch()

Telephone (Audio Only)

```py
stream.fastphone()
```

FastAPI

app = FastAPI()
stream.mount(app)

# Optional: Add routes
@app.get("/")
async def _():
    return HTMLResponse(content=open("index.html").read())

# uvicorn app:app --host 0.0.0.0 --port 8000