Stand Alone Application
License : 5 (free for customers)
- FFX-16 i.o.p.
The very first steps
STEP -1- : Click in empty slot.
STEP -2- : Select a Plug-in.
STEP -3- : drag and drop to move it.
STEP -4- : Double click to display the GUI.
Latency Time !
Here, the latency time defines the global time elapsed between input and output signals (real-time
processing). It means the time the signal takes to go through your computer (with FFX16 i.o.p), or the
time it takes to execute the following tasks :
- Getting a signal from the inputs.
- Processing this signal.
- Sending the processed signal to the outputs.
What Latency Settings ?!
By default, FFX16 i.o.p's latency is set to a fairly high value (long latency), in order to guatantee
compatibility with most PC setups. But the user can choose shorter latency settings.
When using FFX16 i.o.p as a measurement system, a 50-ms latency is enough to ensure an acceptable
synchronization between what you can hear and what you can see on your screen. 30 ms usually corresponds
to the best compromise between real-time processing, display velocity and reaction time on most Windows
Genuine Real Time ?
If you really want to turn a PC into a real-time audio processing unit and use it this way in a live
context (process a singer's voice, create guitar effects...) you'll have to use very short latency times.
10 ms corresponds to an acceptable maximum setting (if you want to get some idea, 10 ms is the time it
takes a guitarist to hear the sound when playing 3.3 metres away from his amplifier) and 1ms, an ideal
setting which is impossible to reach reliably on general-purpose, non real-time operating systems (this is
why professional audio applications use dedicated expansion cards, in order to guarantee short latency
times in a reliable way).
FFX16 i.o.p. (Input / Output Processing) is the standalone version of
FFX16 Plug-in ((You will find more information about FFX16 concept and
features on the FFX16 information pages).
The i.o.p. version just brings you a way to use VB-AUDIO DX plug-ins without the need of any HOST Audio
programs. FFX16 i.o.p. gets audio from the inputs of your audio card, sends it into the rack to be
processed by the plug-in chain and sends the resulting signal to the outputs of your audio card. This is
real-time live processing
To Start processing a Stream,
simply click on the ON/OFF button. However, you'd better configure the rack before : by clicking the OPT
Each effect is represented in the rack by a 19-inch aluminum-styled unit with
SOLO and MUTE buttons. To display the effect's GUI (Graphic User Interface) you just have to double click
the unit or right click it to get access to the contextual popup menu.
Audio Format :
The AUDIO FORMAT dialog box allows you to define the audio format used by your
Driver (ASIO drivers always work with 32-bit float data).
Here, you can define the required sample rate and the DATA TYPE when possible.
Note that windows drivers can emulate and therefore offer data formats not really supported by your audio
card. It is always better to choose a format that your hardware can really handle (it is useless to select
24 bits integer, for example, or even 32 bits float if the audio card is not able to hardware-handle this
format) because FFX16 works with 32 bits float and already takes care of conversions (the less the driver
has to work the better real-time streams will get).
Note that with ASIO Drivers, only modifying the samplerate makes sense. ASIO Drivers handle 32 bits float
ASIO Drivers :
When you select ASIO drivers, a button called "ASIO CONTROL PANEL" appears.
Clicking it opens the ASIO control panel of the selected driver (the card's control panel). If the control
panel does not appear of if the system crashes, it means that there is a compatibility issue between FFX16
and your ASIO drivers. Then you'll have to select another driver.
Windows Drivers :
When selecting Windows driver (even the WAVE MAPPER) you can see 6 push buttons
appearing to define the granularity of the buffer frames used in the real time audio stream.
Sometimes, using buffer sizes which are multiples of numbers to the 2nd (like 16 or 32
samples, for example) improves the stability of real-time audio streams with low latency settings