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libtgvoip/webrtc_dsp/absl
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Grishka 5caaaafa42 Updated WebRTC APM 
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.
2018-11-23 04:02:53 +03:00
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algorithm Updated WebRTC APM 2018-11-23 04:02:53 +03:00
base Updated WebRTC APM 2018-11-23 04:02:53 +03:00
container Updated WebRTC APM 2018-11-23 04:02:53 +03:00
memory Updated WebRTC APM 2018-11-23 04:02:53 +03:00
meta Updated WebRTC APM 2018-11-23 04:02:53 +03:00
strings Updated WebRTC APM 2018-11-23 04:02:53 +03:00
types Updated WebRTC APM 2018-11-23 04:02:53 +03:00
utility Updated WebRTC APM 2018-11-23 04:02:53 +03:00