Update README.md (#2)

* Update README.md

README.md

@@ -5,15 +5,17 @@
 <a href="https://github.com/freddyaboulton/gradio-webrtc" target="_blank"><img alt="Static Badge" style="display: block; padding-right: 5px; height: 20px;" src="https://img.shields.io/badge/github-white?logo=github&logoColor=black"></a>
 <a href="https://freddyaboulton.github.io/gradio-webrtc/" target="_blank"><img alt="Static Badge" src="https://img.shields.io/badge/Docs-ffcf40"></a>
 </div>
+<div align="center">
+<strong>中文|<a href="./README_en.md">English</a></strong>
+</div>
 
-<h3 style='text-align: center'>
+本仓库是从原有的 gradio_webrtc 仓库 fork 而来，主要增加了`video_chat`作为允许的入参，并默认开启，这个模式和原有的`modality="audio-video"`且`mode="send-receive"`的行为保持一致，但重写了 UI 部分，增加了更多的交互能力(更多的麦克风操作，同时展示本地视频信息），其视觉表现如下图。
-Stream video and audio in real time with Gradio using WebRTC.
+
+如果手动将`video_chat`参数设置为`False`，则其用法与原仓库保持一致 https://freddyaboulton.github.io/gradio-webrtc/
 
 ![picture-in-picture](docs/image.png)
 ![side-by-side](docs/image2.png)
 
-</h3>
-
 ## Installation
 
 ```bash
@@ -31,6 +33,8 @@ https://freddyaboulton.github.io/gradio-webrtc/
 
 ## Examples
 
+使用时需要一个 handler 作为组件的入参，并实现类似以下代码：
+
 ```python
 import asyncio
 import base64
@@ -84,18 +88,22 @@ class VideoChatHandler(AsyncAudioVideoStreamHandler):
             output_frame_size=self.output_frame_size,
         )
 
+    #处理客户端上传的视频数据
     async def video_receive(self, frame: np.ndarray):
         newFrame = np.array(frame)
         newFrame[0:, :, 0] = 255 - newFrame[0:, :, 0]
         self.video_queue.put_nowait(newFrame)
 
+    #准备服务端下发的视频数据
     async def video_emit(self) -> VideoEmitType:
         return await self.video_queue.get()
 
+    #处理客户端上传的音频数据
     async def receive(self, frame: tuple[int, np.ndarray]) -> None:
         frame_size, array = frame
         self.audio_queue.put_nowait(array)
 
+    #准备服务端下发的音频数据
     async def emit(self) -> AudioEmitType:
         if not self.args_set.is_set():
             await self.wait_for_args()
@@ -140,8 +148,8 @@ if __name__ == "__main__":
 
