WebRTC Audio Processing for Python

Python binding of WebRTC Audio Processing.

Requirements

• swig
• compile toolchain
• python

Build

There are two way to build the package.

1. using setup.py

   git clone https://github.com/xiongyihui/python-webrtc-audio-processing.git
   cd python-webrtc-audio-processing
   git submodule init && git submodule update
   python setup.py build
   sudo python setup.py install
   

2. using Makefile

   git clone https://github.com/xiongyihui/python-webrtc-audio-processing.git
   cd python-webrtc-audio-processing
   git submodule init && git submodule update
   cd webrtc-audio-processing
   ./autogen.sh
   ./configure --with-pic
   make
   cd ../src
   make
   

Usage

from webrtc_audio_processing import AudioProcessingModule as AP

ap = AP(enable_vad=True, enable_ns=True)
ap.set_stream_format(16000, 1)      # set sample rate and channels
ap.set_ns_level(1)                  # NS level from 0 to 3
ap.set_vad_level(1)                 # VAD level from 0 to 3

audio_10ms = '\0' * 160 * 2         # 10ms, 16000 sample rate, 16 bits, 1 channel

# only support processing 10ms audio data each time
audio_out = ap.process_stream(audio_10ms)
print('voice: {}'.format(ap.has_voice()))