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 AAMS IntroductionManual

AAMS Auto Audio Mastering System

As a musician or technician working on music sound material, you need the best sound possible when releasing material to the public. How do you know when audio material is equalized, compressed and maximized correctly and plays loud and evenly on all audio systems when it has been mastered ? To master a mix in general takes a lot of time and the procedure is mostly done after the mix is polished enough to go through the mastering process.  To make a master that sounds alike on all speaker systems and also sounds like a real professional commercial recording is a difficult and time consuming task.

This is were AAMS steps in and takes control!

AAMS is a Limited Freeware software package that provides suggestions for Equalizer, Multi-Band Compression and Loudness settings with internal DSP Processing to make all such audio corrections within the AAMS Program and creates a final mastered audio file. This makes the Audio Mastering Process easy and by far less time consuming and turns your mix into a great sounding commercial quality Audio Master.

Now you can listen to what you expect!

AAMS Auto Audio Mastering System is an all-in-one solution for mastering and a good learning tool for mixing and mastering audio in general. If you need a great sounding master with a minimum of effort, AAMS does automatically create a master out of your stereo mixes in minutes of time! With easy to learn automatic features the end result is a good sounding master fit for commercial release. AAMS V3 is a software program that inputs audio files, masters the audio file according to a reference style with its own processing tools and outputs the mastered audio file to disk. The easy part is that the user does not have to know about audio mastering but can master an audio file by just supplying a reference sound. With only a few button clicks, the user can master any audio file with AAMS!

The user interface is quite simple, once you understand the main functions of AAMS, it is allways very easy.

Your own music audio files are called ‘Source’.  As Source Material, these files you will deliver and import to AAMS as your own made music / audio / stereo mixes, these files are placed into a directory. The AAMS V3 database of reference styles is supplied with AAMS software to give the user a great starting point for a good sound. But also the user can create their own sound with reference styles or by analyzing and combining reference files and re-use them for audio mastering projects. You can copy and create styles from other sources like other artists, bands or any other audio material that compares to the sound you need! With the source file and reference audio file selected and loaded, AAMS will process the mastering for you automaticly. Your mastered audio is processed to a stereo WAV file inside the same directory.

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AAMSV30 AnalyzerFrequencySpectrum

Evaluate your Mix and Learn

AAMS is mainly a mastering tool, but the outcome (a mastered mix or track) can help you to listen and evaluate your mix better. This helps most users to understand their mix problems better and most users therefore can handle the mastering process better. You might go back to your mix and do single tracks adjustments and the re-evaluate again with AAMS.  The main focus of AAMS is automatic and easy, but AAMS can be setup for Full Automatic Mastering for Single Files, as well as Albums or Collections. AAMS also can be setup for Semi-Automatic Mastering and Manual Mastering. So for every user there is a way to master with AAMS. AAMS offers endless possibilities for analyzing, mixing and mastering audio, all functions are included and made simple with ease in mind. For the amateur the Automatic features are easy as clicking a few buttons. For the more professional users AAMS can be a helpful learning and viewing tool. Previously mastered material can be re-analyzed, viewed, processed and re-processed.

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How to get the sound you want and re-use it ?

With AAMS V3 there are several ways to get to your own sound, the sound you like! First we want to explain that with AAMS V3 standard references, you can get very close to a good general sound on your masterd audio file. But you need to EQ the outcome if you are not satisfied yet ? Well that is one possibillity. The other possibillities are explained below.

AAMS V3 uses Reference Files to define a complete frequency spectrum from 5 Hz to 20.000 Hz, it also contains compression, loudness scans and other sound analyzed settings found by AAMS analyzer. These files can be found in AAMS V3 Database of Styles (References). For example the ‘Mastering_RMS’ is a general reference preset for a beginning user to start from. This general mastering preset is a good point to start from, but however maybe will not result into what you want to hear. So that will fix a lot, maybe it will not get you exactly the sound with accuracy 100%. This is because maybe you use speakers or an environment room that is not linear and maybe you hear some more bass or less high ? Anyway, try to listen on as many speaker systems you can find, even headphones, even in your car or even somebody else audio setup. Before you decide what you want or need to change (into the frequency spectrum), to make you happy. Anyway, if you know you are right about the sound you want, how do you get there exactly ? With AAMS V3 it is now possible to get very close or exactly there! Read about the solutions below.
And watch the Video!

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Combining out of the Reference Database of Styles

Until now we have only used the ‘Mastering_RMS’ single reference preset out of the AAMS V3 Database of Musical Styles. But we can also combine several references out of the database and batch them into one new single reference. For this we only have to select multiple references with the 'CTRL' key pressed and use the batch function. See the blue selected references out of the database and press the green 'Batch Selected files into One Single Reference'. You can name the file and save it to disk with the filemanager that will pop up. Now the adjusted reference is saved, you can use it as your own adjusted reference, again and again. 

Analyze audio from other sources

One of the options is to analyze audio from other sources, like from other artists, tracks, albums. Collect  them in one single directory and analyze them with AAMS Analyzer (read the manual for more explenation). When you have analyzed them all, you can batch a new reference out of the collection of analyzed files.  AAMS will then batch a new reference out of all analyzed files and saves it to disk. The new reference is a good starting point and maybe is a better way to get the sound you need.

Adjust the Reference you already use or have chosen

With some testing and maybe working a longer time with AAMS V3, you maybe have some idea’s witch references you are using most, or witch reference are giving you a good sound from the beginning. Maybe you are using the ‘Mastering_RMS’ reference from the database ? Or maybe you have an own reference ? Anyway, the solution is you can always adjust the reference to your own needs and therefore be much closer to the sound you want.

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We have loaded the ‘Mastering_RMS’ reference out of the database as our tutorial reference. We have not loaded the source file. Off course you can load any reference file instead. See in yellow the name of the reference that has been loaded.

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On the Reference Adjust Tab we have changed the Spectrum for the Highs and some Mids with the curve and 4 points. In yellow above you can see the effect it has on the original spectrum, that results in raised frequencies in the upper bands.

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For more adjusting, we have created some extra bass frequencies in the spectrum around 80 hz and have a little cutoff in the beginning at 5-20 hz.

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Finally we can save the adjusted reference (see the green button) Save Adjusted Reference. You can name the file and save it to disk with the filemanager that will pop up. Now the adjusted reference is saved, you can use it as your own adjusted reference, again and again. This is a fast explenation of how to adjust the spectrum reference to your own needs and adjust it towards creating your own sound.

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Playing the Source Audio with the Adjusted Reference 

Have turned to the tab 'DSP-Player' that can play at any time the source audio material. Above we have loaded the source and reference (yellow buttons). When we press 'Play'(green) the source audio file will be played trough your computer soundcard. If you adjust the Reference like explained above, the changes can be directly heard, so with the Reference Centre or Left or Right Adjust Tabs you can change the reference spectrum while playing. You can hear the differences from your changes at any time, AAMS will adapt the sound with the DSP-EQ processor and updates directly. When you are satisfied, you can save the reference by 'Save Adjusted Reference' to disk. You can name the file and save it to disk with the filemanager that will pop up. Now the adjusted reference is saved, you can use it as your own adjusted reference, again and again.


Any combination of the above will work

If you are using AAMS V3 database of Reference Styles or you analyze your own tracks into AAMS. Or if you use the Reference Adjust Tabs, you can alther and change the way AAMS will master your own music, towards the Reference. Any combination will work and you can progress and build toward a final reference that suits your own musical style. You can even build your own library of references, it is up to you and your musical differences how many styles you need to cover your own music. We have only explained the basic steps to building your own sound reference files. And just to make it faster for mastering, once you have a good reference you like, it can be re-used again.

Now you can listen what you expect!

