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/* == */
/* FMOD Studio - Built-in effects header file. */
/* Copyright (c), Firelight Technologies Pty, Ltd. 2004-2020. */
/* */
/* In this header you can find parameter structures for FMOD system registered DSP effects */
/* and generators. */
/* */
/* == */
/*
[ENUM]
[
[DESCRIPTION]
These definitions can be used for creating FMOD defined special effects or DSP units.
[REMARKS]
To get them to be active, first create the unit, then add it somewhere into the DSP network,
either at the front of the network near the soundcard unit to affect the global output
(by using System::getDSPHead), or on a single channel (using Channel::getDSPHead).
[SEE_ALSO]
System::createDSPByType
]
*/
typedef enum
FMOD_DSP_TYPE;
/*
===================================================================================================
FMOD built in effect parameters.
Use DSP::setParameter with these enums for the 'index' parameter.
===================================================================================================
*/
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_OSCILLATOR filter.
[REMARKS]
[SEE_ALSO]
DSP::setParameterFloat
DSP::setParameterInt
DSP::getParameterFloat
DSP::getParameterInt
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_OSCILLATOR;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_LOWPASS filter.
[REMARKS]
Deprecated and will be removed in a future release, to emulate with FMOD_DSP_TYPE_MULTIBAND_EQ:
// Configure a single band (band A) as a lowpass (all other bands default to off).
// 24dB rolloff to approximate the old effect curve.
// Cutoff frequency can be used the same as with the old effect.
// Resonance can be applied by setting the 'Q' value of the new effect.
FMOD_DSP_SetParameterInt(multiband, FMOD_DSP_MULTIBAND_EQ_A_FILTER, FMOD_DSP_MULTIBAND_EQ_FILTER_LOWPASS_24DB);
FMOD_DSP_SetParameterFloat(multiband, FMOD_DSP_MULTIBAND_EQ_A_FREQUENCY, frequency);
FMOD_DSP_SetParameterFloat(multiband, FMOD_DSP_MULTIBAND_EQ_A_Q, resonance);
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_LOWPASS;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_ITLOWPASS filter.<br>
This is different to the default FMOD_DSP_TYPE_ITLOWPASS filter in that it uses a different quality algorithm and is
the filter used to produce the correct sounding playback in .IT files.<br>
FMOD Studio's .IT playback uses this filter.<br>
[REMARKS]
Note! This filter actually has a limited cutoff frequency below the specified maximum, due to its limited design,
so for a more open range filter use FMOD_DSP_LOWPASS or if you don't mind not having resonance,
FMOD_DSP_LOWPASS_SIMPLE.<br>
The effective maximum cutoff is about 8060hz.
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_ITLOWPASS;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_HIGHPASS filter.
[REMARKS]
Deprecated and will be removed in a future release, to emulate with FMOD_DSP_TYPE_MULTIBAND_EQ:
// Configure a single band (band A) as a highpass (all other bands default to off).
// 12dB rolloff to approximate the old effect curve.
// Cutoff frequency can be used the same as with the old effect.
// Resonance can be applied by setting the 'Q' value of the new effect.
FMOD_DSP_SetParameterInt(multiband, FMOD_DSP_MULTIBAND_EQ_A_FILTER, FMOD_DSP_MULTIBAND_EQ_FILTER_HIGHPASS_12DB);
FMOD_DSP_SetParameterFloat(multiband, FMOD_DSP_MULTIBAND_EQ_A_FREQUENCY, frequency);
FMOD_DSP_SetParameterFloat(multiband, FMOD_DSP_MULTIBAND_EQ_A_Q, resonance);
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_HIGHPASS;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_ECHO filter.
[REMARKS]
Note. Every time the delay is changed, the plugin re-allocates the echo buffer. This means the echo will dissapear at that time while it refills its new buffer.<br>
Larger echo delays result in larger amounts of memory allocated.<br>
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_ECHO;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_FADER filter.
