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SYSTEM SETTINGS
Audio Settings
The Audio Settings page contains options and settings concerning
the live sound (wave) output from FL Studio. The options change
depending if ASIO or Direct Sound (standard) drivers are selected
in the Output selector.
- Output - Select the 'live' output device to be
used by FL Studio from the list of loaded soundcard drivers
(DirectSound and ASIO driver standards are supported). If you have
more than one soundcard installed, this control can be used to
switch between them. Select ASIO drivers if
possible, they are identified by word 'ASIO' somewhere in
the name. ASIO (Audio Stream Input/Output) drivers allow the
soundcard to communicate with the host computer with high
efficiency, so that the audio stream is processed with much lower
delays than with standard audio drivers. ASIO generally allows
lower latencies with less CPU load when compared to Direct Sound
drivers. If your soundcard does not 'natively' support ASIO, there
is a 3rd party soundcard driver 'work-around' you can try at
www.asio4all.com.
Please be aware that you use this ASIO driver at your own
risk. If possible, avoid drivers which
have 'emulated' in the name, as they have worse performance than
DirectSound or ASIO drivers.
- Auto close device - Releases soundcard output
when FL Studio looses focus, so other applications may use the same
output.
-
ASIO Drivers:
Visible only when using ASIO driver.
- Buffer Length - Shows information about the
buffer latency the ASIO device will use and is a 'read-only'
property. As the buffer is increased the delay between playing a
MIDI keyboard, tweaking a knob or making changes in FL Studio and
the sound response increases. The aim is to minimize the buffer
size without causing buffer underruns (described below). The delay
is equal to the buffer setting, shown in ms.
- Clock Source - Some audio cards might provide
external clock source which can fix sync/output problems. However
for most cards work properly with the default
"Internal" source selected.
- CPU Limit - Due to architectural specifics of
ASIO drivers, high CPU usage in some projects might render the host
computer system non-responsive. Enabling this option allows FL
Studio do detect CPU peaks and prevent such system lockups.
- Show ASIO Panel - Opens the ASIO driver
settings panel, use this to change latency settings. Settings
between 1-4 ms without underruns are 'cutting edge', 5-10 ms are
excellent and 11-20 ms are good.
- Priority - Sets the priority of the audio
mixing thread. Higher = more CPU devoted to the audio mixing
thread, but increases the risk of lockups/freezing when CPU demands
become high. Lower = greater risk of buffer underruns. Adjust this
(in combination with the buffer settings) if you have problems with
lockups and or buffer underruns.
- Safe overloads - Off: The
audio mixing thread is given a very high priority, so that the GUI
doesn't cause hiccups in the audio engine. When the audio mixing
thread is using all the CPU, it may leave nothing to the Graphical
User Interface (GUI), which will then appear frozen.
On (default): 'Safe overloads' adapts the mixer
priority when CPU overloads occur, leaving a little CPU to run the
GUI, so that you can sill interact with FL and minimize the CPU
usage.
- Underruns - Shows the total underrun count with the current
settings. An underrun is counted when the temporary store that
holds audio prior to output to your soundcard runs out of data (a
click or pop is usually heard), it means the CPU didn't process
information fast enough. Testing to reduce underruns should be
carried out with a typical project (song) playing. There are a
number of ways of reducing underruns as described below. After each
change, if the underrun count stops increasing, try to reduce the
Buffer length setting further. Your goal is to
find the shortest setting with no new underruns:
- First, a reminder that as the Buffer length is
increased, underruns decrease, but the delay between playing a MIDI
keyboard, tweaking a knob and the response of FL Studio also
increases. The aim is to minimize the buffer size without causing
buffer underruns. For ASIO drivers, settings of 1-4 ms are 'cutting
edge', 5-10 ms are 'excellent' and 11-20 ms are 'good'.
- Make sure the Mixer Interpolation is set to
Linear and the Sample rate is
48,000 Hz or less.
- Increase the audio thread 'Priority' setting
to 'Highest'.
- Turn the 'Safe overloads' switch off.
- Download the latest ASIO drivers from your
soundcard manufacturer. We recommend sticking with the native ASIO
drivers and only trying alternatives if you experience problems
with them.
- In some cases the 3rd party www.asio4all.com drivers outperform native ASIO
drivers and may resolve underrun issues. Please be aware
that you use this ASIO driver at your own risk.
- Decrease polyphony of the instrument
channels.
- Turn off 'Keep on disk' for Sampler and
Audio-Clip channels. This loads samples into memory which is
faster.
- Record
mixer channels to audio and disable the instruments feeding those
mixer channels.
- Note: If your Buffer length setting is greater
than 50 ms and your CPU usage meter peaks over 80%, it may be
simply be your computer is not fast enough to play the project.
Welcome to the never ending cycle of PC upgrades!
Standard Drivers:
Visible only when using Standard drivers (DirectSound,
WDM, Primary etc).
- Buffer Length - This slider controls the audio
buffer latency. As the buffer is increased the delay between
playing a MIDI keyboard, tweaking a knob or making changes in FL
Studio and the sound response increases. The aim is to minimize the
buffer size without causing buffer underruns (described below). The delay
is equal to the buffer setting, shown in ms. Setting between 5-10
ms without underruns are 'cutting edge', 11-20 ms are excellent and
21-50 ms are good.
- Use Polling - Polling is a technique for
managing DirectSound's audio buffer, which usually allows much
smaller buffer without underruns. On some PC-s, however, it
can have the opposite effect.