 

This video will show how to analyze multiple audio files with AAMS V3. And then create one single reference out of them, with a Batch Reference! And will show you how to adjust the Reference with the Analyzer Spectrum Adjust. For Centre Channels, Left or Right Channels. And save the Reference for later use. Therefore being in control of your own sound! And re-use it anytime you want.

Now you can listen what you expect!

AAMSV3DSPEQ

Equalizers Explained

An equaliser is designed to alter the tonal quality of audio passing through it, this by using a number of filter circuits. Filters are capable of applying gain to audio signals within specific frequency ranges, positive or negative, referred to a 'boost' or 'cut'. The more filters the better control over the whole frequency range in steps, but with crossovers in the filters there can be upcomming distortion. AAMS uses a linear spectrum equalizer with 100 filters, what basically means there is lot's of control and the sidewalls are so steap, the filtering does not affect each other. The quality of the AAMS DPS-EQ Spectrum Equalizer with 100 EQ Bands is very high and by our users atmitted as the one of the best EQ's around.

The AAMS internal DSP-EQ Player / Processor explained

The settings can be previewed and processed by the AAMS internal Players alike Analyzer and DSP-EQ, so you can hear the proposed changes. Processing is all done internally by aams, so an mastered outcome audio file is written to disk. The built-in DSP audio processing includes a 100-band DSP-EQ, 8-band DSP-Compressor and DSP-Loudness that can process audio in fully automatic, semi-automatic or manual modes. And AAMS will take care of all in between audio problems as clipping, dithering or normalizing. 

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A 100 Band Spectrum Equalizer, setup by AAMS suggestion calculattions.

The functions and processing of the DSP-EQ is not only doing EQ-ing, but in corporation with AAMS Analyzer will process first Spectrum and EQ results by calculations and processing made internally.  Mid and Side processing (m/s) is being applied as well as normal EQ processing. The possible 100 band EQ is defining a more spectrum wise way of using EQ. AAMS automatic suggestions will tell DSP-EQ how to behave. So that the best balance and spectrum is possible, AAMS will compare source and reference, making decisions what frequencies to adjust to have the same sound. This is not a copy of the reference, the source and reference are compared by AAMS Analyzer to get the best sound without just copying from the reference spectrum. 

AAMS 100 Band Graphic Equalizer (standard for AAMS Professional Version)
AAMS 50 Band Graphic Equalizer (locked standard for AAMS Freeware)
AAMS 25 Band Graphic Equalizer
AAMS 50 Band Graphic Equalizer
AAMS 15 Band Graphic Equalizer
Standard 31 Band Graphic Equalizer
Standard 61 Band Graphic Equalizer
Standard 15 Band Graphic Equalizer
Alesis 24 Band Graphic Equalizer
59 Bands Bottom End Graphic Equalizer
97 Bands Bottom End Graphic Equalizer
49 Bands Mid Lows Graphic Equalizer
41 Bands High Mids Graphic Equalizer
41 Bands Highs Graphic Equalizer
99 Bands Graphic Equalizer
Keys 86 Band Graphic Equalizer
Samsung 15 Band Graphic Equalizer
Firium 50 Band Graphic Equalizer
61 Band Graphic Equalizer
Sony Plugin 10 Band Graphic Equalizer
Sony Plugin 20 Band Graphic Equalizer
Winamp 11 Band Graphic Equalizer

AAMS calculates for the best possible settings.

The DSP-EQ takes these settings and will EQ spectrum the audio. Also DSP-EQ is accurate meaning straight and each Frequency Band is 41 steep. Next to each other they can form a 100 band EQ, and apply as much differences as possible, because of the higher band count. With the user functions to adjust source to reference and adjust the EQ, the user can make little corrections if needed. Select the DSP-EQ Tab, this represents a Graphic Equalizer that can have a single EQ band or up to 100 EQ Bands. The DSP-EQ also functions as a player for your audio material, just press Load Audio File or press Play and a window will pop up asking for a *.wav file to play. Press Stop to stop the player and press Pause to pause playing. On top of the faders is the Equalizer frequency band, lowering the faders below will change the gain of the EQ Frequency band in db. The first row of red Faders are adjusting all Left EQ frequencies and the second row of blue Faders are adjusting all Right EQ Frequencies. You can move the faders up or down and set all faders to 0dB with Zero All Faders, Zero Left, and Zero Right. If you already have loaded a Source and Reference, AAMS has calculated suggestions for the DSP-EQ and they can be copied with Copy EQ Suggestion. In this way you won’t have to set up the DSP-EQ yourself as AAMS does all the hard manual work for you automatically. Now you can listen to the differences that AAMS suggests. You can always turn OFF the EQ to hear the original audio material. You can also use Zero All Faders and Copy EQ suggestion to A/B your audio material.

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Playing and Equalizing with AAMS internal DSP-EQ processor

The file Example1.aam file is the Source file and is located in the same directory as Example1.afd. The Example1.aam file was created by AAMS analysis of the Example1.wav file located in the same directory. The Example1.wav file can be listened to using the DSP-EQ Player. Go to the DSP-EQ tab and press the button Play or Load Audio File and select the Example1.wav file. AAMS has already copied the Graphic EQ Suggestion into the faders of the DSP-EQ, so there is no need to set up the DSP-EQ. You can adjust the DSP-EQ Bands later on, if needed. What you hear now is the audio file playing with the DSP-EQ turned ON and the AAMS Suggestions applied. If you want to hear how the original audio material sounds like, you can Zero All Faders or Turn OFF the EQ. If you want to hear the suggestion again, turn ON the EQ and press Copy EQ Suggestion.

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DSP-EQ Real-time Views

The Real-time DSP views are preset in the DSP-EQ Tabs. When an audio file is loaded and playing these views will display and update the Spectrum, Spectrogram, Scope and Master-Out. All views as well as all level meters are updated with their own overflow LED's. When the signal is distorted by passing the highest possible maximum sound level an overflow LED will turn red. The overflow LED will return to green if all levels are returning to normal. The Hold LED’s will stay red until they are reset which can be done manually with a click of the mouse.

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DSP Overflow

When audio becomes too loud for the AAMS internal DSP-Processor an overflow is registered by one of the LED’s. AAMS will correct such overflows automatically when the player is playing, lowering the volume faders step by step. This is called AGC (Automatic Gain Control) and the DSP-EQ, DSP-Compressor and DSP-Loudness all use the AGC automatically to reduce the volume to a level that does not contain overflows. In this way you can be assured that levels will stay below 0dB and will not distort your sound. The Automatic Overflow feature can be turned off in the Options Tab, but this is not recommended.

DSP-EQ

All processing is done by AAMS V3 internal DSP-Processor, by the player and DSP pipeline processing. The DSP-EQ is quite an improvement of AAMS V3 and will allow you to hear your audio material directly inside AAMS without the use of other programs, you can also edit the DSP-EQ Bands Centre, Left and Right. Changing the Reference will update all settings and you can hear differences directly, choosing a reference to match the sound you want to hear. A good starting point is loading References from the Reference Database’s 200+ Styles. If in doubt, choose the style closest to your music and play that one first. Initially the Example file is intended to explain how AAMS works. The same example will be used to complete a full mastering job later on, but for now take the time to get familiar AAMS features and have some fun with it.

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DSP-EQ Centre Adjust

To have influence on the outcome of DSP-EQ Calculations between Source and Reference (Analyzer Suggestions) the DSP-EQ Centre Adjust presents an adjustable envelope chart for all DSP-EQ Bands to be edited upwards or downwards. By clicking into the envelope chart new envelope points are created and can be adjusted. The Left and Right EQ Bands (Centre) will be adjusted accordingly. The DSP-EQ Chart will be adjusted each time you change the envelope. For users who want to adjust the DSP-EQ Bands without changing the Reference, this can be a handy tool. You can off course load a corresponding audio file into the DSP-EQ Player and listen directly to the changes in audio sound. You can reset the Envelope pressing the Reset Button. Or clear the envelope by pressing the Clear Button.

