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Move readme from main
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Improved styling for readme
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Merge pull request #1 from Pterosaur/v-zhonxia/updateReadme
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219 changes: 191 additions & 28 deletions README.md
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# AlphaCC
# AlphaRTC

AlphaCC is the ML-based bandwidth estimation on WebRTC.
<table>
<tr>
<td align="center">master</td>
<td align="center">
<a href="https://dev.azure.com/OpenNetLab/ONL-github/_build/latest?definitionId=3&branchName=main">
<img src="https://dev.azure.com/OpenNetLab/ONL-github/_apis/build/status/OpenNetLab.AlphaRTC?branchName=main">
</a>
</td>
</tr>
<tr>
<td align="center">dev</td>
<td align="center">
<a href="https://dev.azure.com/OpenNetLab/ONL-github/_build/latest?definitionId=3&branchName=dev">
<img src="https://dev.azure.com/OpenNetLab/ONL-github/_apis/build/status/OpenNetLab.AlphaRTC?branchName=dev">
</a>
</td>
</tr>
<tr>
<td align="center">issues</td>
<td align="center">
<a href="https://github.com/OpenNetLab/AlphaRTC/issues">
<img src="https://img.shields.io/github/issues-raw/OpenNetLab/AlphaRTC">
</a>
</td>
</tr>
</table>

## WebRTC
## Motivation

AlphaRTC is a fork of Google's WebRTC project using ML-based bandwidth estimation, delivered by the OpenNetLab team. By equipping WebRTC with a more accurate bandwidth estimator, our mission is to eventually increase the quality of transmission.

AlphaRTC replaces Google Congestion Control (GCC) with ONNXInfer, an ML-powered bandwidth estimator, which takes in an ONNX model to make bandwidth estimation more accurate. ONNXInfer is proudly powered by Microsoft's [ONNXRuntime](https://github.com/microsoft/onnxruntime).

## Environment

Ubuntu 18.04 is the only officially supported distro at this moment. For other distros, you may be able to compile your own binary, or use our pre-provided Docker images.

## Compilation

To compile AlphaRTC, please refer to the following steps

1. Prerequisites

Make sure Docker is installed on your system.

``` shell
# Install Docker
curl -fsSL get.docker.com -o get-docker.sh
sudo sh get-docker.sh
```

2. Clone the code

``` shell
git clone https://github.com/OpenNetLab/AlphaRTC.git
```

3. Build Docker images

``` shell
cd AlphaRTC
sudo make all
```

You should then be able to see two Docker images, `alphartc` and `alphartc-compile` using `sudo docker images`

## Demo

AlphaRTC consists of many different components. `peerconnection_serverless` is an application for demo purposes that comes with AlphaRTC. It establishes RTC communication with another peer without the need of a server.

**WebRTC is a free, open software project** that provides browsers and mobile
applications with Real-Time Communications (RTC) capabilities via simple APIs.
The WebRTC components have been optimized to best serve this purpose.
In order to run the application, you will need a configuration file in json format. The details are explained in the next chapter.

**Our mission:** To enable rich, high-quality RTC applications to be
developed for the browser, mobile platforms, and IoT devices, and allow them
all to communicate via a common set of protocols.
In addition to the config file, you will also need other files, such as video/audio source files and an ONNX model.

The WebRTC initiative is a project supported by Google, Mozilla and Opera,
amongst others.
To run an AlphaRTC instance, put the config files in a directory, e.g., `config_files`, then mount it to an endpoint inside `alphartc` container

### Basic Information
``` shell
sudo docker run -v config_files:/app/config_files alphartc peerconnection_serverless /app/config_files/config.json
```

* Commit ID: @656590dabce079db46a7c6676a55cd81268ea786
* Repository: <https://webrtc.googlesource.com/src>
* Branch: master
Since `peerconnection_serverless` needs two peers, you may spawn two instances (a receiver and a sender) in the same network and make them talk to each other. For more information on Docker networking, check [Docker Networking](https://docs.docker.com/network/network-tutorial-standalone/)

### Development
### Configurations for *peerconnection_serverless*

See http://www.webrtc.org/native-code/development for instructions on how to get
started developing with the native code.
This section describes required fields for the json configuration file.

