The lowest possible input-to-output latency of an audio workstation is always two times the period size (samples/buffer) (plus a few microseconds of internal delays in the ADCs, controllers, ...)

The sound chips operate on integral periods or a fixed number of samples, and when you get a period worth of audio data from your ADC, the DAC will already have started putting out the first samples of the next period. Hence, you prepare your audio samples for the second-to-next period.

Assume you have audio processing code roughly looking like this:

    while (1) {
        poll(); /* some API function waiting for the "next period" */
        read(soundcard, block_of_samples);  /* or let DMA do it */
        process_samples(block_of_samples);
        write(soundcard, block_of_samples); /* or let DMA do it */
    }

Let's try some ASCII art:

      v- audio samples going into your soundcard
       _     _     _     _     _     _     _     _     _     _     _     _
      /1\   /2\   /3\   /4\   /5\   /6\   /7\   /8\   /9\   /0\   /1\   /2\ ...
         \_/   \_/   \_/v  \_/   \_/   \_/   \_/   \_/   \_/   \_/   \_/   \
                        |
      \________________/\________________/\________________/\________________/
       Period 1          Period 2          Period 3          Period 4
                        |
                        |
                        [*] here, Period 1 has been DMA'ed from the soundcard
                            to an mmapped buffer of your audio application
    
                          <~~~~~~~~~~~>  here processing of your audio takes place
                        
                                       [*] this is the latest point at which processing
                                         | must complete so that there will be data for
                                         | the soundcard to output. DMA will start
                                         | from the buffer to the soundcard DAC.
                                         |
       _     _     _     _     _     _   v _     _     _     _     _     _
      / \   / \   / \   / \   / \   / \   /1\   /2\   /3\   /4\   /5\   /6\ ...
         \_/   \_/   \_/   \_/   \_/   \_/   \_/   \_/   \_/   \_/   \_/   \
    
      \________________/\________________/\________________/\________________/
    
                                           ^- processed audio samples will come out
                                              of your laptop's speakers...

Obviously, your machine has to be fast enough to do the whole real-time audio computation in a little less than the time between two interrupts, that's the period size. And it must reliably be able to do this, because if it misses the time to have a block of samples ready for the DAC. If it misses that goal, a "underrun" will take place, and the audio application will have to resynchronize, possibly causing some clicking, intermittent audio, ...