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DSP-EQ Left Adjust

To have influence on the outcome of DSP-EQ Calculations between Source and Reference (Analyzer Suggestions) the DSP-EQ Left Adjust presents an adjustable envelope chart for all Left DSP-EQ Bands to be edited upwards or downwards. By clicking into the envelope chart new envelope points are created and can be adjusted. The Right EQ Bands will be adjusted accordingly. The DSP-EQ Chart will be adjusted each time you change the envelope. For users who want to adjust the DSP-EQ Bands without changing the Reference, this can be a handy tool. You can off course load a corresponding audio file into the DSP-EQ Player and listen directly to the changes in audio sound. You can reset the Envelope pressing the Reset Button. Or clear the envelope by pressing the Clear Button.

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DSP-EQ Right Adjust

To have influence on the outcome of DSP-EQ Calculations between Source and Reference (Analyzer Suggestions) the DSP-EQ Centre Adjust presents an adjustable envelope chart for all Right DSP-EQ Bands to be edited upwards or downwards. By clicking into the envelope chart new envelope points are created and can be adjusted. The Right EQ Bands will be adjusted accordingly. The DSP-EQ Chart will be adjusted each time you change the envelope. For users who want to adjust the DSP-EQ Bands without changing the Reference, this can be a handy tool. You can off course load a corresponding audio file into the DSP-EQ Player and listen directly to the changes in audio sound. You can reset the Envelope pressing the Reset Button. Or clear the envelope by pressing the Clear Button.

Auto Record DSP-EQ

This function is fully automatic. It has 2 Stage processing, first Hunting down the audio file for the suggested volume settings. This will prevent overflows that go over 0dB keeping the signal below 0dB while playing. The second stage is recording the audio file with the DSP-EQ. There is no need to set up the DSP-EQ manually as all settings are taken care of automatically as long as a valid Source and Reference are loaded into AAMS. The recorded audio file has the same name as the original audio file with the addition of the extension '_EQ.wav'.

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Overflow LED’s

The overflow LED’s turn from green to red when an overflow is present in one of AAMS DSP processors. You can press each LED and the Overflow Hold LED’s turn green again. An overflow is basically a digital domain term, when a signal goes over the 0dB limit an overflow is present. Recording in the digital domain with signals over +0dB will result in damaged audio with distortion therefore it is better to adjust the volume (gain) to a lower level. You can also adjust the faders of the Equalizer to compensate for overflows but this
is quite tedious.

Recording the DSP-EQ

After you are satisfied, you can process the audio material with the Record button. The Record button processes the audio and saves it as a new audio *.wav file. The Example file Example1.wav would be saved as Example1_EQ.wav.

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Settings

The DSP-EQ can handle from 1 to 100 Band Graphic EQ settings. The Default is 100 Band Graphic EQ which is a very good setting for all applications. Whenever you need a bit more detail, you can use the 100 Band Graphic EQ setting. Although the 100 Band Graphic EQ will give more detail this will affect processing speed. As a general rule using more EQ Bands can bring phasing into the sound. This will depend on frequency overflow in the audio material and the way a Graphic EQ does its job. Whenever you hear Phasing starting to begin, switchback to a lower DSP-EQ Setting. Most likely Phasing sometimes will only happen using the 100 Band EQ Setting. However 95% of all audio material does not introduce Phasing when using the 100 Band EQ Settings. If you better be on the safe side and in general a 50 0r 32 Band Graphic EQ is a very good default.

Manual and Automatic Mode

Because AAMS is usually set up for Automatic Mastering at installation, if you wish to work manually with the AAMS Processors it is recommended that you use the Record button of each processor. Also for configuring AAMS Options for working in manual mode, there are some settings on the Options Preferences Tab. For all Automatic Mastering purposes, switch back to Automatic Mode in Options Preferences Tab.

Specifications

The internal DSP-EQ has a natural sounding algorithm that equalizes in an exact linear manner with no resonance peaks. The frequency range is 5Hz-22050Hz. The amount of Graphic Equalizer Bands range from 1 band to 100 with 'Natural' and 'Steep' filters. Depending on the amount of EQ-Bands and the Frequency range calculated by AAMS Suggestions, the frequency range 0-5 Hz or 20000-22050 Hz is rolled-off and all Factory Presets have a Roll-On and Roll-Off Frequency range. Gain ranges from -12dB to +12dB, and can be set in fractions of 0.1dB. The Q-factor of each filter is displayed for convenience and changes depending on the amount of Graphic Equalizer filters that are in use. Designed to operate at sampling rates ranging from 44.1 KHz to 192 KHz, although the Sample rate is normally 44.1 KHz, depending on the Audio File outputted by the Player. 
The Internal DSP-Processing can handle 16/24/32 Bit calculations while the rest of the programming handles 64-bit code. The DSP-EQ allows comparison of the AAMS EQ Suggestions and the Original Audio.

AAMSV3DSPCompressor

A multiband compressor is to compress several different frequency ranges.

Compression is one of the most useful tools and is widely used in the recording process from tracking to mixing to mastering. The majority of compressors act on the entire audio frequency range, but when we need more control over specific audio frequencies we use a multiband compresser. A multiband compressor is a collection of several individual compressors, typically 3, 4, 5 or 8 multibands. Dividing up the frequency range via crossovers per multiband.

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AAMS DSP-Multiband Compressor

A 8 Band Soft Tube Multiband Compressor, setup by AAMS and by compressing automatic on the fly! The AAMS multiband compressor does setup and selfmaintains the compression levels, because it is setup by AAMS Analyzers and hunts for its own compression levels. AAMS DSP Compressor constantly is active to see and hear the signals it is compressing and adjusting to the perfect levels. No more under- or overcompression!

8 Multibands

! Soft tube behavior
! Fully Automatic Compressor

Multiband compressor with automatic gain or limiter functions, 8 multibands possible. Soft tube behavior, not curved but also not straight lined soft tube compressor for each band. Automatic hunting function for calculation of threshold levels per band and automatic gaining functions. AAMS Compressors are setup by AAMS Analyzer calculations in suggestions, these are applied by the automatic DSP-Compressor processing on the audio. A compressor reduces the dynamic range of an audio signal if its amplitude exceeds a specified threshold. The amount of gain reduction is determined by Attack, Delay, Threshold and Ratio settings. A Multiband Compressor has more Bands and each frequency Range (Band) can be separately compressed. A Single Band Compressor has basically no use for Mastering purposes, instead a Multiband Compressor can be highly recommended for Mastering Purposes. AAMS Compressor and multibands are based on compressor functions as for mastering is known. 

The main function is balancing the audio track after DSP-EQ has done its spectral functions.

The DSP-Compressor is based on spectrum frequency as well as on dynamics and dynamic balancing. Mostly complex to setup by user or manual compressors. Maybe also because compressor as a subject is less easy to understand and maintained as EQ does. Compression in mastering is more subtle. It tries to EQ a bit and Loudness a bit. The DSP-EQ in front and the DSP-Loudness afterwards, they are good friends in AAMS Mastering chain.

The AAMS internal DSP-Compressor Player / Processor explained

Select the DSP-Compressor Tab. This represents a Multi-Band Compressor that can have a single band compressor simulated toward 8 Multi-Bands. The DSP-Compressor functions as a player for your audio material, just press Load Audio File or press Play and a window will pop up asking for a *.wav file to play. Press Stop to stop the player and press Pause to pause playing. If you have already loaded a Source and Reference, AAMS has calculated suggestions for the DSP-Compressor and they can be copied with Copy EQ Suggestion. In this way you won’t have to set up the DSP-Compressor yourself as AAMS does all the hard manual work for you automatically. You can also use Zero All Faders and Copy EQ suggestion to A/B your audio material.

