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Audio Mastering Suite

Download the JAVA-Application (version 2.0 - 28/04/2010) (36kByte) which works for Windows, Mac and
If you don't have the JRE (JAVA Runtime Environment) installed then you'll get it here:

- Native signal processing by using the resolution and sampling frequency of the source material
- The supported audio format is wav (PCM) 16/24Bit with 44.1kHz, 48kHz, 88.2kHz, 176.4kHz, 96kHz or 192kHz sampling

- High quality resampling of any audio format into 16Bit/44.1kHz by applying "Bandlimited Interpolation"
- 64Bit floating point signal processing
- The signal processing chain (order of components) is freely configurable.
- Audio monitoring switchable between input and output signal:
+ Level Meter with peak and average signal indication
+ Clipping detector
+ Correlation Meter to assure mono compatibility
+ Spectrum Analyzer
+ Oscilloscope with a 5ms time base (mono signal)
+ Vector Scope to validate the stereo and mono compatibility
- 8 band parametric equalizer (Q, Frequency [HZ], Gain [db]) providing the following filter functions:
+ EQ (standard peak equalizer)
+ Low Shelf
+ High Shelf
+ Low Pass
+ High Pass
+ Band Pass
+ Notch
- Stereo Imager
+ Convert to mono
+ Channel balance in dB
+ Center (mono) level (0-100%)
+ Side (left & right) level (0-100%)
- Ambiophonics
+ Creating a deeper and broader sound stage by applying RACE (Recursive Ambiophonic Crosstalk
+ RACE Attenuation (1dB to 9dB)
+ Center Presence (0% to 100%)
+ Delay depending on the sampling frequency of the source material (n x 1/Sampling Frequency)
+ Gain (-15dB to 0dB) to reduce the signal amplitude after applying RACE
- Vintage Tube Warmer/Maximizer
+ Adds harmonic distortions to simulate the behavior of a tube amplifier and compressor to warm and maximize the signal
- Compressor:
+ Input gain (-20dB to 20dB)
+ Threshold (-60dB to 0dB)
+ Ratio (1:1 up to 10:1)
+ Attack (0 - 30ms)

+ Release (0 - 2s)
+ Gain reduction indicator
+ Make up gain (0 - 30dB)
- Limiter:
+ Input gain (-20dB to 20dB)
+ Threshold (-10dB to 0dB)
+ Release (0 - 1s)
+ Gain reduction indicator

The User Interface

File & Info Menu:
- Online Info - -> Links to the on hand web page
- Exit -> Closes the program

Audio Monitoring:

Level Indicator:
The level meter represents the peak and average values of the audio signal.
Clipping Indicator:
Counts any 0dB overshoot.

Correlation Meter:
Indicates whether the phase sum of the left and right channel is in general mono compatible. The instrument should be used
together with the vector scope to get a reliable indication for mono compatibility.
Spectrum Analyzer:
Represents the amplitude spectrum of the signal between 50Hz and 20kHz (-80dB to 0dB).
The oscilloscope draws the mono signal (Left + Right) with a time base of 5ms.
Vector Scope:
Helps to judge the stereo distribution (L = Left; C = Center; R = Right).
Monitoring Input/Output and Volume:
Allows monitoring (Audio and Indicator) of the input or output signal.
The volume slider sets the audio monitoring level.

Audio Input/Player/Output:

The audio player allows play back during setting of the equalizer, compressor and limiter parameter. The monitoring signal always
uses 16Bit with the native source signal sampling frequency. Please make sure that your sound card is able to handle the audio

Opens a dialog to load the audio file for mastering.
Audio File:
The filename of the loaded audio file.
Resolution & Sample Rate:
Indicates the source signal audio format.
The button changes between "Play" and "Pause" to control the audio playback.
The slider indicates the current position and allows fast moving within the audio file.
Render & Export:
Exports the Audio file by applying the Audio Mastering Suite components and their parameters.
Attention: It does not warn whether the file already exists!
16Bit / 44.1kHz Output:
If selected then the exporting algorithm applies high quality "Bandlimited Interpolation" to resample to 16Bit and 44.1kHz sampling
rate for Compact Disc production. If that option is not selected then the audio output in sample resolution (16/24Bit) and sample
frequency (44.1kHz up to 192kHz) is just the same as the input format.
Please keep in mind that the resampling algorithm is processing intensive and therefore needs some computing time.
Component Chain:
The order of the components is freely configurable by clicking the exchange buttons [< >].

Parametric Equalizer:

Equalizer Bypass:
If the check box is selected then the audio signal bypasses the equalizer.
Updates the equalizer parameter. The update usually happens automatically by hitting the <return> key after changing a
The equalizer works like an analog parametric equalizer which changes the phase of the signal. If the phase check box is selected
then the green curve represents the phase change caused by the selected filter combination.
Filter Type:
The following filter types are available:
+ EQ (standard peak equalizer)
+ Low Shelf
+ High Shelf
+ Low Pass
+ High Pass
+ Band Pass
+ Notch
Q defines the bandwith or steepness of the selected filter.

