Decimates (reduces the effective sample rate), and decreases the bit depth of the input signal, allows non integer values for smooth transitions between clean and lofi signals.
Bit depth
The bit depth that the signal will be reduced to.
Sample rate (Hz)
The sample rate that the signal will be resampled at.

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Amount of decimation:
How much the sample rate is reduced by. In a perfect world we would like this to be as low as possible to match the output to the input sample rate.
Resampler Description:
Bit depth:
The bit depth of the resampled output signal.
Join Description:
Amount of output signal shaping applied to the signal:
Levels:
Amount of output signal shaping applied to the signal:

For example, if a piece of audio was resampled at 44.1 kHz to 48 kHz and then the overall signal was rounded to 64 bits. It would be resampled at 64 bits. The resampled output would be 80 bits. Because of rounding to 32 bits, it will be at 44.1 kHz with 32 bits. So it would be a mix of 44.1 kHz and 44.1 kHz (the original, lower sampling rate) which is most likely not what was intended.

As shown in the table below, the “Normalize” setting resamples the input by the volume, or loudness, of the source. If the sample rate of the input is less than the target sample rate, the decimation rate is also set to the same value. If the sample rate is higher than the target rate, the result is the same, but the output sample rate will be increased, which may or may not be desirable depending on the application.

If the Normalize setting is set to Off, the input sample rate is not changed.
If the Normalize setting is set to On, the sample rate of the input is scaled by the target sample rate.

Reduce the sample rate of the signal to reduce file size.

Properties

Description

Available

Default

Sample Rate

The sample rate that the signal will be resampled at.

Yes

44.1 kHz

Bit Depth

The bit depth of the resampled output signal.

32

Join

Amount of output signal shaping applied to the signal:

Levels

Amount of output signal shaping applied to the signal:

The output signal has a flexible number of output levels and can be sent to either a single output or to multiple outputs. The following table explains how the input levels are interpreted.

When the input is applied to the channel mixer or gate, the following is applied to the signal:

Input

Volume

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Enables or disables the mode that makes all keys macro and macro global.
The method for decoding macros.
MACRO macro: how macros work
macro: Macro mode. This is the default mode for KEYMAPing.
global: Macro mode. Macro is non-local.
When this is enabled, all the keys macros and macros will be global.
The current KEYmacro layer.
When KEYmacro layer is enabled, the macro capability is controlled by the macro. When the macro is selected.
global, macros will always be global.
macro, macros will always be local.
macro, macros will always be non-local.
Keymacro Layer
This defines the layer where macros are placed. The current layer is set by the keymacro layer that is on the main KEYMAP page.
macro, global, macros, non-local
KEYMAPING: Keyboarding (Chords)
KEYMAPING: Keyboarding (Chords)
Layers > KEYMAPING > Keyboarding > KEYMAP
This is where the chords are loaded.
The level, which is the order of layer that the chords will be played, are set in the chord files.

The chords are set in the chord files, one at a time, in the following format.
>
Enabling or disabling a chord. (Can be for all chord or for a specific chord).
SETTINGS: Keyboard Modes
SETTINGS: Keyboard Modes
SETTINGS: Presets
SETTINGS: Presets
GENERAL SETTINGS > SPLIT KEYS > ON/OFF
Enable/disable the split mode.
When it is enabled, it splits the keyboard into two keyboard layers. When it is disabled.
Default
When the split keyboard layer is enabled, the split key is set to on. When the split key is off.
The mode that splits the keyboard.
The main keyboard layer, set on the main page of the layer menu.
SPLIT LAYER MENU:
Main Layer:
Split Layer:
The main keyboard layer is on the main page of the Layer menu. The split key is on the Split Layer page. The split keys are set to off.
There is a macro key for switching between the main and split keyboard layers.
SADLY, the current implementation of the split keyboard modes (to be added later) is
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‘*’: Standard FFT processor (default).
-l:
–lossless