DSPplayer

Playing and with AAMS internal DSP-Compressor processor

The file Example1.aam file is the Source file and is located in the same directory as Example1.afd. The Example1.aam file was created by AAMS analysis of the Example1.wav file located in the same directory. The Example1.wav file can be listened to using the DSP-Compressor Player. Go to the DSP-Compressor tab and press the button Play or Load Audio File and select the Example1.wav file. 

The DSP-Compressor

Depending on the Setup chosen from the Settings Tab, a single band compressor up to an eight band Multi-Band compressor can be simulated. Each Multi-Band compressor has its own settings for Threshold, Attack, Decay, Ratio and Window. The Threshold fader of each Multi-Band can be set manually and when you click on the value below the Threshold Fader it will reset to 0dB. The DSP-Compressor has got a natural sound and is there to compress the Multi-Band as natural as possible, meaning there are no extra functions like EQ or Exciters used. 

Auto Record DSP-Compressor

This function is fully automatic and has two processing stages. The first searches the audio file for the suggested compressor settings. The second stage recordings the audio file with the correct threshold settings found by the AGC. There is no need to set up the DSP-Compressor manually as all settings are taken care of automatically as long as a valid Source and Reference are loaded into AAMS. The recorded audio file has the same name as the original audio file with the addition of the extension '_C.wav'.

Offline AGC Hunting

This function is the first stage and scans the audio file for the correct AGC settings as suggested by AAMS calculations. The outcome is shown by the 'Suggested Threshold Targets' and is automatically copied to the threshold faders of each Multi-Band. After this function you can press Record to proceed to the Second stage.

The Faders

The threshold fader for left and right can be set manually or by choosing a setup. When you click on the value below the fader will reset to 0dB. The Attack, Delay, Ratio and Window settings are preset by the Setup you have chosen but can be changed manually.

The AGC Control

AGC Control is the automatic feature that runs the DSP-Compressor. When the AGC is turned ON, the DSP-Compressor will automatically hunt the audio that is being played or processed for the correct levels. These levels are based on the Compressor Suggestion AAMS has calculated. When the AGC is turned On the DSP-Compressor is hunting for these levels and the AGC Graph and AGC Data displays the results while the audio file is being played or processed. In the AGC Data grid you can see all levels and the suggestion in db. The 1-8 LED’s below show the correct found levels when they turn to Green, indicating the Multi-Band Left or Right threshold level has been found by the AGC. It is possible that Green LED’s turn off and on for a while as the AGC is waiting for the audio signal to be corrected. When all Left and Right LED’s are green a timer will run and after some time the AGC Correct LED will also show green meaning all Multi-Bands are now ok. All correct Multi-Band threshold levels have been found based on the AAMS Compressor suggestion. The suggested Threshold level of every Multi-Band is shown as 'Threshold Suggestion Left' and 'Threshold Suggestion Right'. It is best to turn AGC ON when playing and let the AGC Control hunt down the right threshold levels, playing through the whole file. Then turn the AGC off and set 'All targets threshold' and Record the Audio File (when you press the Record button the AGC is automatically turned to off).

The AGC Chart

The AGC Chart shows the selected Multi-Band and shows the real time audio compressed signal in red and blue lines for Left and Right. The maroon and dark blue lines are the AC controls signal that will try to stay close to the yellow and green Suggestion lines. The nearer that the maroon and dark blue lines come to the yellow and green Suggestion lines, the better the result. The AC will compensate with the Left and Right threshold faders of each Multi-Band and will show the 'Suggested Threshold' as a result while playing the audio signal. You can reset the AC chart and found levels with the buttons besides the chart.

The AC Data

The AC Data grid will show all data that has been measured for each Multi-Band, including the suggestions for each Multi-Band as calculated by AAMS. When the AC is “Hunting” a correct threshold level is not found yet. When the AC is “Ok” the AC has found correct levels. The LED’s below the AC will turn to green for each found Multi-Band level that is correct.

Filters

Each set up of the DSP-Compressor consists of one or several Multi-Bands each with a specific frequency range. The frequency ranges for each Multi-Band are shown in the Filters Chart.

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ANARealtime ANASpectrogram

DSP-Compressor Real-time Views

The Real-time DSP views are preset in the DSP-Compressor Tabs. When an audio file is loaded and playing these views will display and update the Spectrum, Spectrogram, Scope and Master-Out. All views as well as all level meters are updated with their own overflow LED's. When the signal is distorted by passing the highest possible maximum sound level an overflow LED will turn red. The overflow LED will return to green if all levels are returning to normal. The Hold LED’s will stay red until they are reset which can be done manually with a click of the mouse.

Settings

Here you can set up a DSP-Compressor from several useful presets. The AAMS 1-8 Multi-Band Compressor settings are suitable for most purposes. The AAMS 1-8 Low Band Multi-Band Compressor settings are for lower quality recordings like MP3 and Tape Copies and these settings will concentrate more on the lower rather than high frequencies. Furthermore there are some settings that will help you work with plugins using the correct settings. In a normal situation a 4 Band Multi-Band Compressor is the default setting. Changing to an 8 Multi-Band Compressor will affect processor speed and will improve quality a little more. Use the 8 Multi-Band Compressor when you think you need the extra Multi-Bands, otherwise the 4 Band Multi-Band Compressor setting is default. Selecting a Preset will change the setup of the DSPCompressor and when audio is being played the player will stopped...

The Volume Faders and Level Meters

When the DSP-Compressor is playing the volume faders are automatically lowered when an overload is present on the master-out. To be sure that there are no overloads you should listen to the whole audio material until the end, then the volume faders are set just right and the master-out will not go over 0dB. You can turn off this function in the Options Tab.

AAMSdspCoverflow

Overflow LED’s

The overflow LED’s turn from green to red when an overflow is present in one of AAMS DSP processors.You can press each LED and the Overflow Hold LED’s turn to green again. An overflow is basically a digital domain term when a signal goes over the 0dB limit. Recording in the digital domain with signals over +0dB will result in damaged, distorted audio. So it is better to adjust the volume (gain) to a lower level. You can also adjust the faders of the Equalizer to compensate for overflows, but this is quite tedious.

Recording the DSP-Compressor

After you are satisfied with the results you can process the audio material with the Record button. The Record button processes the audio and saves it as a new audio *.wav file. The Example file Example1.wav would be saved as Example1_C.wav.

AAMSdspComp8b

General Rules

The default 4 Band Compressor setting is the basic set up for compression. This will give good results on all kinds of processor speeds and available memory. When you are using a modern computer you can switch to the 6 or 8 Multi-Band Compressor for some slightly better results. 8 Multi-Bands almost halves the speed of operation, so switch back to a 6 or 4 Band Multi-Band Compressor if you experience problems. For best results it is also important that all AC LED’s are green, confirming that good compression levels have been found. When using Automatic Mastering there is a Mastering Document saved alongside the Mastered Audio file, here you can check that the right levels are found (AC Correct).

Manual and Automatic Mode

Because AAMS is set up for Automatic Mastering by default, when working manually with AAMS Processors it is recommended that you use the Record button of each processor. Also for configuring AAMS Options for working in manual mode, there are some settings on the Options Preferences Tab. For all Automatic Mastering purposes, switch back to Automatic Mode in Options Preferences Tab.