- Use Hardware Buffer - Uses the hardware audio
buffer of DirectSound enabled sound cards.
- Priority - Sets the priority of the audio
mixing thread. Higher = more CPU devoted to the audio mixing
thread, but increases the risk of lockups/freezing when CPU demands
become high. Lower = greater risk of buffer underruns. Adjust this
(in combination with the buffer settings) if you have problems with
lockups and or buffer underruns.
- Safe overloads - Off: The
audio mixing thread is given a very high priority, so that the GUI
doesn't cause hiccups in the audio engine. When the audio mixing
thread is using all the CPU, it may leave nothing to the Graphical
User Interface (GUI), which will then appear frozen.
On (default): 'Safe overloads' adapts the mixer
priority when CPU overloads occur, leaving a little CPU to run the
GUI, so that you can sill interact with FL and minimize the CPU
usage.
- Underruns - Shows the total underrun count with the current
settings. An underrun is counted when the temporary store that
holds audio prior to output to your soundcard runs out of data (a
click or pop is usually heard), it means the CPU didn't process
information fast enough. Testing to reduce underruns should be
carried out with a typical project (song) playing. There are a
number of ways of reducing underruns as described below. After each
change, if the underrun count stops increasing, try to reduce the
Buffer length setting further. Your goal is to
find the shortest setting with no new underruns:
-
- First, a reminder that as the Buffer length is
increased, underruns decrease, but the delay between playing a MIDI
keyboard, tweaking a knob and the response of FL Studio also
increases. The aim is to minimize the buffer size without causing
buffer underruns. For standard drivers, settings of 5-10 ms are
'cutting edge', 11-20 ms are 'excellent' and 21-50 ms are
'good'.
- Make sure the Mixer Interpolation is set to
Linear and the Sample rate is
48000 Hz or less.
- Increase the audio thread 'Priority' setting
to 'Highest'.
- Turn the 'Safe overloads' switch off.
- Try each of the 4 possible combinations of the 'Use
polling' and 'Use hardware buffer'
switches. 4 combinations? Two switches with two states (on/off)
mean 4 possible combinations, try them all.
- Switch to ASIO mode (if supported by your
soundcard). There is also 3rd party 'work-around' at a www.asio4all.com
that allows many non native ASIO soundcards to operate in ASIO
mode. Please be aware that you use this ASIO driver at your
own risk.
- Decrease polyphony of the instrument
channels.
- Turn off 'Keep on disk' for Sampler and
Audio-Clip channels. This loads samples into memory which is
faster.
- Record
mixer channels to audio and disable the instruments feeding those
mixer channels.
- Note: If your Buffer length setting is greater
than 100 ms and your CPU usage meter peaks over 80%, it may be
simply be your computer is not fast enough to play the project.
Welcome to the never ending cycle of PC upgrades!
- Auto Close Device - Releases the wave output
device when FL Studio loses focus, so other applications may use
the same output.
Plugin output:
Visible only when using FL Studio with the VSTi/DXi connection plugin or as a
ReWire client.
- Slave Tempo - When turned on, FL Studio will
synchronize with the tempo of the host.
- Record Automation - When turned on, remote
control messages (MIDI) from the host will be recorded during
recording sessions.
Mixer:
- Sample Rate - Sets the sample play-back rate
used by the mixer. Where possible use the default sample rate of
44100Hz. Many Creative brand cards (the Audigy series for example)
have a minimum sample rate of 48000Hz. In this case, please be
aware that some older plugins may not perform correctly (usually
tuning related issues) although the majority of plugins today are
multi-rate compatible.
- Interpolation - Sets the live playback sample
interpolation method for sampler channels (as opposed to the same
settings found in the rendering/exporting audio
dialog). NOTE: The interpolation used for
exporting audio files is set in the export dialog, so you do not need
to modify the Audio Settings interpolation method once set.
- Interpolation is the process of
smoothly creating changes in sample data when the system is called
to ‘invent’ intermediate volume levels between any two known sample
points. This happens when samples are transposed from their
original pitch, so the benefits of higher quality interpolation
will only be audible for transposed samples. FL Studio provides
several methods -
- Linear interpolation provides the lowest CPU
hit with basic linear averaging between samples, however this may
result in aliasing (high frequency noises) when samples are
transposed far from their original pitch. We recommend linear
settings for most live mixing situations.
- 6-point Hermite is the fastest curve
interpolation method and as it provides superior quality to linear
interpolation. If you have a fast PC you may like to try this
method during critical mixing sessions. However it will use more
CPU than linear.
- 64, 128, 256, 512-point Sinc methods provide,
increasingly, the highest quality interpolation, but they are also
very cpu intensive. Anything above 64-point Sync is not
suitable for live-playback (perhaps one day when we have
8-core 5 GHz CPUs). So why are these methods available? So that if
someone requires the highest quality live interpolation they can
have it.
- Reset Plugins on Transport - Resets all
plugins when changing song position, starting playback etc. Uncheck
for faster response when changing song position.
- Use Mixer as Playback Position - Enable this
option if you experience unstable position indicators with WDM
drivers (usually under Windows 2000/XP). If you don't have any
problems leave this option disabled as it reduces position
indicators' refresh rate with large audio buffers.
- Preview Mixer Track -
Select the mixer track to be used for metronome clicks, audio
previews from the Browser and Wave Editor, etc. By default these sounds are
sent to the Master Track by (select none /"--"/ to
send to the Master Track).
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