mirror of
https://github.com/HumanAIGC-Engineering/gradio-webrtc.git
synced 2026-02-05 18:09:23 +08:00
[feat] update some feature
sync code of fastrtc, add text support through datachannel, fix safari connect problem support chat without camera or mic
This commit is contained in:
63
demo/app.py
63
demo/app.py
@@ -1,10 +1,17 @@
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import asyncio
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import base64
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from io import BytesIO
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import json
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import math
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import queue
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import time
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import uuid
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import threading
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from fastrtc.utils import Message
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import gradio as gr
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import numpy as np
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from gradio_webrtc import (
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from fastrtc import (
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AsyncAudioVideoStreamHandler,
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WebRTC,
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VideoEmitType,
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@@ -26,6 +33,7 @@ def encode_image(data: np.ndarray) -> dict:
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base64_str = str(base64.b64encode(bytes_data), "utf-8")
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return {"mime_type": "image/jpeg", "data": base64_str}
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frame_queue = queue.Queue(maxsize=100)
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class VideoChatHandler(AsyncAudioVideoStreamHandler):
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def __init__(
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@@ -38,7 +46,7 @@ class VideoChatHandler(AsyncAudioVideoStreamHandler):
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input_sample_rate=24000,
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)
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self.audio_queue = asyncio.Queue()
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self.video_queue = asyncio.Queue()
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self.video_queue = frame_queue
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self.quit = asyncio.Event()
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self.session = None
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self.last_frame_time = 0
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@@ -50,6 +58,25 @@ class VideoChatHandler(AsyncAudioVideoStreamHandler):
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output_frame_size=self.output_frame_size,
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)
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chat_id = ''
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async def on_chat_datachannel(self,message: Message,channel):
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# 返回
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# {"type":"chat",id:"标识属于同一段话", "message":"Hello, world!"}
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# {"type":"avatar_end"} 表示本次对话结束
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if message['type'] == 'stop_chat':
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self.chat_id = ''
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channel.send(json.dumps({'type':'avatar_end'}))
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else:
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id = uuid.uuid4().hex
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self.chat_id = id
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data = message["data"]
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halfLen = math.floor(data.__len__()/2)
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channel.send(json.dumps({"type":"chat","id":id,"message":data[:halfLen]}))
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await asyncio.sleep(5)
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if self.chat_id == id:
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channel.send(json.dumps({"type":"chat","id":id,"message":data[halfLen:]}))
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channel.send(json.dumps({'type':'avatar_end'}))
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async def video_receive(self, frame: np.ndarray):
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# if self.session:
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# # send image every 1 second
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@@ -61,10 +88,11 @@ class VideoChatHandler(AsyncAudioVideoStreamHandler):
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# print(frame.shape)
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newFrame = np.array(frame)
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newFrame[0:, :, 0] = 255 - newFrame[0:, :, 0]
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self.video_queue.put_nowait(newFrame)
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# self.video_queue.put_nowait(newFrame)
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async def video_emit(self) -> VideoEmitType:
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return await self.video_queue.get()
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# print('123123',frame_queue.qsize())
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return frame_queue.get()
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async def receive(self, frame: tuple[int, np.ndarray]) -> None:
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frame_size, array = frame
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@@ -114,14 +142,35 @@ with gr.Blocks(css=css) as demo:
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},
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}
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)
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handler = VideoChatHandler()
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webrtc.stream(
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VideoChatHandler(),
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handler,
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inputs=[webrtc],
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outputs=[webrtc],
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time_limit=150,
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time_limit=1500,
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concurrency_limit=2,
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)
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# 线程函数:随机生成 numpy 帧
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def generate_frames(width=480, height=960, channels=3):
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while True:
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try:
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# 随机生成一个 RGB 图像帧
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frame = np.random.randint(188, 256, (height, width, channels), dtype=np.uint8)
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# 将帧放入队列
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frame_queue.put(frame)
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# print("生成一帧数据,形状:", frame.shape, frame_queue.qsize())
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# 模拟实时性:避免过度消耗 CPU
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time.sleep(0.03) # 每秒约生成 30 帧
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except Exception as e:
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print(f"生成帧时出错: {e}")
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break
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thread = threading.Thread(target=generate_frames, daemon=True)
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thread.start()
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if __name__ == "__main__":
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demo.launch()
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