Specifications

The internal DSP-Compressor has a natural sounding algorithm that provides linear compression. The DSPCompressor AC Control can handle AAMS Compressor Suggestions automatically or manually. The frequency range is 5Hz to 22050 Hz. Compressor Bands range from 1 to 8 depending on the Factory Preset in use. The frequency range 0-5 Hz or 20000-22050 Hz is rolled-off and all Factory Presets have a Roll-On and Roll-Off Frequency range. The application is designed to operate at sampling rates ranging from 44.1 kHz to 192 kHz. The Sample rate is normally 44.1 kHz depending on the Audio File output by the Player. The Internal DSP-Processing can handle 16/24/32 Bit calculations while the rest of the programming handles 64 bit code.

AAMSV3Loudness

Today, most spoken audio issue is control of loudness, daily millions of people adjust their volume controls because of louder or softher audio signals. Modern Music recordings are louder then recordings from the past, promos and commercials are generally much louder than normal. Even the difference by playing recordings, will make you adjust the volume. In general more loudness means we are in the Loudness War, each time upping the loudness levels of recordings. In fact we will loose loudness and the sparkle of the recording (mix) when boost too much, so everything is compressed hard at the tops to 0 db! Speakers need to move, and what happens when we want more loudness ? They actually move less! In general the music industry is still upping the levels, but there is a great understanding now that this is only hurting the actual recording. Loudness is a balance between hearing it loud and hearing soft and hard clearly. So we need to have the loudness levels tamed! AAMS Loudness is according to the setup rules!

Loudness and Limiters

Loudness is the quality of a sound that is the primary psychological correlate of physical intensity. Loudness, a subjective measure, is often confused with objective measures of sound intensity such as decibels. Filters attempt to adjust intensity measurements to correspond to loudness as perceived by the average human. However, true perceived loudness varies from person to person and cannot be measured this way. Loudness is also affected by parameters other than intensity, including: frequency and duration. A Peak or RMS Limiter is a circuit that allows signals below a set value to pass unaffected and compresses the peaks of stronger signals that exceed this set value. A Brick wall Limiter is a circuit that allows signals below a set value to pass unaffected and cutoff the peaks of stronger signals that exceed this set value and to avoid clipping at 0db or above 0db.

The AAMS internal DSP-Loudness Processor explained

AAMS loudness is based on AAMS Analyzer suggestions and a pre-setup desired level. Select the DSP-Loudness Tab and Select Loudness. The DSP-Loudness Player and Processor contain a Balancing Tool, Loudness Maximizer, RMS Limiter, Peak Limiter and Brick wall Limiter in one.  Also the loudness DSP has analog properties, meaning the compressor curves are soft and round.  Main functions of the DSP-Loudness processor is RMS Audio Scanning, Peak Audio Scanning, Balance Search that balances Left and Right of the whole track.  A Gain processor, that functions on AAMS Peak RMS based system or the user can change to RMS more Dynamic based system.  In whole the DSP-Loudness Mastering routine that does DSP-Loudness functions automatically or user based. The DSP-Loudness also functions as a player for your audio material. Simply press Load Audio File or press Play and a window will pop up asking for a *.wav file. Press Stop to stop the player and press Pause to pause playing.

Chart - Display Levels

When an audio file is loaded into the player the file will be automatically scanned for its levels. These levels are shown on the chart and above the chart are the Peak / RMS levels. After scanning is finished the suggestions for Balancing and Gaining are shown below the chart. The basic intention of the DSP-Loudness is to balance then add gain to the audio file. The Balance suggestion shows a value in dB, a minus - value is balanced to the Left and a plus + value is balanced to the Right. You can set the Balance Faders (There are three in total connected to each other) with the mouse. You can reset the Balance Faders by clicking on the value below the fader and the fader will reset to 0dB. The red LED’s for Balance and Gain will show up when suggestions matching the Audio File are correct.

Auto Record Loudness

This function will automatically process Balance and Gaining onto the audio file you load into the DSP-Loudness player/processor. A balanced audio file with the extension '_B.wav' and a gain processed audio file with the extension '_L.wav' will be created. All you have to do is wait until processing is finished. This function is fully automatic, just choose an audio file and AAMS will do the balancing and gaining for you. There is no need to set up the DSP-Loudness functions before using this function as all calculations are done automatically. This function is a two-stage process with Balancing as the first stage and gaining the second stage.

Auto Record Balance

This function is the first stage of the DSP-Loudness processor and balances the audio file you loaded into the Player/Processor. A recorded and processed file with the extension '_B.wav' will be created as the balanced audio file. This function is fully automatic, just choose an audio file and AAMS will do the balancing. There is no need to set up the DSP-Loudness functions before using this function as all calculations are done automatically.

Auto Record Gain

This function is the second stage of the DSP-Loudness processor and gains the audio file you loaded into the Player/Processor. A recorded and processed file with the extension '_L.wav' will be created as the gained audio file. This function is fully automatic, just choose an audio file and AAMS will automatically do the Loudness Gaining. There is no need to set up the DSP-Loudness functions before using this function as all calculations are done automatically.

1 or 2 Stages?

You can use 1 or 2 Stage functions with AAMS. A 1 Stage manual job involves playing the audio file with the Player and setting up Balance and Gain, ensuring that the signal does not go over 0dB, and then record the results. A 1 Stage processing job is less accurate than the 2 Stage type. A 2 Stage Processing job is more accurate and means you Balance the Audio File First Stage 1, record it, and then do Stage 2.

The internal Processing Route

The routing of the DSP-Processor is as follows. The Audio is processed by the Loudness part of the DSPLoudness processor. The Loudness Tab shows Balancing Faders and Loudness Gain Volume faders. The RMS Limiter is there to compress/limit the audio signal when necessary. The RMS limiter is based on an RMS compressor and has as large a ratio range as possible. The Peak Limiter limits some the highest peaks and can be set towards Brick walling or the more moderate Peak Limiting. Finally the Brick wall Limiter limits any signals below 0dB or lower. All Limiters can be turned off or on by depending on what set up from the Setting Tab you have chosen.

DSPplayer

Playing and with AAMS internal DSP-Compressor processor

The file Example1.aam file is the Source file and is located in the same directory as Example1.afd. The Example1.aam file was created by AAMS analysis of the Example1.wav file located in the same directory. The Example1.wav file can be listened to using the DSP-Compressor Player. Go to the DSP-Compressor tab and press the button Play or Load Audio File and select the Example1.wav file. 

The Setups

The Settings Tab will show the DSP-Loudness setup and you can choose any of the listed DSP-Loudness settings. The Maximal Setups are for Maximal Loudness and will try to get the loudest sound possible and is the default preset. The Average Setups are for Average Loudness and will try to get the best sound possible. The Minimal Setups are for Minimal Loudness and will try to get the sound towards a minimal level. This is useful for compiling CD's or multiple tracks and saves some loudness space for later use. The values for the RMS Limiter, Peak Limiter and Brick wall limiter are also listed. When a Limiter is not shown in the chosen setup this means that it is not required in the current configuration. When an Audio File is loaded into the DSP-Loudness processor / Player, the levels in the file are scanned. You can change the Scan Level in the Settings Tab, which will raise or decrease the Loudness Level that is being scanned for (and are subsequently used in the suggestions. When the correct level is reached for the Maximal 0dB Setup the Suggestion Correct LED’s will show Green. When the correct level is reached, for the other Setups the Suggestion Correct LED’s will show Green or Yellow. The Yellow Led indicates that the Used Setup highest level has not been reached yet as only the Maximal 0dB Setup will do this. You can use the Average and Minimal Setups for a single stage pass. When you apply the Average and Minimal Setups more than once the Yellow LED will stay and will turn green only when it reaches the Maximum Setup Loudness Levels. When you use the Average or Minimal DSP-Loudness Setups the first pass of DSP-Loudness Processing is sufficient, so making multiple passes is not recommended. You can always use more Loudness Gaining to make it louder although this is also not recommended. When you use the Maximal DSP-Loudness Setup, the first pass will gain the loudness directly to the Maximal and then the Green Led will show-up. On start up the Average 1 Setup is the default to ensure a good Average Loudness. When you change the DSPCompressor Setup to Maximal, be aware that some distortion in the audio signal cannot be prevented. Only use the Maximal Setup when you are certain ofof it and the intended result. If you are unsure, a good start is the Average Setups of the DSP-Compressor.

