xiongyihui / python-webrtc-audio-processing Goto Github PK

Traceback (most recent call last):
File "wav_ns.py", line 36, in
data_out = ap.process_stream(data)
File "/home/amber/Desktop/env/.env/lib/python3.5/site-packages/webrtc_audio_processing/webrtc_audio_processing.py", line 82, in process_stream
def process_stream(self, *args): return _webrtc_audio_processing.AudioProcessingModule_process_stream(self, *args)
TypeError: in method 'AudioProcessingModule_process_stream', argument 2 of type 'std::string const &'

I try to modify the echo cancellation script follows your aec script.
it failed and out empty audio.

TypeError: in method 'AudioProcessingModule_process_stream', argument 2 of type 'std::string const &'

git submodule init && git submodule update
python-webrtc-audio-processing>python setup.py build
Traceback (most recent call last):
File "setup.py", line 33, in
ap_sources.remove('webrtc-audio-processing/webrtc/system_wrappers/source/rw_lock_generic.cc')
ValueError: list.remove(x): x not in list

after build the package, I run the code from Useage,then got a TypError:

TypeError: in method 'AudioProcessingModule_process_stream', argument 2 of type 'std::string const &'

Anyone can help me？

Mac is not supported????

谢谢

/configure: line 11730: syntax error near unexpected token GNUSTL,' ./configure: line 11730: PKG_CHECK_MODULES(GNUSTL, gnustl)'
I encounter such errors in ./configure, thanks for any help.

hardware:raspberry pi zero w
OS:raspbian
python: 2.7
Installation method : read me - build - 1.using setup.py
I install python webrtc audio processing in raspberrypi zero according to the instructions, and then test the wav_ns.py "python wav_ns.py audio.wav out.wav, but report an error: "illegal instruction",how to solve it?thanks

git submodule init && git submodule update
python setup.py build
creating build\temp.win-amd64-3.7\Release\src
C:\Program Files (x86)\Microsoft Visual Studio\2019\Enterprise\VC\Tools\MSVC\14.25.28610\bin\HostX86\x64\cl.exe /c /nologo /Ox /W3 /GL /DNDEBUG /MD -DWEBRTC_LINUX -DWEBRTC_POSIX -DWEBRTC_NS_FLOAT -DWEBRTC_AUDIO_PROCESSING_ONLY_BUILD -Isrc -Iwebrtc-audio-processing -ID:\InstallPath\Develop\Anaconda3\5.3.1\envs\AudioCompare02\include -ID:\InstallPath\Develop\Anaconda3\5.3.1\envs\AudioCompare02\include "-IC:\Program Files (x86)\Microsoft Visual Studio\2019\Enterprise\VC\Tools\MSVC\14.25.28610\ATLMFC\include" "-IC:\Program Files (x86)\Microsoft Visual Studio\2019\Enterprise\VC\Tools\MSVC\14.25.28610\include" "-IC:\Program Files (x86)\Windows Kits\NETFXSDK\4.8\include\um" "-IC:\Program Files (x86)\Windows Kits\10\include\10.0.18362.0\ucrt" "-IC:\Program Files (x86)\Windows Kits\10\include\10.0.18362.0\shared" "-IC:\Program Files (x86)\Windows Kits\10\include\10.0.18362.0\um" "-IC:\Program Files (x86)\Windows Kits\10\include\10.0.18362.0\winrt" "-IC:\Program Files (x86)\Windows Kits\10\include\10.0.18362.0\cppwinrt" /EHsc /Tpwebrtc-audio-processing/webrtc\common_types.cc /Fobuild\temp.win-amd64-3.7\Release\webrtc-audio-processing/webrtc\common_types.obj -std=c++11
cl: 命令行 warning D9002 :忽略未知选项“-std=c++11”
common_types.cc
webrtc-audio-processing\webrtc/common_types.h(299): error C2039: "strcasecmp": 不是 "global namespace'" 的成员 webrtc-audio-processing\webrtc/common_types.h(299): error C3861: “strcasecmp”: 找不到标识符 webrtc-audio-processing\webrtc/common_types.h(723): error C2039: "strcasecmp": 不是 "global namespace'" 的成员
webrtc-audio-processing\webrtc/common_types.h(723): error C3861: “strcasecmp”: 找不到标识符
error: command 'C:\Program Files (x86)\Microsoft Visual Studio\2019\Enterprise\VC\Tools\MSVC\14.25.28610\bin\HostX86\x64\cl.exe' failed with exit status 2

(AudioCompare02) D:\Projects\SpeechCompare\SilenceDetector\python-webrtc-audio-processing>

We have a question on the limitation on the framesize to 10ms. We need to process audio stream with smaller size. Is the limitation from the WebRTC audio processing or Python wrapper? Is there a way to walk around this limitation?

In addition, when doing noise suppression on short audio stream buffer (e.g. 50ms) of a long audio, the denoised original audio have beat-like noise at the boundaries of the audio stream buffer. What causes the noise?