[REMARKS]
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum FMOD_DSP_FADER
FMOD_DSP_FADER;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_FLANGE filter.
[REMARKS]
Flange is an effect where the signal is played twice at the same time, and one copy slides back and forth creating a whooshing or flanging effect.<br>
As there are 2 copies of the same signal, by default each signal is given 50% mix, so that the total is not louder than the original unaffected signal.<br>
<br>
Flange depth is a percentage of a 10ms shift from the original signal. Anything above 10ms is not considered flange because to the ear it begins to 'echo' so 10ms is the highest value possible.<br>
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_FLANGE;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_DISTORTION filter.
[REMARKS]
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_DISTORTION;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_NORMALIZE filter.
[REMARKS]
Normalize amplifies the sound based on the maximum peaks within the signal.<br>
For example if the maximum peaks in the signal were 50% of the bandwidth, it would scale the whole sound by 2.<br>
The lower threshold value makes the normalizer ignores peaks below a certain point, to avoid over-amplification if a loud signal suddenly came in, and also to avoid amplifying to maximum things like background hiss.<br>
<br>
Because FMOD is a realtime audio processor, it doesn't have the luxury of knowing the peak for the whole sound (ie it can't see into the future), so it has to process data as it comes in.<br>
To avoid very sudden changes in volume level based on small samples of new data, fmod fades towards the desired amplification which makes for smooth gain control. The fadetime parameter can control this.<br>
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_NORMALIZE;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_LIMITER filter.
[REMARKS]
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_LIMITER;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_PARAMEQ filter.
[REMARKS]
Deprecated and will be removed in a future release, to emulate with FMOD_DSP_TYPE_MULTIBAND_EQ:
// Configure a single band (band A) as a peaking EQ (all other bands default to off).
// Center frequency can be used as with the old effect.
// Bandwidth can be applied by setting the 'Q' value of the new effect.
// Gain at the center frequency can be used the same as with the old effect.
FMOD_DSP_SetParameterInt(multiband, FMOD_DSP_MULTIBAND_EQ_A_FILTER, FMOD_DSP_MULTIBAND_EQ_FILTER_PEAKING);
FMOD_DSP_SetParameterFloat(multiband, FMOD_DSP_MULTIBAND_EQ_A_FREQUENCY, center);
FMOD_DSP_SetParameterFloat(multiband, FMOD_DSP_MULTIBAND_EQ_A_Q, bandwidth);
FMOD_DSP_SetParameterFloat(multiband, FMOD_DSP_MULTIBAND_EQ_A_GAIN, gain);
Parametric EQ is a single band peaking EQ filter that attenuates or amplifies a selected frequency and its neighbouring frequencies.
When a frequency has its gain set to 1.0, the sound will be unaffected and represents the original signal exactly.
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_PARAMEQ;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_MULTIBAND_EQ filter.
[REMARKS]
Flexible five band parametric equalizer.
[SEE_ALSO]
DSP::setParameterInt
DSP::getParameterInt
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum FMOD_DSP_MULTIBAND_EQ
FMOD_DSP_MULTIBAND_EQ;
/*
[ENUM]
[
[DESCRIPTION]
Filter types for FMOD_DSP_MULTIBAND_EQ.
[REMARKS]
[SEE_ALSO]
FMOD_DSP_MULTIBAND_EQ
]
*/
typedef enum FMOD_DSP_MULTIBAND_EQ_FILTER_TYPE
FMOD_DSP_MULTIBAND_EQ_FILTER_TYPE;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_PITCHSHIFT filter.