Loudness

The Balance Fader can be set manually and clicking on the value below the fader will reset it to 0dB. When you use 'Copy Balance Suggestion' the suggested Balance is copied to the Balance Fader. The Gain fader can be set manually and clicking on the value below the Gain Fader will reset it to 0dB. When you use 'Copy Gain Suggestion' the suggested gain is copied to the Gain fader. The buttons 'DSP-Loudness Preset' and 'DSP-Loudness Reset' are setting or resetting the chosen Setup, which is useful for comparing its effect.

AAMSV3LoudnessRMSlimiter

RMS Limiter

The threshold fader for left and right can be set manually or by choosing a setup and when you click on the value below the fader it will reset to 0dB. The Attack, Delay, Ratio and Window settings are preset by the Setup you have chosen but can be changed manually. The RMS Limiter is not as fast in correcting the audio signal as the Peak Limiter is. Most of the times when threshold levels for left and right are low, the RMS Limiter will only work on loud parts of the audio signal, making it slightly lower when needed. The RMS Limiter will pass signals higher than 0dB to the Peak Limiter and is only meant for moderately correcting the audio signal.

AAMSV3LoudnessPeakLimiter

The Peak Limiter

The threshold fader for left and right can be set manually or by choosing a setup and when you click on the value below the fader it will reset to 0dB. The Attack, Delay, Ratio and Window settings are preset by the Setup you have chosen but can be changed manually. The Peak Limiter is very fast in correcting the audio signal and can also be used as a Brick wall Limiter when the attack time is set to <= 1ms. The Peak Limiter will adjust every signal that goes above the threshold level with the ratio chosen. A ratio of 50:1 will compress / limit the audio signal a lot more than the RMS Limiter will ever do. The Peak Limiter is meant to 'Scrape off' some of the loudest peaks of the audio signal.

AAMSV3LoudnessBrickwall

The Brick wall Limiter

The threshold fader for left / right and mono can be set manually or by choosing a setup and when you click on the value below the fader it will reset to 0dB. Every audio signal that goes over the Threshold Level will be cut off immediately. A Threshold Level of -0.05 dB will cut off all audio signals before it rises above 0dB. The LED’s indicate only if the Brick wall limiter is correcting the audio signal, and are not the same as an overflow. The LED’s can be reset by clicking the corresponding button.

The Volume Faders and Level Meters

When the DSP-Loudness is playing and Gaining / limiting the volume faders are automatically lowered when an overload is present on the master-out. To be sure that there are no overloads you should listen to the whole file until the end, then the volume faders are set just right and the master-out will not go over 0dB. You can turn off this function in the Options Tab.

AAMSdspLoverflow

Overflow LED’s

The overflow LED’s turn from green to red when an overflow is present in one of AAMS DSP processors. Press each LED and the Overflow Hold LED’s turn green again. An overflow is basically a digital domain term. When a signal goes over the 0dB limit an overflow is present. Recording in the digital domain with signals over +0dB will result in distorted audio, so it is better to adjust the volume (gain) to a lower level. You can also adjust the faders of the Equalizer to compensate for overflows, but this is quite tedious.

Recording DSP-Loudness

When you are satisfied with the results you can process the audio material with the Record button. The Record button processes the audio and saves it as a new audio *.wav file. The Example file Example1.wav would be saved as Example1_L.wav.

ANARealtime ANASpectrogram

DSP-Loudness Real-time Views

The Real-time DSP views are preset in the DSP-Loudness Tabs. When an audio file is loaded and playing these views will display and update the Spectrum, Spectrogram, Scope and Master-Out. All views as well as all level meters are updated with their own overflow LED's. When the signal is distorted by passing the highest possible maximum sound level an overflow LED will turn red. The overflow LED will return to green if all levels are returning to normal. The Hold LED’s will stay red until they are reset which can be done manually with a click of the mouse.

Manual and Automatic Mode

Because AAMS is set up for Automatic Mastering by default, when working manually with AAMS Processors it is recommended that you use the Record button of each processor. Also when configuring AAMS Options for working in manual mode there are some settings on the Options Preferences Tab. For all Automatic Mastering purposes, switch back to Automatic Mode in Options Preferences Tab.

General Rules

DSP-Loudness Processor settings define how loud the end result will be. By default the Loudness Scan setting is 'Average', which is quite a conservative setting. Usually you can master multiple tracks (for example for an album of tracks) with the 'Average' Setting and this will give good results on all kinds of tracks. Whenever you need some more loudness power, set the DSP-Loudness Settings higher.

Specifications

The internal DSP-Loudness has a natural sounding algorithm to produce exact, linear Loudness Gain with no resonance peaks. The DSP-Loudness processor has several main functions, Balancing, Loudness, RMS Limiter, Peak Limiter, Brick wall Limiter. The frequency range is 5Hz to 22050 Hz. The frequency ranges 0-5 Hz and 20000-22050 Hz are rolled-off and all Factory Presets have a Roll-On and Roll-Off Frequency range. The amount of gain ranges from - 80dB to +9dB, and can be set in fractions of 0.1dB. Designed to operate at sampling rates ranging from 44.1 KHz to 192 KHz, the Sample rate is normally 44.1 KHz depending on the Audio File output by the Player. The Internal DSP-Processing can handle 16/24/32 Bit calculations while the rest of the programming handles 64 bit code.

AAMS Loudness System

The AAMS Loudness system is based on a percentage of peaks and RMS levels or RMS/Peak Levels. The AAMS system will do a more peak based search of the audio and also scans for RMS levels. But however AAMS system will calculate and process audio as being more careful to avoid distortion levels. This means that for most it will bring your audio input to appropriate levels. Not doing the Loudness War, but being careful and still have a loud sound. AAMS DSP-Loudness process will anticipate troubles and avoids them. Still Loudness that is applied to much, will still have distortion or overflows. So limiters and calculations in processing are used in the AAMS Loudness Based system option. When needed this system can do full album mastering based on AAMS System of transferring loudness levels. Also on its main settings Single Audio Mastering can be an advantage over the RMS Dynamic Levels system.

RMS Dynamic Level System

The RMS Dynamic Level system is based on more know RMS and Peak scanning methods. If you are into RMS levels, there is much to say about it. Because it is the main system used by everyone into the discussion of the loudness war. So read about that on the internet if you are unknown. 
The main function is RMS and Dynamics / some say Dynamic Range.

Why use a Loudness Processor and what is it doing?

Well the user can setup a desired RMS level for the whole track. The peaks will be calculated and shaved off the whole track, so that the RMS Level is what is needed as loudness or RMS levels for the whole track. Basically the user setup is for RMS Levels, and the Limiters or Calculations a computer can do to process the audio that loud. For different genres there are RMS levels appropriate. So the user can make up the RMS, and does not really care about peaks that will be hurt. RMS Levels are nowadays convenient method to get the loudness desired. With shaving the peaks and making things louder, comes at a price. Distortion and Overflows. We hope that the Limiters and internal processing will do not do too much damage. If the user sets the RMS Levels too high (-6dB to 0db!) be sure of artifacts inside the audio.

You can select RMS levels and Dbfs levels.