Frequency [Hz]:
The center frequency of the EQ peaking filter, Notch or Band Pass as well as the 3dB cut-off frequency of the Low-/High-Shelf or
Low-/High-Pass filter.
Gain [dB]:
Amplification or attenuation for the selected filter.

Stereo Imager:

Stereo Imager bypass:

If the check box is selected then the audio signal bypasses the Stereo Imager.
If selected then the output signal gets converted into mono: (left + right)/2.0
Left & Right:
Sets the balance between the channels in dB.
Sets the level of the center (mono) signal.
Sets the level of the pure left and right signals.

Ambiophonics uses RACE* (Recursive Ambiophonic Crosstalk Elimination) to enable binaural listening. The RACE algorithm has
been made available by the Ambiophonics Institute (
The speakers need to be positioned with a separation angle smaller than 20 (called Ambiodipol) instead of 60 used in stereo. No
worries, the sound stage goes far beyond the boundaries set by the speakers (up to 120).

Ambiophonics bypass:
If the check box is selected then the audio signal bypasses the Ambiophonics encoder.
RACE Attenuation:
Increasing the attenuation value reduces the cross talk elimination which results in a smaller sound stage.
Center Presence:
A higher center value reduces the crosstalk elimination of "equiphase/similar amplitude" signals to emphasize center sounds like
The delay defines the time offset of the cross talk cancellation wavefront to make sure that the signal arrives the ear at the right
point in time. The amount depends on speaker separation and listening distance. It is a good starting point to go for a speaker
separation angle of 20 and a delay of around 70s. Please keep in mind that the delay is a discrete value which changes in steps
of "1/sampling frequency" (e.g 22.7s for 44.1kHz or 10.4s for 96kHz).

The gain slider reduces the amplitude (-dB) to compensate for the amplitude increase caused by the RACE algorithm.

Vintage Tube Warmer/Maximizer:

Vintage Tube bypass:

If the check box is selected then the audio signal bypasses the Vintage Tube Warmer/Maximizer.
A value of 0 adds no distortions whereas a value of 100 adds a maximum of k2 and k4 even harmonic distortions.
Adds harmonic distortions to create density and loudness.

The Spectrum of a 1kHz signal with its k2 = 2kHz and k4 = 4kHz distortions after passing the Warmer stage:

The Spectrum of a 1kHz signal after passing 100% Warmer and 100% Maximizer stages:


Compressor Bypass:
If the check box is selected then the audio signal bypasses the compressor.
Input Gain:
Adapting the amplitude of the compressors input signal.
Sets the level from where the signal gets compressed by applying attenuation defined by the ratio.
Compression factor (1:1 - 10:1).
The attack time range goes from 0 - 30ms to allow transients to pass the compressor without getting attenuated.
The time the compressor takes to go back to a gain value of one (0 - 2s).
Gain Reduction:
The indicator shows the amount of gain reduction for the chosen threshold, ratio and input gain.
Make Up Gain:
Compensates the gain reduction caused by the compression process.


Limiter Bypass:
If the check box is selected then the audio signal bypasses the limiter.
Input Gain:
Increase/Decrease the input gain until the right amount of limiting applies to the audio signal.
The maximum allowed signal level. It is good practice to choose at least -0.1dB instead of 0dB to provide some head room for later
audio compression (MP3, AAC, etc.).
The time the limiter takes to go back to a gain value of one (0 - 1s).
Gain Reduction:
The indicator shows the amount of gain reduction applied to keep the signal level below the selected threshold.

First Steps Tutorial:

1.) Go to the "Audio Input/Player/Output" tab
2.) Load an audio file of the supported format 16/24Bit wav with a sampling frequency of 44.1kHz, 48kHz, 96kHz or 192kHz
3.) Press "Play"
4.) Go to the Equalizer tab

a.) Deselect the "Equalizer Bypass"

b.) Select the Filter Type, Q, Freqency and Gain
c.) Attenuate exessive frequencies instead of amplifying weak frequencies
5.) Go to the Compressor tab
a.) Deselect the "Compressor Bypass"
b.) Set the Ratio, Threshold, Attack- and Release-Time as well as the Make Up Gain
c.) Go for a low compression to keep the natural dynamic
6.) Go to the Limiter tab
a.) Deselect the "Limiter Bypass"
b.) Set the limiter parameter to keep the limiting within acceptable values
7.) After setting up all components go back to the "Audio Input/Player/Output" tab
8.) Click "Render & Export Audio" to export the audio data to the selected file by applying the Audio Mastering Suite components

Disclaimer and License:

This Software is provided "as-is", without any express or implied warranty.
In no event will the author be held liable for any damages arising from the use of this software.
Permission is granted to anyone to use this software except for commercial use.
Please contact me (Stephan Hotto) if there is a wish for a commercial implementation.