[FFmpeg-user] High audio latency (although low with ffplay!)
arifd86 at gmail.com
Wed Sep 8 12:44:01 EEST 2021
I want to have ffmpeg grab my microphone at a high sample rate, apply some
processing, downsample and feed it to a virtual PulseAudio device to then
be able to use live in any application.
But even in the most vanilla setting, it is adding about 4 seconds of
$ ffmpeg -f alsa -acodec pcm_s32le -i hw:1,0 -f pulse out
Doing this, say in Audacity, introduces no noticeable latency.
Let's try from pulse to pulse:
$ ffmpeg -f pulse -acodec pcm_s16le -i default -f pulse out
Again, 4 seconds latency.
Interestingly, if I use ffplay, there is no noticeable latency!
$ ffplay -f pulse -acodec pcm_s16le -i default
Taking the input from ffmpeg and piping it to ffplay...
$ ffmpeg -f pulse -acodec pcm_s16le -i default -f wav - | ffplay -f wav -
about 3 seconds latency.
I have also experimented with these flags: -thread_queue_size, -fflags
nobuffer, -flags low_delay ,-strict experimental, -re, -deadline realtime
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