Files
gradio-webrtc/demo/video_chat/app.py
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

199 lines
6.3 KiB
Python

import asyncio
import base64
from io import BytesIO
import json
import math
import queue
import time
from typing import TypedDict
import uuid
import threading
# from fastrtc.utils import Message
from fastapi import FastAPI, WebSocket, WebSocketDisconnect
import gradio as gr
import numpy as np
from fastrtc import (
AsyncAudioVideoStreamHandler,
WebRTC,
VideoEmitType,
AudioEmitType,
)
from PIL import Image
import uvicorn
class Message(TypedDict):
type: str
data: any
def encode_audio(data: np.ndarray) -> dict:
"""Encode Audio data to send to the server"""
return {"mime_type": "audio/pcm", "data": base64.b64encode(data.tobytes()).decode("UTF-8")}
def encode_image(data: np.ndarray) -> dict:
with BytesIO() as output_bytes:
pil_image = Image.fromarray(data)
pil_image.save(output_bytes, "JPEG")
bytes_data = output_bytes.getvalue()
base64_str = str(base64.b64encode(bytes_data), "utf-8")
return {"mime_type": "image/jpeg", "data": base64_str}
frame_queue = queue.Queue(maxsize=100)
class VideoChatHandler(AsyncAudioVideoStreamHandler):
def __init__(
self, expected_layout="mono", output_sample_rate=24000, output_frame_size=480
) -> None:
super().__init__(
expected_layout,
output_sample_rate,
output_frame_size,
input_sample_rate=24000,
)
self.audio_queue = asyncio.Queue()
self.video_queue = frame_queue
self.quit = asyncio.Event()
self.session = None
self.last_frame_time = 0
def copy(self, **kwargs) -> "VideoChatHandler":
return VideoChatHandler(
expected_layout=self.expected_layout,
output_sample_rate=self.output_sample_rate,
output_frame_size=self.output_frame_size,
)
def on_pc_connected(self, webrtc_id):
print(webrtc_id)
chat_id = ''
async def on_chat_datachannel(self,message: Message,channel):
# 返回
# {"type":"chat",id:"标识属于同一段话", "message":"Hello, world!"}
# {"type":"avatar_end"} 表示本次对话结束
if message['type'] == 'stop_chat':
self.chat_id = ''
channel.send(json.dumps({'type':'avatar_end'}))
else:
id = uuid.uuid4().hex
self.chat_id = id
data = message["data"]
halfLen = math.floor(data.__len__()/2)
channel.send(json.dumps({"type":"chat","id":id,"message":data[:halfLen]}))
await asyncio.sleep(5)
if self.chat_id == id:
channel.send(json.dumps({"type":"chat","id":id,"message":data[halfLen:]}))
channel.send(json.dumps({'type':'avatar_end'}))
async def video_receive(self, frame: np.ndarray):
# if self.session:
# # send image every 1 second
# if time.time() - self.last_frame_time > 1:
# self.last_frame_time = time.time()
# await self.session.send(encode_image(frame))
# if self.latest_args[2] is not None:
# await self.session.send(encode_image(self.latest_args[2]))
# print(frame.shape)
newFrame = np.array(frame)
newFrame[0:, :, 0] = 255 - newFrame[0:, :, 0]
# self.video_queue.put_nowait(newFrame)
async def video_emit(self) -> VideoEmitType:
# print('123123',frame_queue.qsize())
return frame_queue.get()
async def receive(self, frame: tuple[int, np.ndarray]) -> None:
frame_size, array = frame
self.audio_queue.put_nowait(array)
async def emit(self) -> AudioEmitType:
if not self.args_set.is_set():
await self.wait_for_args()
array = await self.audio_queue.get()
return (self.output_sample_rate, array)
def shutdown(self) -> None:
self.quit.set()
self.connection = None
self.args_set.clear()
self.quit.clear()
css = """
footer {
display: none !important;
}
"""
with gr.Blocks(css=css) as demo:
webrtc = WebRTC(
label="Video Chat",
modality="audio-video",
mode="send-receive",
video_chat=True,
avatar_type="",
avatar_assets_path="",
avatar_ws_route='/ws',
elem_id="video-source",
track_constraints={
"video": {
"facingMode": "user",
"width": {"ideal": 500},
"height": {"ideal": 500},
"frameRate": {"ideal": 30},
},
"audio": {
"echoCancellation": True,
"noiseSuppression": {"exact": True},
"autoGainControl": {"exact": False},
"sampleRate": {"ideal": 24000},
"sampleSize": {"ideal": 16},
"channelCount": {"exact": 1},
},
}
)
handler = VideoChatHandler()
webrtc.stream(
handler,
inputs=[webrtc],
outputs=[webrtc],
time_limit=1500,
concurrency_limit=2,
)
# 线程函数:随机生成 numpy 帧
def generate_frames(width=480, height=960, channels=3):
while True:
try:
# 随机生成一个 RGB 图像帧
frame = np.random.randint(188, 256, (height, width, channels), dtype=np.uint8)
# 将帧放入队列
frame_queue.put(frame)
# print("生成一帧数据,形状:", frame.shape, frame_queue.qsize())
# 模拟实时性:避免过度消耗 CPU
time.sleep(0.03) # 每秒约生成 30 帧
except Exception as e:
print(f"生成帧时出错: {e}")
break
thread = threading.Thread(target=generate_frames, daemon=True)
thread.start()
@demo.app.websocket("/ws")
async def on_websocket(websocket: WebSocket):
await websocket.accept()
while True:
try:
data = await websocket.receive_text()
await websocket.send_text(f"Message text was: {data}")
except WebSocketDisconnect:
break
if __name__ == "__main__":
demo.launch()