For single files and genres of music, when mastering to a certain level is needed. Dbfs system is the same as RMS system but it relies on other measurement system called dbfs. Both RMS and dbfs systems are good when you are wanting to do things manually and adjust the loudness. When mastering or even after mastering. You can hunt down your appropriate levels and setup AAMS DSP-Loudness to follow. That must be digital distortion? Both systems AAMS system and the RMS System are not opponents, but friends!

AAMSdspLsettings

AAMS Loudness System

AAMS Sub Zero 0: +6.00 dB
AAMS Sub Zero 1: +5.75 dB
AAMS Sub Zero 2: +5.50 dB
AAMS Sub Zero 3: +5.25 dB
AAMS Sub Zero 4: +5.00 dB
AAMS Sub Zero 5: +4.75 dB
AAMS Sub Zero 6: +4.50 dB
AAMS Sub Zero 7: +4.25 dB
AAMS Loudness War I 1: +4.00 dB
AAMS Loudness War I 2: +3.75 dB
AAMS Loudness War I 3: +3.50 dB
AAMS Loudness War I 4: +3.25 dB
AAMS Loudness War II 1: +3.00 dB
AAMS Loudness War II 2: +2.75 dB
AAMS Loudness War II 3: +2.50 dB
AAMS Loudness War II 4: +2.25 dB
AAMS Increased 1: +2.00 dB
AAMS Increased 2: +1.75 dB
AAMS Increased 3: +1.50 dB
AAMS Increased 4: +1.25 dB
AAMS over the Top 1: +1.00 dB
AAMS Over the Top 2: +0.75 dB
AAMS Over the Top 3: +0.50 dB
AAMS Over the Top 4: +0.25 dB
AAMS Ultra Hard Loudness: 0.00 dB
AAMS Hard Loudness: -0.25 dB
AAMS Super Strong Loudness: -0.50 dB
AAMS Strong Loudness: -0.75 dB
AAMS Good Loudness: -1.00 dB
AAMS Average Loudness: -1.25 dB - Normal levels AAMS target setting
AAMS Medium Loudness 1: -1.50 dB
AAMS Medium Loudness 2: -1.75 dB
AAMS Soft Low ends 1: -2.00 dB
AAMS Soft Loudness 2: -2.25 dB
AAMS Uttara Soft Loudness 1: -2.50 dB
AAMS sutra Soft Loudness 2: -2.75 dB
AAMS Headroom Loudness 1: - 3.00 dB
AAMS Headroom Loudness 2: - 3.25 dB
AAMS Headroom Loudness 3: - 3.50 dB
AAMS Headroom Loudness 4: - 3.75 dB
AAMS Minimal Loudness 1: - 4.00 dB
AAMS Minimal Loudness 2: - 4.25 dB
AAMS Minimal Loudness 3: - 4.50 dB
AAMS Minimal Loudness 4: - 4.75 dB
AAMS Minimal Loudness 5: - 5.00 dB
AAMS Minimal Loudness 6: - 5.25 dB
AAMS Minimal Loudness 7: - 5.50 dB
AAMS Minimal Loudness 8: - 5.75 dB
AAMS Minimal Loudness 9: - 6.00 dB

AAMSdspLsettings

RMS Dynamic Loudness System

RMS Range Ultra Stupid: 0 dB
RMS Range Ultra Stupid: -2 dB
RMS Range Ultra Stupid: -3 dB
RMS Range Ultra Hard: -4 dB
RMS Range Ultra Hard: -5 dB
RMS Range Ultra: -6 dB
RMS Range Ultra: -7 dB
RMS Range Loud: -8 dB
RMS Range Loud: -9 dB
RMS Range Normal: -10 dB – Normal levels target setting
RMS Range Normal: -11 dB
RMS Range Normal: -12 dB
RMS Range Normal: -13 dB
RMS Range Normal: -14 dB
RMS Range Normal: -15 dB
RMS Range Normal: -16 dB
RMS Range Low: -17 dB
RMS Range Low: -18 dB
RMS Range Low: -19 dB
RMS Range Low: -20 dB
Dbfs Range Ultra Stupid: 0 dB
Dbfs Range Ultra Stupid: -2 dB
Dbfs Range Ultra Stupid: -3 dB
Dbfs Range Ultra Hard: -4 dB
Dbfs Range Ultra Hard: -5 dB
Dbfs Range Ultra: -6 dB
Dbfs Range Ultra: -7 dB
Dbfs Range Loud: -8 dB
Dbfs Range Loud: -9 dB
Dbfs Range Normal: -10 dB – Normal levels target setting
Dbfs Range Normal: -11 dB
Dbfs Range Normal: -12 dB
Dbfs Range Normal: -13 dB
Dbfs Range Normal: -14 dB
Dbfs Range Normal: -15 dB
Dbfs Range Normal: -16 dB
Dbfs Range Low: -17 dB
Dbfs Range Low: -18 dB
Dbfs Range Low: -19 dB
Dbfs Range Low: -20 dB

AAMSV3Tools

AAMS Tools Tab

Audio file convetions and multiple dithering in 64bit calculations. Mostly understand AAMS will work best with WAV 32 or 16 Bit Stereo 44.1 KHz Files. You can import MP3 and WAV files.  When AAMS has a problem with these files, use the tools tab. Or convert your files with a convertor program / software, audio editor of your choose.

Normalize Audio File

Normalize an audio file to 0dB. Select the file format, click on Normalize Audio File. Select the input audio file that will be normalized. Select the output audio file that will be saved.

Convert MP3 Audio File

Convert an audio file to Wav. Select the file format, click on Convert MP3 Audio File. Select the input MP3 file that will be converted. Select the output audio file that will be saved.

Convert Wav to MP3 Audio File

Converts Wave Format files to MP3 audio files. Select the file format, click on Convert Wav Audio File. Select the input Wav file that will be converted. Select the output audio file that will be saved.

Wav Bitrate Converter

Select the file format, click on Convert Wav Audio File. Select the input Wav file that will be converted. Select the output audio file that will be saved.

AudioFormats

AAMSV3main

To make full use of AAMS V3, you must register for AAMS V3 Full Version. The AAMS software program, website and this manual gives information how to do so. AAMS Limited Freeware Version can be freely used. This program is Free Distributable, i.e. you can evaluate this functional Limited Freeware Version. Distribution of AAMS installer and AAMS software can be done on a freeware basis, take into account that asking money or payments for the AAMS software is prohibited and not legal. AAMS is distributed on NON-COMMERIAL basis, so if you paid money for AAMS, ask your money back from the distributor. When you commercially distribute AAMS Software or AAMS Coding or AAMS Install files or any reference that suggests commercial use to make money out of AAMS, you must stop your activities!

AAMS Auto Audio Mastering System - Is Limited Freeware!

This software package is available free of charge. You can show your appreciation and support future development by donating on existing and upcoming products. When you are using this software for a longer time or using this software for commercial use, to earn money, you must think about giving a part of it to the author and register AAMS for a Full Version License.

Thank You!

Denis van der Velde
Sined Supplies Inc.

AAMSV3License

AAMS Auto Audio Mastering System V3 - Limited Freeware Version, upgradable to Full Version.

This software package is available free of charge, as for the AAMS Limited Freeware Version. But with a high encouragement that the user Registers for AAMS Full Version when used over a longer period, to the author and main supplier of the AAMS product. Registration for a Full Version License can only be done by the author and main supplier of AAMS. Go to www.curioza.com / Registration Page for more details.

This software package AAMS V3 is free of use for the Limited Freeware Version. You can continue to use AAMS V3 Limited Freeware Version, without registering for the AAMS V3 Full Version. But more professional options are not available in the AAMS V3 Limited Freeware Version. The AAMS V3 Limited Freeware Version is mainly intended for the ' easy' and ' fast' user. were mastering of a single track is done with a few button clicks and is done by AAMS V3 automatic functions. The end result will be a fully mastered track and all free of charge, AAMS V3 will stay Limited Freeware! You can master as many tracks as you like.

