Python bindings of webrtc audio processing
Project description
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.
-
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
-
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()))
Project details
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