[REMARKS]
This pitch shifting unit can be used to change the pitch of a sound without speeding it up or slowing it down.<br>
It can also be used for time stretching or scaling, for example if the pitch was doubled, and the frequency of the sound was halved, the pitch of the sound would sound correct but it would be twice as slow.<br>
<br>
<b>Warning!</b> This filter is very computationally expensive! Similar to a vocoder, it requires several overlapping FFT and IFFT's to produce smooth output, and can require around 440mhz for 1 stereo 48khz signal using the default settings.<br>
Reducing the signal to mono will half the cpu usage.<br>
Reducing this will lower audio quality, but what settings to use are largely dependant on the sound being played. A noisy polyphonic signal will need higher fft size compared to a speaking voice for example.<br>
<br>
This pitch shifter is based on the pitch shifter code at http://www.dspdimension.com, written by Stephan M. Bernsee.<br>
The original code is COPYRIGHT 1999-2003 Stephan M. Bernsee <smb@dspdimension.com>.<br>
<br>
'<i>maxchannels</i>' dictates the amount of memory allocated. By default, the maxchannels value is 0. If FMOD is set to stereo, the pitch shift unit will allocate enough memory for 2 channels. If it is 5.1, it will allocate enough memory for a 6 channel pitch shift, etc.<br>
If the pitch shift effect is only ever applied to the global mix (ie it was added with ChannelGroup::addDSP), then 0 is the value to set as it will be enough to handle all speaker modes.<br>
When the pitch shift is added to a channel (ie Channel::addDSP) then the channel count that comes in could be anything from 1 to 8 possibly. It is only in this case where you might want to increase the channel count above the output's channel count.<br>
If a channel pitch shift is set to a lower number than the sound's channel count that is coming in, it will not pitch shift the sound.<br>
<br>
<b>NOTE!</b> Not supported on PlayStation 3.<br>
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
ChannelGroup::addDSP
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_PITCHSHIFT;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_CHORUS filter.
[REMARKS]
Chorus is an effect where the sound is more 'spacious' due to 1 to 3 versions of the sound being played along side the original signal but with the pitch of each copy modulating on a sine wave.<br>
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_CHORUS;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_ITECHO filter.<br>
This is effectively a software based echo filter that emulates the DirectX DMO echo effect. Impulse tracker files can support this, and FMOD will produce the effect on ANY platform, not just those that support DirectX effects!<br>
[REMARKS]
Note. Every time the delay is changed, the plugin re-allocates the echo buffer. This means the echo will dissapear at that time while it refills its new buffer.<br>
Larger echo delays result in larger amounts of memory allocated.<br>
<br>
As this is a stereo filter made mainly for IT playback, it is targeted for stereo signals.<br>
With mono signals only the FMOD_DSP_ITECHO_LEFTDELAY is used.<br>
For multichannel signals (>2) there will be no echo on those channels.<br>
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_ITECHO;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_COMPRESSOR unit.
This is a multichannel software limiter that is uniform across the whole spectrum.
[REMARKS]
The limiter is not guaranteed to catch every peak above the threshold level,
because it cannot apply gain reduction instantaneously - the time delay is
determined by the attack time. However setting the attack time too short will
distort the sound, so it is a compromise. High level peaks can be avoided by
using a short attack time - but not too short, and setting the threshold a few
decibels below the critical level.
<br>
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
DSP::setParameterBool
DSP::getParameterBool
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_COMPRESSOR;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_SFXREVERB unit.<br>
[REMARKS]
This is a high quality I3DL2 based reverb.<br>
On top of the I3DL2 property set, "Dry Level" is also included to allow the dry mix to be changed.<br>
<br>
These properties can be set with presets in FMOD_REVERB_PRESETS.
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
FMOD_REVERB_PRESETS
]
*/
typedef enum
FMOD_DSP_SFXREVERB;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_LOWPASS_SIMPLE filter.
[REMARKS]
Deprecated and will be removed in a future release, to emulate with FMOD_DSP_TYPE_MULTIBAND_EQ:
// Configure a single band (band A) as a lowpass (all other bands default to off).
// 12dB rolloff to approximate the old effect curve.
// Cutoff frequency can be used the same as with the old effect.
// Resonance / 'Q' should remain at default 0.707.
