On 9/4/18, Arseniy Skvortsov <ettav...@gmail.com> wrote: > Hello, > > I'm trying to add AEC from libspeexdsp to libavfilter. > 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. > > First line of the log file attached shows the commandline I use for > testing. What the filtergraph means: grab mono from USB webcam, pass it as > > 'record' to AEC, loopback 'cleaned' output with a frame delay as > 'playback' for AEC, build a stereo output from a 'cleaned' output and a > channel of silence. > OFFTOPIC: Had to use ALSA directly because pulse demuxer has some weird > delay problem (https://bbs.archlinux.org/viewtopic.php?id=239893). > > So far AEC already does something useful because positive > self-amplification (or what's the term for this?) starts on a [slightly] > larger volume then without this filter. > Still, it doesn't remove the effect completely. I suspect, that it may be > related to warnings > [alsa @ 0x56554d388bc0] ALSA buffer xrun > and FF_FILTER_FORWARD_STATUS_BACK_ALL dropping all queued frames when > output isn't ready. How do you think, is it true? > > I've also explored possibility of adding webrtc-audio-processing's AEC, > but for that I'd need to build a C/C++ wrapper. Of which I haven't found > any example in FFmpeg's code. People out there say that it's better than > libspeexdsp's. Plus Xiph seems to be very insisting on 'one-quartz > recording & playback', while my use case is definitely nothing like this. > What do you think about this? > > Any suggestions are welcome.
Try with input as files, not streams. _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