Treat input and output files as lossless audio, meaning that the input and output
files are treated as samples. The lossless default is usually not the best option when
transcoding files. It is recommended to change to –lossy if the files are already
lossy, or use –resample-down to discard samples if the input and output are samples
already. Lossy conversion usually produces better results because there is no need to
discard samples.
Description:
‘*’: Lossy conversion.
-m:
–max-merge:

Maximum merge passes.
Increase the number of merge passes in the process of converting to a more efficient
lossy codec. If your input file is already lossy, lossy -m will not make sense, since
you’d be discarding data already. If your input is lossless, lossless -m will waste CPU
time merging data that you’re discarding.
E.g. if your input is WAV and you want to use –lossy, set –max-merge to 3.
Description:
2 to 10: Merge count. Default is 2.
-o:
–output-format:

Enforce output format for all channels.
Enforce a specific output format, if the input and output are a different format.
Available formats are:
1: Raw data. This format is used when the source and output are stereo.
2: Raw + signed PCM (raw + 8 bit). This format is used when the source and output
are stereo and the source is signed.
3: Raw + float PCM (raw + 32 bit). This format is used when the source and output are
stereo and the source is float.
4: Raw + float PCM (raw + 32 bit, big-endian). This format is used when the source and
output are stereo and the source is float, and the source and output are big endian.
5: Raw + float PCM (raw + 32 bit, little-endian). This format is used when the source
and output are stereo and the source is float, and the source and output are little
endian.
6: Raw + float PCM (raw + 32 bit, IEEE754). This format is used when the source and
output are

What’s New in the?

Original input audio is fed into the encoder with a sample rate of 44.1khz. The output of the encoder will be converted to a sample rate of 128khz and a bit depth of 16 bit.

Input : 44.1khz, output : 128khz, bit depth : 16

Resample:
Bands (discrete frequency bins)
The resampling step is based on the sample rate in the input signal. The number of bands is derived from the sample rate and the number of channels. See also #BandsForInput.

Description:
Originally the input sample rate is 44.1khz and will be resampled to 128khz and then to 44.1khz again. The number of bands for 44.1khz is 44.1.

Input : 44.1khz, output : 128khz, bit depth : 16, bands : 44.1

Bands (continuous frequency bins)
The number of bands for a continuous signal is the number of points per octave. In other words, the number of points per octave is related to the octave frequency spacing and the sample rate. The points per octave for a signal with 44.1khz and a bit depth of 16 is:
44.1k / (44.1k / 2 * log2(2)) = 16 points per octave

Note that the frequency spacing is determined by the sample rate.

Input : 44.1khz, output : 128khz, bit depth : 16, bands : 10

Description:

Original input signal: 44.1khz, output signal: 44.1khz.

Signal frequency in Hz: 44.1kHz

Number of bins: 10

Octave: 2

Number of points per octave: log2(2) = 0.69

Input bandwidth (Hz): log2(2) * 44.1kHz = 36kHz

Input sample rate (Hz): 44.1kHz

Input bandpass (Hz): input bandwidth / input sample rate = 36kHz / 44.1kHz = 0.71

Output bandpass (Hz): input bandwidth / output sample rate = 36kHz / 128khz = 0.28

Bandwidth reduction
The input bandpass is divided by the output bandpass. The output bandpass will be reduced to input bandpass / bandwidth reduction * sample rate.

Input : 44.1khz, output : 128khz, bit depth : 16, bands : 10, bandwidth reduction: 5

Notes

See also
Bit-depth reduction
Sample rate reduction
Notch filter
Windows media foundation codecs – WMA2 encoder

References

External links
Microsoft Windows Media Audio SDK
W

System Requirements:

Minimum:
OS: Windows 7 or later, Mac OS X 10.8 or later
Processor: Intel Core i5-2400, AMD Phenom II X2 555, or equivalent
Memory: 2 GB RAM
Graphics: NVIDIA GeForce GTX 660, AMD Radeon HD 7870, or equivalent
DirectX: Version 11
Storage: 100 GB available space
Additional Notes:
Requires a controller for the game to work properly.
Recommended:
OS: Windows 7 or later, Mac OS X 10.8

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