Register for the AAMS V3 Full Version.

This software package is available free of charge, but with an encouragement that the user makes a registration to the Full Professional Version. You can show your appreciation and support future development by registering AAMS, and make full use of the AAMS Software Full License Package. 
Without any blocking of options. A registered and licensed user can make use of all AAMS V3 professional functions! The price for registering a single computer and full license for AAMS V3 will be around 65 dollars or 65 Euros. For each single computer after registration is around 33 Dollars or 33 Euros.

Go to www.curioza.com / Registration Page for more details.

Thank You!

Denis van der Velde
Sined Supplies Inc.

EinsteinEVO

For all users who want to know the background to of the creation of AAMS Auto Audio Mastering System, here are some words about the author.

AAMS V1 to AAMS V3 - Year 2004 to 2016!

The programming of AAMS Started in mid-2004, at which time I had done enough manual mastering to see that some aspects could be automated. At first the AAMS V0.5 Beta program was simply creating suggestions for EQ, Multi-Band Compression and Loudness. This information was displayed and could be used to set up external equipment like plugins or outboard gear. This was time saving and made the mastering process more visible.  Then in AAMS V0.97 I added the routine for saving a Firium Preset based on the Graphic EQ Suggestions that would help setting up Firium EQ without having to do this manually (which that was time consuming).  I could listen each time AAMS calculated suggestions and confirm that it really did speed up the mastering process. After adding some more routines, testing and bug testing the small Reference database with 100+ presets to use and scan for a good sound. AAMS V1.0 was released on 01/01/2005 and soon a user base was established. The best thing about releasing AAMS V1.0 to the public, was that more users were giving information back on their feelings about the program. After some time and changing some functions to be more defined, AAMS V1.1 was released. This version was quite stable and gave good information to users in its suggestions. The information back from users confirmed that the suggestions were quite good and helped most users very well. Although AAMS V1.1 needed some work to understand most users were very pleased with AAMS results. Most complaints about AAMS V1.1 were that the calculations were not very fast and the application needed some guidance when installing.

So for AAMS V1.5 I had to speed up the programs calculations and change the AAMS V1.1 platform. This meant full recoding and programming. For the requested use of DSP it was also necessary to increase the speed and the way AAMS V1.1 operated. After some DSP-EQ coding and reprogramming, AAMS V1.5 was released with its own internal Player and DSP-EQ. Now it was possible to listen and play AAMS Suggestions through a windows soundcard that helps scanning for a good sound. The DSP-EQ had a natural sound and worked correctly for most users, so the programming for DSP-Compressor and DSPLoudness could continue.

Now days AAMS V3 has got its own internal Mastering Rig! 
With the combination of DSP-EQ, DSPCompressor, DSP-Loudness you can complete a quality mastering job within AAMS with ease. I have a good understanding of mastering and AAMS contains those ideas. AAMS users also have a lot to say and have good ideas, so if you have something to add your ideas will be listened to. I do hope you have as much fun as me using the AAMS Program.

We do make a lot of work making AAMS a good and steady mastering alternative and I know AAMS can make a good sounding master. So please donate for this software when you are using AAMS and like it! We can use the donations for future updates and for keeping AAMS alive. Also we will use the donations for creating more software tools in future SSI releases. Register AAMS V3 for Full Version.

Thank You!
Denis van der Velde
AAMS Author

AAMSDenis

BelieveIt1

Some music makers, mixing their own tracks into a stereo file, will always depend on a good hearing and good use of processors to make their mix into a good sounding master. Therefore AAMS does not replace a professional mastering engineer, but however AAMS does contribute towards learning and results in a good mastered sound when you apply these rules;

You can’t polish a turd.

JeremyTurd JodaTurd

Meaning if you mix lacks a good sound and is not mixed well enough, do not expect a mastering engineer or AAMS to make your sound better. AAMS does in most cases a good job. If not? Refer back to your mix, when mastering results in bad results or you expected better results. Mainly then AAMS can be a good learning tool, by mastering your mix down into a mastered audio file. At least you can hear the differences and find out were you mix needs improvement. It is always better to take care of separate mix tracks, resulting in a better overall sound. AAMS is based on that the user can expect what AAMS is going to do, so maybe understanding better what can be done inside a mix, and what can be done at the mastering stage. I would say Mixing is not about Mastering, vice versa.

For instance when you hear less or more bass frequencies, do you expect AAMS to correct them perfectly?

Well, AAMS does improve them. But when you are not happy with the amount of bass, you could adjust the Base drum and Bass inside your mix more and return back to AAMS, and see and hear the results. In some cases a professional mastering engineer will reject your mix down and will warn you for end results. Using AAMS as a learning tool for Mixing and Mastering, will improve your mixing skills and your sound. Some believe that only hearing, is the only way to master. That is old fashion true. But however, in most cases hearing is distorted by misuse of equipment, mistakes, and the room were you listen. Fatigue hearing is the most common distortion in mixing and mastering. When your ears are hearing the same music over and over, you will get distracted and will have to rest and come back another day, with clean ears! This is time consuming. Mastering manually is always time consuming. I say a master in 10 minutes can only be done by AAMS, or by manual results with a very good created mix down. The AAMS path is that audio files can be mastered inside 10 minutes by AAMS. So that is a timesaver, and the user can concentrate on the mix.

If this is mixing again, or using aams to have the best sound possible?

Even when you have the Best sound on one speaker system, why is it possible when you listen on another system the sound can still be bad or worse ? Well, some mix or master with headphones. Some mix and master with expensive speaker systems, enclosed or open rooms. Some are hearing bass frequencies better than others, some hear treble frequencies better than others. Still, when you switch speakers or the user, the same master will sound differently. The difference inside a car and household rooms can affect hearing the same sound. AAMS is mastering equally for all sound systems to have the best possible sound quality on all sound systems. AAMS does the same job, each time you use AAMS, the results are the same way, each time you can expect what AAMS does. AAMS has no distortions as fatigue or room environments or bad hearing. AAMS is purely based on precise calculations and you can expect what is coming out, is just the same. Each time. In the longer run you will expect what your sound will be, running it through AAMS. Because you can depend on AAMS doing the same job, over and over.

Do you believe this or not?

Fiddling with knobs, listening for the right sound in all rooms or speaker systems, taking more time and still thinking you did not get it exactly right ? This can be all over with thrusting AAMS to make a good master out of a good mix. In most cased Mastering is a byproduct. You have mixed your track or album, and you need it out to the public fast. Finally it is your choice!

Do you spend money on AAMS or a real professional mastering engineer ? For users who want to be in the creative music and mixing process, AAMS is a real solution with easy and costs far less money and will have a change of making your mix into a good sounding master, easy. And off course, never will you hear a real mastering engineer say ‘use AAMS instead’. Because they need your money and workload.

What are you going to do?

The music buss collapsed by free music. The money to be made is not the same as 10 years ago. The equipment is not the same. At least you can now save money for mastering your ‘free’ music. And thrust me AAMS does a really good job. 
I say 70.7% at least and more. Just concentrate on making good music and mixing it good! AAMS will do a great job then for you….

Still it is up to you.

We intent to make AAMS to create the best sound possible on all systems. If you do not thrust this statement, you can still try AAMS Limited Free Version and listen how your mix will sound, for Free! Or if you do not have any thrust, let your masters be done by a real mastering engineer or mastering website and still compare that with AAMS results ? Who is the better ? And if you are a real doing it yourself person ? AAMS is a great learning tool also! Need easy mastering ?

Use AAMS! We thrust in it!

AAMS V3 is Freeware to Try, So before bashing it, Try it!

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