[LAD] Latency measurements: External round-trip vs. absolute in/out values

[Fons Adriaensen]

On Fri, Apr 05, 2019 at 06:38:59PM -0400, Tim wrote:
 
> [PC] -->--[ex. Par. port]-->--[Triggers]-->--[Sin oscs]
>      ---                                         |
>        |                                         |
>        --<--[Audio input]---------------<---------
> 
> It's similar to how round-trip latency might be measured,
>  except there the sines are generated digitally by audio output.

You don't even need the oscillators. Just connect the digital
output to a line input. If you make sure the pulse doesn't
overload the audio input (using resistors to attenuate it
if necessary), then the delay can easily be measured with
sub-sample accuracy.

The tricky part of is how to define input or output
latency in an unambiguous way, and to understand when
and how it matters. It doesn't for example when you
record a new track via the sound card (i.e. using a mic
or DI of an external instrument) while listening to
already recorded tracks. In that case only the round-trip
latency matters.

It *does* matter when recording a new track which is
not input from the sound card, but generated internally.
In that case you have input latency on the MIDI input
which drives the software instrument, and output latency
on the audio output playing back the already recorded
tracks. This again is a round-trip latency, but with
the two components originating in different systems.

It also matters when you you want to find out to which
audio sample a mouse click or keyboard event corresponds,
e.g. to mark a position for editing.

Let's assume you can generate a digital output pulse at 
a time measured by jack_time(). On a Raspberry for example
you could use one of GPIO pins which can be manipulated
from user space with almost no system delay.

Somewhat later, a Jack client will detect this pulse in
the audio stream. Note that the time at which this Jack
client runs is irrelevant. A Jack client can run anywhere
in the current cycle, it doesn't make any difference for
the audio processing.

The time that matters is the start time of the Jack cycle
in which the pulse is detected in the audio stream, plus
the time represented by the position of the pulse in the
current period. You can get the start and end times of
the current cycle using jack_get_cycle_times().

With

Tp     = jack time at which the digital pulse is generated
T0, T1 = the start and end times of the current cycle
k      = the position (in samples) of the pulse in the period
N      = the period size

then the input latency (in microsends) is

   T0 + k * (T1 - T0) / N - Tp


Ciao,

-- 
FA