[Jackaudio] About Jack Capture and Playback Latency

[Ruhul Amin]

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.

*So, how can I make my application sample accurate or what is the correct
procedure?*

It will be a great help and advance thanks.

Best Regards
Ruhul Amin
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