mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-02 17:51:06 +01:00
5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way. |
||
---|---|---|
.. | ||
AudioInputAndroid.cpp | ||
AudioInputAndroid.h | ||
AudioInputOpenSLES.cpp | ||
AudioInputOpenSLES.h | ||
AudioOutputAndroid.cpp | ||
AudioOutputAndroid.h | ||
AudioOutputOpenSLES.cpp | ||
AudioOutputOpenSLES.h | ||
JNIUtilities.h | ||
OpenSLEngineWrapper.cpp | ||
OpenSLEngineWrapper.h | ||
VideoRendererAndroid.cpp | ||
VideoRendererAndroid.h | ||
VideoSourceAndroid.cpp | ||
VideoSourceAndroid.h |