On 9/4/18, Arseniy Skvortsov <ettav...@gmail.com> wrote:I'm trying to add AEC from libspeexdsp to libavfilter.Try with input as files, not streams.
Tried this way: play some file while recording from mic. Then pass recording and that file to AEC. Well, it works, but far from what I'd like to have. WebRTC's AEC seems to achieve much better cancellation. So, the question is: how to make ffmpeg's build system compile C/C++ wrapper?
Target use case: two Android smartphones recording audio, transmitting it with RTP to some processor, which cancels echo, adds a delay to sync withanother video stream, mixes with music and outputs to a stereo system.
Just realized this is more like a karaoke system then what's usually needed for VoIP. ☺ Also, I got Speex'es AEC to do something useful int this setup. Just had to realize unitless delay is seconds, not µs.
Attached current state of the work for history purpose.
0001-AEC-filter-from-Speex.patch
Description: Binary data
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