FMOD_DSP_SetParameterInt(multiband, FMOD_DSP_MULTIBAND_EQ_A_FILTER, FMOD_DSP_MULTIBAND_EQ_FILTER_LOWPASS_12DB);
FMOD_DSP_SetParameterFloat(multiband, FMOD_DSP_MULTIBAND_EQ_A_FREQUENCY, frequency);
This is a very simple low pass filter, based on two single-pole RC time-constant modules.
The emphasis is on speed rather than accuracy, so this should not be used for task requiring critical filtering.
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_LOWPASS_SIMPLE;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_DELAY filter.
[REMARKS]
Note. Every time MaxDelay is changed, the plugin re-allocates the delay buffer. This means the delay will dissapear at that time while it refills its new buffer.<br>
A larger MaxDelay results in larger amounts of memory allocated.<br>
Channel delays above MaxDelay will be clipped to MaxDelay and the delay buffer will not be resized.<br>
<br>
<b>NOTE!</b> Not supported on PlayStation 3.
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_DELAY;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_TREMOLO filter.
[REMARKS]
The tremolo effect varies the amplitude of a sound. Depending on the settings, this unit can produce a tremolo, chopper or auto-pan effect.<br>
<br>
The shape of the LFO (low freq. oscillator) can morphed between sine, triangle and sawtooth waves using the FMOD_DSP_TREMOLO_SHAPE and FMOD_DSP_TREMOLO_SKEW parameters.<br>
FMOD_DSP_TREMOLO_DUTY and FMOD_DSP_TREMOLO_SQUARE are useful for a chopper-type effect where the first controls the on-time duration and second controls the flatness of the envelope.<br>
FMOD_DSP_TREMOLO_SPREAD varies the LFO phase between channels to get an auto-pan effect. This works best with a sine shape LFO.<br>
The LFO can be synchronized using the FMOD_DSP_TREMOLO_PHASE parameter which sets its instantaneous phase.<br>
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_TREMOLO;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_SEND DSP.
[REMARKS]
[SEE_ALSO]
DSP::setParameterInt
DSP::getParameterInt
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_SEND;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_RETURN DSP.
[REMARKS]
[SEE_ALSO]
DSP::setParameterInt
DSP::getParameterInt
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_RETURN;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_HIGHPASS_SIMPLE filter.
[REMARKS]
Deprecated and will be removed in a future release, to emulate with FMOD_DSP_TYPE_MULTIBAND_EQ:
// Configure a single band (band A) as a highpass (all other bands default to off).
// 12dB rolloff to approximate the old effect curve.
// Cutoff frequency can be used the same as with the old effect.
// Resonance / 'Q' should remain at default 0.707.
FMOD_DSP_SetParameterInt(multiband, FMOD_DSP_MULTIBAND_EQ_A_FILTER, FMOD_DSP_MULTIBAND_EQ_FILTER_HIGHPASS_12DB);
FMOD_DSP_SetParameterFloat(multiband, FMOD_DSP_MULTIBAND_EQ_A_FREQUENCY, frequency);
This is a very simple single-order high pass filter.
The emphasis is on speed rather than accuracy, so this should not be used for task requiring critical filtering.
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_HIGHPASS_SIMPLE;
/*
[ENUM]
[
[DESCRIPTION]
Parameter values for the FMOD_DSP_PAN_2D_STEREO_MODE parameter of the FMOD_DSP_TYPE_PAN DSP.
[REMARKS]
[SEE_ALSO]
FMOD_DSP_PAN
]
*/
typedef enum
FMOD_DSP_PAN_2D_STEREO_MODE_TYPE;
/*
[ENUM]
[
[DESCRIPTION]
Parameter values for the FMOD_DSP_PAN_MODE parameter of the FMOD_DSP_TYPE_PAN DSP.
