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AAS Strum Acoustic GS-1 Manual

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70 Audio and MIDI Settings
8.2.7 Defining a Default MIDI Map
It is possible to define a default MIDI map that will be loaded automatically when Strum Acoustic
is launched.
• First select a MIDI map by clicking on its icon in the browser and choose the MIDI Link
Info command from the Edit or the Ctrl-I/Apple-I keyboard shortcut. One can also right-
click/control-click on the MIDI map icon and choose the MIDI Link Info command.
• To change the default MIDI map select the Mark As Default option.
8.2.8 MIDI Program Changes
MIDI program changes can be used to switch between programs while playing. Strum Acoustic
will change the number of the current program used by the synthesis engine to the number corre-
sponding to the MIDI program change received by the application.
8.3 Latency Settings
The latency is the time delay between the moment you send a control signal to your computer (for
example when you hit a key on your MIDI keyboard) and the moment when you hear the effect.
Roughly, the latency will be equal to the duration of the buffers used by the application and the
sound card to play audio and MIDI. To calculate the total time required to play a buffer, just divide
the number of samples per buffer by the sampling frequency. For example, 256 samples played
at 48 kHz represent a time of 5.3 ms. Doubling the number of samples and keeping the sampling
frequency constant will double this time while changing the sampling frequency to 96 kHz and
keeping the buffer size constant will reduce the latency to 2.7 ms.
It is of course desirable to have as little latency as possible. Strum Acoustic however requires
a certain amount of time to be able to calculate sound samples in a continuous manner. This time
depends on the power of your computer, the preset played, the sampling rate, and the number of
voices of polyphony used. Note that it will literally take twice as much CPU power to process
audio at a sampling rate of 96 kHz as it would to process the same data at 48 kHz, simply because
you need to calculate twice as many samples in the same amount of time.
Depending on your machine you should choose, for a given sampling frequency, the smallest
buffer size that allows you to keep real-time for a reasonable number of voices of polyphony. To
adjust these parameters:
• Launch the Audio Control Panel
• Choose the sampling frequency and the audio format (16, 24, 32 bits)
• Adjust the buffer size
Note that this might not be possible on Mac OS or with ASIO drivers on Windows.
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