Hi guys, 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 latency: $ 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 Any ideas? _______________________________________________ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