[REMARKS]
[SEE_ALSO]
FMOD_DSP_PAN
]
*/
typedef enum
FMOD_DSP_PAN_MODE_TYPE;
/*
[ENUM]
[
[DESCRIPTION]
Parameter values for the FMOD_DSP_PAN_3D_ROLLOFF parameter of the FMOD_DSP_TYPE_PAN DSP.
[REMARKS]
[SEE_ALSO]
FMOD_DSP_PAN
]
*/
typedef enum
FMOD_DSP_PAN_3D_ROLLOFF_TYPE;
/*
[ENUM]
[
[DESCRIPTION]
Parameter values for the FMOD_DSP_PAN_3D_EXTENT_MODE parameter of the FMOD_DSP_TYPE_PAN DSP.
[REMARKS]
[SEE_ALSO]
FMOD_DSP_PAN
]
*/
typedef enum
FMOD_DSP_PAN_3D_EXTENT_MODE_TYPE;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_PAN DSP.
[REMARKS]
FMOD_DSP_PAN_3D_PAN_BLEND controls the percentage of the effect supplied by FMOD_DSP_PAN_2D_DIRECTION and FMOD_DSP_PAN_2D_EXTENT.
For FMOD_DSP_PAN_3D_POSITION, the following members in the FMOD_DSP_PARAMETER_3DATTRIBUTES_MULTI struct should be non zero.
- numlisteners - This is typically 1, can be up to 8. Typically more than 1 is only used for split screen purposes. The FMOD Panner will average angles and produce the best compromise for panning and attenuation.
- relative[listenernum].position - This is the delta between the listener position and the sound position. Typically the listener position is subtracted from the sound position.
- relative[listenernum].forward - This is the sound's forward vector. Optional, set to 0,0,1 if not needed. This is only relevant for more than mono sounds in 3D, that are spread amongst the destination speakers at the time of panning.
If the sound rotates then the L/R part of a stereo sound will rotate amongst its destination speakers.
If the sound has moved and pinpointed into a single speaker, rotation of the sound will have no effect as at that point the channels are collapsed into a single point.
For FMOD_DSP_PAN_2D_STEREO_MODE, when it is set to FMOD_DSP_PAN_2D_STEREO_MODE_DISCRETE, only FMOD_DSP_PAN_2D_STEREO_SEPARATION and FMOD_DSP_PAN_2D_STEREO_AXIS are used.
When it is set to FMOD_DSP_PAN_2D_STEREO_MODE_DISTRIBUTED, then standard FMOD_DSP_PAN_2D_DIRECTION/FMOD_DSP_PAN_2D_EXTENT parameters are used.
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
DSP::setParameterInt
DSP::getParameterInt
DSP::setParameterData
DSP::getParameterData
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_PAN;
/*
[ENUM]
[
[DESCRIPTION]
Parameter values for the FMOD_DSP_THREE_EQ_CROSSOVERSLOPE parameter of the FMOD_DSP_TYPE_THREE_EQ DSP.
[REMARKS]
[SEE_ALSO]
FMOD_DSP_THREE_EQ
]
*/
typedef enum
FMOD_DSP_THREE_EQ_CROSSOVERSLOPE_TYPE;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_THREE_EQ filter.
[REMARKS]
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
DSP::setParameterInt
DSP::getParameterInt
FMOD_DSP_TYPE
FMOD_DSP_THREE_EQ_CROSSOVERSLOPE_TYPE
]
*/
typedef enum
FMOD_DSP_THREE_EQ;
/*
[ENUM]
[
[DESCRIPTION]
List of windowing methods for the FMOD_DSP_TYPE_FFT unit. Used in spectrum analysis to reduce leakage / transient signals intefering with the analysis.<br>
This is a problem with analysis of continuous signals that only have a small portion of the signal sample (the fft window size).<br>
Windowing the signal with a curve or triangle tapers the sides of the fft window to help alleviate this problem.