[Authoritative list](native-api.md) of directories that contain the
native API header files.
- **serverless_connection**
- **sender**
- **enabled**: If set to `true`, the client will act as sender and automatically connect to receiver when launched
- **send_to_ip**: The IP of serverless peerconnection receiver
- **send_to_port**: The port of serverless peerconnection receiver
- **receiver**
- **enabled**: If set to `true`, the client will act as receiver and wait for sender to connect.
- **listening_ip**: The IP address that the socket in receiver binds and listends to
- **listening_port**: The port number that the socket in receiver binds and listends to
- **autoclose**: The time *in seconds* before close automatically (always run if autoclose=0)

### More info
***Note: one and only one of `sender.enabled` and `receiver.enabled` has to be `true`. I.e., `sender.enabled` XOR `receiver.enabled`***

- **bwe_feedback_duration**: The duration the receiver sends its estimated target rate every time(*in millisecond*)

- **onnx**
- **onnx_model_path**: The path of the [onnx](https://www.onnxruntime.ai/) model

- **video_source**
- **video_disabled**:
- **enabled**: If set to `true`, the client will not take any video source as input
- **webcam**:
- **enabled**: __Windows-only__. If set to `true`, then the client will use the web camera as the video source. For Linux, please set to `false`
- **video_file**:
- **enabled**: If set to `true`, then the client will use a video file as the video source
- **height**: The height of the input video
- **width**: The width of the input video
- **fps**: The frames per second (FPS) of the input video
- **file_path**: The file path of the input video in [YUV](https://en.wikipedia.org/wiki/YUV) format

***Note: one and only one of `video_source.webcam.enabled` and `video_source.video_file.enabled` has to be `true`. I.e., `video_source.webcam.enabled` XOR `video_source.video_file.enabled`***

- **audio_source**
- **microphone**:
- **enabled**: Whether to enable microphone output or not
- **audio_file**:
- **enabled**: Whether to enable audio file input or not
- **file_path**: The file path of the input audio file in WAV format

- **save_to_file**
- **enabled**: Whether to enable file saving or not
- **audio**:
- **file_path**: The file path of the output audio file in WAV format
- **video**
- **width**: The width of the output video file
- **height**: The height of the output video file
- **fps**: Frames per second of the output video file
- **file_path**: The file path of the output video file in YUV format

#### Example

``` json
{
"serverless_connection": {
"sender": {
"enabled": false,
"dest_ip": "127.0.0.1",
"dest_port": 8888
}
"autoclose": 20
},

"bwe_feedback_duration": 200,

"onnx": {
"onnx_model_path": "onnx-model.onnx"
},

"video_source":{
"video_disabled": {
"enabled": true
},
"webcam": {
"enabled": false
},
"video_file": {
"enabled": true,
"height": 480,
"width": 640,
"fps": 24,
"file_path": "testmedia/test.yuv"
}
},

"audio_source": {
"microphone": {
"enabled": false
},
"audio_file": {
"enabled": true,
"file_path": "testmedia/test.wav"
}
},
"save_to_file": {
"enabled": true,
"audio": {
"file_path": "outaudio.wav"
},
"video": {
"width": 640,
"height": 480,
"fps": 24,
"file_path": "outvideo.yuv"
}
}
}
```

## Who Are We

The OpenNetLab team is an inter-academia research team, initiated by the Networking Reasearch Group at Microsoft Research Asia. Our team members are from different research institudes, including Peking University, Nanjing University and Nanyang Technological University.

## WebRTC

* Official web site: http://www.webrtc.org
* Master source code repo: https://webrtc.googlesource.com/src
* Samples and reference apps: https://github.com/webrtc
* Mailing list: http://groups.google.com/group/discuss-webrtc
* Continuous build: http://build.chromium.org/p/client.webrtc
* [Coding style guide](style-guide.md)
* [Code of conduct](CODE_OF_CONDUCT.md)
You can find the Readme of the original WebRTC project [here](./README.webrtc.md)
28 changes: 28 additions & 0 deletions README.webrtc.md
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**WebRTC is a free, open software project** that provides browsers and mobile
applications with Real-Time Communications (RTC) capabilities via simple APIs.
The WebRTC components have been optimized to best serve this purpose.

**Our mission:** To enable rich, high-quality RTC applications to be
developed for the browser, mobile platforms, and IoT devices, and allow them
all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera,
amongst others.

### Development

See http://www.webrtc.org/native-code/development for instructions on how to get
started developing with the native code.

[Authoritative list](native-api.md) of directories that contain the
native API header files.

### More info

* Official web site: http://www.webrtc.org
* Master source code repo: https://webrtc.googlesource.com/src
* Samples and reference apps: https://github.com/webrtc
* Mailing list: http://groups.google.com/group/discuss-webrtc
* Continuous build: http://build.chromium.org/p/client.webrtc
* [Coding style guide](style-guide.md)
* [Code of conduct](CODE_OF_CONDUCT.md)