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
10 KiB
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
Streamcan 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
Examples
See the Cookbook for examples of how to use the library.
🗣️👀 Gemini Audio Video ChatStream BOTH your webcam video and audio feeds to Google Gemini. You can also upload images to augment your conversation! |
🗣️ Google Gemini Real Time Voice APITalk to Gemini in real time using Google's voice API. |
🗣️ OpenAI Real Time Voice APITalk to ChatGPT in real time using OpenAI's voice API. |
🤖 Hello ComputerSay computer before asking your question! |
🤖 Llama Code EditorCreate and edit HTML pages with just your voice! Powered by SambaNova systems. |
🗣️ Talk to ClaudeUse the Anthropic and Play.Ht APIs to have an audio conversation with Claude. |
🎵 Whisper TranscriptionHave whisper transcribe your speech in real time! |
📷 Yolov10 Object DetectionRun the Yolov10 model on a user webcam stream in real time! |
🗣️ Kyutai MoshiKyutai's moshi is a novel speech-to-speech model for modeling human conversations. |
🗣️ Hello Llama: Stop Word DetectionA 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! |
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