[REMARKS]
Cyclic signals such as a sine wave that repeat their cycle in a multiple of the window size do not need windowing.<br>
I.e. If the sine wave repeats every 1024, 512, 256 etc samples and the FMOD fft window is 1024, then the signal would not need windowing.<br>
Not windowing is the same as FMOD_DSP_FFT_WINDOW_RECT, which is the default.<br>
If the cycle of the signal (ie the sine wave) is not a multiple of the window size, it will cause frequency abnormalities, so a different windowing method is needed.<br>
<exclude>
<br>
FMOD_DSP_FFT_WINDOW_RECT.<br>
<img src="..\static\overview\rectangle.gif"></img><br>
<br>
FMOD_DSP_FFT_WINDOW_TRIANGLE.<br>
<img src="..\static\overview\triangle.gif"></img><br>
<br>
FMOD_DSP_FFT_WINDOW_HAMMING.<br>
<img src="..\static\overview\hamming.gif"></img><br>
<br>
FMOD_DSP_FFT_WINDOW_HANNING.<br>
<img src="..\static\overview\hanning.gif"></img><br>
<br>
FMOD_DSP_FFT_WINDOW_BLACKMAN.<br>
<img src="..\static\overview\blackman.gif"></img><br>
<br>
FMOD_DSP_FFT_WINDOW_BLACKMANHARRIS.<br>
<img src="..\static\overview\blackmanharris.gif"></img>
</exclude>
[SEE_ALSO]
FMOD_DSP_FFT
]
*/
typedef enum
FMOD_DSP_FFT_WINDOW;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_FFT dsp effect.
[REMARKS]
Set the attributes for the spectrum analysis with FMOD_DSP_FFT_WINDOWSIZE and FMOD_DSP_FFT_WINDOWTYPE, and retrieve the results with FMOD_DSP_FFT_SPECTRUM and FMOD_DSP_FFT_DOMINANT_FREQ.
FMOD_DSP_FFT_SPECTRUM stores its data in the FMOD_DSP_PARAMETER_DATA_TYPE_FFT. You will need to cast to this structure to get the right data.
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
DSP::setParameterInt
DSP::getParameterInt
DSP::setParameterData
DSP::getParameterData
FMOD_DSP_TYPE
FMOD_DSP_FFT_WINDOW
]
*/
typedef enum
FMOD_DSP_FFT;
/*
[ENUM]
[
[DESCRIPTION]
Deprecated and will be removed in a future release.
Parameter types for the FMOD_DSP_TYPE_ENVELOPEFOLLOWER unit.
This is a simple envelope follower for tracking the signal level.<br>
[REMARKS]
Deprecated and will be removed in a future release.
This unit does not affect the incoming signal.
<br>
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
DSP::setParameterData
DSP::getParameterData
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_ENVELOPEFOLLOWER;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_CONVOLUTIONREVERB filter.
[REMARKS]
Convolution Reverb reverb IR.<br>
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
DSP::setParameterData
DSP::getParameterData
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_CONVOLUTION_REVERB;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_CHANNELMIX_OUTPUTGROUPING parameter for FMOD_DSP_TYPE_CHANNELMIX effect.
[REMARKS]
[SEE_ALSO]
DSP::setParameterInt
DSP::getParameterInt
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_CHANNELMIX_OUTPUT;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_CHANNELMIX filter.
[REMARKS]
For FMOD_DSP_CHANNELMIX_OUTPUTGROUPING, this value will set the output speaker format for the DSP, and also map the incoming channels to the
outgoing channels in a round-robin fashion. Use this for example play a 32 channel input signal as if it were a repeating group of output signals.
Ie.
FMOD_DSP_CHANNELMIX_OUTPUT_ALLMONO = all incoming channels are mixed to a mono output.
FMOD_DSP_CHANNELMIX_OUTPUT_ALLSTEREO = all incoming channels are mixed to a stereo output, ie even incoming channels 0,2,4,6,etc are mixed to left, and odd incoming channels 1,3,5,7,etc are mixed to right.
