<div dir="ltr">Hi, Thank you for your great support in Jack. I am trying to build a real-time application on basis of Jack Server and Client architecture. But Jack capture and playback latency is failing me to make the application sample accurate. My Jack server configuration (jackd -dalsa): JACK server starting in realtime mode with priority 10 self-connect-mode is "Don't restrict self connect requests" audio_reservation_init Acquire audio card Audio0 creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: use 2 periods for capture ALSA: final selected sample format for playback: 32bit integer little-endian ALSA: use 2 periods for playback My Understanding : 0. Realtime mode = Yes, 10 priority 1. Sampling Rate = 48000Hz 2. 1 Periods = 1024 frames 3. Buffer Size = 2 Periods = 2048 frames 4. Capture Latency = 2 periods = 1 Buffer = 21.3 x2 ms 5. Playback Latency = 2 periods = 1 Buffer = 21.3 x2 ms Apllication: passthru/simple_client.c of Jack2 repositry with one client for both sterio capture and mono playback My application pipeline: input -----------> capture left channel (stage 1) ------------> playback left channel (stage 2) --------------> capture right channel (stage 3) Latency calculation: After stage 1 --------------- 2 periods = 42.6 ms After stage 2 --------------- 2 + 2 periods = 85.2 ms After stage 3 --------------- 2 + 2 + 2 periods = 127.8 ms Difference between stage 3 and stage 1 6 periods - 2 periods = 4 periods Latency difference should be 4 periods but I am getting larger than 4 periods like 4 periods which also differs with different trials. <div>So, how can I make my application sample accurate or what is the correct procedure?</div><div> </div>It will be a great help and advance thanks. Best Regards Ruhul Amin </div>