 ## Deployment
 
-When deploying in a cloud environment (like Hugging Face Spaces, EC2, etc), you need to set up a TURN server to relay the WebRTC traffic.
+在云环境中部署（例如 huggingface，EC2 等）时，您需要设置转向服务器以中继 WEBRTC 流量。
-The easiest way to do this is to use a service like Twilio.
+最简单的方法是使用 Twilio 之类的服务。国内部署需要寻找适合的替代方案。
 
 ```python
 from twilio.rest import Client

README_en.md

@@ -0,0 +1,182 @@
+<h1 style='text-align: center; margin-bottom: 1rem'> Gradio WebRTC ⚡️ </h1>
+
+<div style="display: flex; flex-direction: row; justify-content: center">
+<img style="display: block; padding-right: 5px; height: 20px;" alt="Static Badge" src="https://img.shields.io/pypi/v/gradio_webrtc"> 
+<a href="https://github.com/freddyaboulton/gradio-webrtc" target="_blank"><img alt="Static Badge" style="display: block; padding-right: 5px; height: 20px;" src="https://img.shields.io/badge/github-white?logo=github&logoColor=black"></a>
+<a href="https://freddyaboulton.github.io/gradio-webrtc/" target="_blank"><img alt="Static Badge" src="https://img.shields.io/badge/Docs-ffcf40"></a>
+</div>
+<div align="center">
+<strong><a href="./README.md">中文</a>|English</strong>
+</div>
+This repository is forked from the original gradio_webrtc repository, primarily adding `video_chat` as an allowed parameter to be enabled by default. This mode is consistent with the behavior of the original `modality="audio-video"` and `mode="send-receive"`, but the UI has been rewritten to include more interactive capabilities (more microphone controls, and the ability to display local video information). The visual presentation is shown below.
+
+If `video_chat` is manually set to `False`, its usage is consistent with the original repository https://freddyaboulton.github.io/gradio-webrtc/
+
+![picture-in-picture](docs/image.png)
+![side-by-side](docs/image2.png)
+
+## Installation
+
+```bash
+gradio cc install
+gradio cc build --no-generate-docs
+```
+
+```bash
+pip install dist/gradio_webrtc-0.0.30.dev0-py3-none-any.whl
+```
+
+## Docs
+
+https://freddyaboulton.github.io/gradio-webrtc/
+
+## Examples
+
+When using it, you need a handler as the entry parameter of the component and implement code similar to the following:
+
+```python
+import asyncio
+import base64
+from io import BytesIO
+
+import gradio as gr
+import numpy as np
+from gradio_webrtc import (
+    AsyncAudioVideoStreamHandler,
+    WebRTC,
+    VideoEmitType,
+    AudioEmitType,
+)
+from PIL import Image
+
+
+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}
+
+
+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 = asyncio.Queue()
+        self.quit = asyncio.Event()
+        self.session = None
+        self.last_frame_time = 0
+
+    def copy(self) -> "VideoChatHandler":
+        return VideoChatHandler(
+            expected_layout=self.expected_layout,
+            output_sample_rate=self.output_sample_rate,
+            output_frame_size=self.output_frame_size,
+        )
+
+    #Process video data uploaded by the client
+    async def video_receive(self, frame: np.ndarray):
+        newFrame = np.array(frame)
+        newFrame[0:, :, 0] = 255 - newFrame[0:, :, 0]
+        self.video_queue.put_nowait(newFrame)
+
+    #Prepare the video data sent by the server
+    async def video_emit(self) -> VideoEmitType:
+        return await self.video_queue.get()
+
+    #Process audio data uploaded by the client
+    async def receive(self, frame: tuple[int, np.ndarray]) -> None:
+        frame_size, array = frame
+        self.audio_queue.put_nowait(array)
+
+    #Prepare the audio data sent by the server
+    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,
+            elem_id="video-source",
+        )
+        webrtc.stream(
+            VideoChatHandler(),
+            inputs=[webrtc],
+            outputs=[webrtc],
+            time_limit=150,
+            concurrency_limit=2,
+        )
+
+
+if __name__ == "__main__":
+    demo.launch()
+
+```
+
+## Deployment
+
+When deploying in a cloud environment (like Hugging Face Spaces, EC2, etc), you need to set up a TURN server to relay the WebRTC traffic.
+The easiest way to do this is to use a service like Twilio.
+
+```python
+from twilio.rest import Client
+import os
+
+account_sid = os.environ.get("TWILIO_ACCOUNT_SID")
+auth_token = os.environ.get("TWILIO_AUTH_TOKEN")
+
+client = Client(account_sid, auth_token)
+
+token = client.tokens.create()
+
+rtc_configuration = {
+    "iceServers": token.ice_servers,
+    "iceTransportPolicy": "relay",
+}
+
+with gr.Blocks() as demo:
+    ...
+    rtc = WebRTC(rtc_configuration=rtc_configuration, ...)
+    ...
+```
+
+## Contributors
+
+[csxh47](https://github.com/xhup)
+[bingochaos](https://github.com/bingochaos)
+[sudowind](https://github.com/sudowind)
+[emililykimura](https://github.com/emililykimura)
+[Tony](https://github.com/raidios)
+[Cheng Gang](https://github.com/lovepope)