[FFmpeg-devel] Adding libspeexdsp's AEC to FFmpeg.

Arseniy Skvortsov ettavolt at gmail.com
Wed Sep 5 17:04:29 EEST 2018

Paul B Mahol <onemda at gmail.com> писал(а) в своём письме Tue, 04 Sep 2018  
10:45:22 +0300:

> On 9/4/18, Arseniy Skvortsov <ettavolt at 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++  

>> Target use case: two Android smartphones recording audio, transmitting  
>> it
>> with RTP to some processor, which cancels echo, adds a delay to sync  
>> with
>> another 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.
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