FMOD_DSP_CHANNELMIX_OUTPUT_ALL5POINT1 = all incoming channels are mixed to a 5.1 output. If there are less than 6 coming in, it will just fill the first n channels in the 6 output channels.
If there are more, then it will repeat the input pattern to the output like it did with the stereo case, ie 12 incoming channels are mapped as 0-5 mixed to the
5.1 output and 6 to 11 mapped to the 5.1 output.
FMOD_DSP_CHANNELMIX_OUTPUT_ALLLFE = all incoming channels are mixed to a 5.1 output but via the LFE channel only.
[SEE_ALSO]
DSP::setParameterInt
DSP::getParameterInt
DSP::setParameterFloat
DSP::getParameterFloat
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_CHANNELMIX;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TRANSCEIVER_SPEAKERMODE parameter for FMOD_DSP_TYPE_TRANSCEIVER effect.
[REMARKS]
The speaker mode of a transceiver buffer (of which there are up to 32 of) is determined automatically depending on the signal flowing through the transceiver effect, or it can be forced.
Use a smaller fixed speaker mode buffer to save memory.
Only relevant for transmitter dsps, as they control the format of the transceiver channel's buffer.
If multiple transceivers transmit to a single buffer in different speaker modes, it will allocate memory for each speaker mode. This uses more memory than a single speaker mode.
If there are multiple receivers reading from a channel with multiple speaker modes, it will read them all and mix them together.
If the system's speaker mode is stereo or mono, it will not create a 3rd buffer, it will just use the mono/stereo speaker mode buffer.
[SEE_ALSO]
DSP::setParameterInt
DSP::getParameterInt
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_TRANSCEIVER_SPEAKERMODE;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_TRANSCEIVER filter.
[REMARKS]
The transceiver only transmits and receives to a global array of 32 channels. The transceiver can be set to receiver mode (like a return) and can receive the signal at a variable gain (FMOD_DSP_TRANSCEIVER_GAIN).
The transceiver can also be set to transmit to a chnnel (like a send) and can transmit the signal with a variable gain (FMOD_DSP_TRANSCEIVER_GAIN).
The FMOD_DSP_TRANSCEIVER_TRANSMITSPEAKERMODE is only applicable to the transmission format, not the receive format. This means this parameter is ignored in 'receive mode'. This allows receivers to receive at
the speaker mode of the user's choice. Receiving from a mono channel, is cheaper than receiving from a surround channel for example.
The 3 speaker modes FMOD_DSP_TRANSCEIVER_SPEAKERMODE_MONO, FMOD_DSP_TRANSCEIVER_SPEAKERMODE_STEREO, FMOD_DSP_TRANSCEIVER_SPEAKERMODE_SURROUND are stored as seperate buffers in memory for a tranmitter channel.
To save memory, use 1 common speaker mode for a transmitter.
The transceiver is double buffered to avoid desyncing of transmitters and receivers. This means there will be a 1 block delay on a receiver, compared to the data sent from a transmitter.
Multiple transmitters sending to the same channel will be mixed together.
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
DSP::setParameterInt
DSP::getParameterInt
DSP::setParameterBool
DSP::getParameterBool
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_TRANSCEIVER;
/*
[ENUM]
[
[DESCRIPTION]
Parameter types for the FMOD_DSP_TYPE_OBJECTPAN DSP. 3D Object panners are meant for hardware 3d object systems like Dolby Atmos or Sony Morpheus.
These object panners take input in, and send it to the 7.1 bed, but do not send the signal further down the DSP chain (the output of the dsp is silence).
[REMARKS]
[SEE_ALSO]
DSP::setParameterFloat
DSP::getParameterFloat
DSP::setParameterInt
DSP::getParameterInt
DSP::setParameterData
DSP::getParameterData
FMOD_DSP_TYPE
]
*/
typedef enum
FMOD_DSP_OBJECTPAN;