Toolbox Features

Scale

      Scaling allows you to amplify or attenuate the volume of a selected block or the entire sound file. The scaling factor controls the level of amplification or attenuation. For example, scaling factors set below one result in signal attenuation (turning the volume down), and factors greater than one cause signal amplification (turning the volume up). Be careful when you amplify the sound, because parts of the audio that may be "sticking out" before scaling could clip if the scaling factor is set too high. To avoid this mishap, You can use the find maximum gain and the program will set a limit so that no clipping will occur after scaling the file.
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Maximize

      Maximize is much like scaling in that it amplifies the volume of an audio file. Maximize raises the volume of the sound file to its highest point without clipping the audio. A benefit of using this function is that sometimes there are imbalances in a sound file’s volume, and using maximize could "balance out" the entire file.
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Reverse

      This option allows you to reverse an entire signal or a fragment. This is especially useful in certain instances. For instance, you may derive superior results through reverse-time that are superior to what can be achieved through normal processing. Sometimes, when you are Declicking a sound file, you may get better results after a regular Declick, by reversing the sound file and then Declicking again. This is because, the shape of a wave file at the beginning of a noise disturbance like a click could sometimes be more defined when it is reversed. Therefor, the program will recognize more noise disturbances, and the quality of the result file that was reversed will probably transcend the quality of the file that reverse was not used.
      Reversing can also be used for more recreational purposes, such as revealing hidden backwards messages embedded within songs and music. These types of messages where especially popular among the counter-culture groups of the 1960’s and 1970’s.
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Add

      Add is a mixing tool that allows you to take two files, the source and the destination, and merge them together into one file. The resulting file will sound like the two files are playing simultaneously. If the file lengths are not equal, then the resulting file is trimmed to the length of the shorter file.
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Subtract

      This function is used to compare two files by subtracting the contents of the destination sound file from the contents of the source sound file. For example, you may want to hear what was taken out of a sound file (just the clicks, pops, hissing, etc.).
  1. To use subtract, find a file that you want to restore.
  2. After performing the restoration, you will see a new renovated destination file.
  3. Open the toolbox menu and click on subrtact.
  4. The new file will contain only the noise that was taken out of the source file, or in other words, the difference of the source and destination files.
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Mix

      Mixing is used when you what to combine two files together. You get the effect of two sources of music playing simultaneously.
  1. To mix two files, open a file and designate it as the source file.
  2. Open the second file and designate it as the destination file.
  3. Now open the toolbox and click on the Mix function.
  4. You can adjust the settings and audition the mix before you process.
  5. Click process when you are satisfied with your results.
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Mono\Stereo

Split is used to separate the channels of a stereo recording into two mono sound files containing the left and right channels of the original file. This is especially useful when you want to work with only one channel of a stereo recording.

Unite is used to combine two mono files in to one stereo file. The resulting file contains the information of the mono files in the left and right channels (one mono file goes to the left channel and the other file goes to the right channel to form a stereo file).

Convert to Mono allows you to convert a stereo sound file into a mono sound file by mixing the left and the right channels.

Convert to stereo allows you to convert a mono sound file to a stereo sound file. By duplicating the mono track.
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How to use smart editing

Smart Mute: If problems with signal discontinuity can be handled properly, local muting may be a fast and efficient way of eradicating disturbances. The simplest way of avoiding clicks due to a local muting, is by positioning both block edges at signal zero-crossings. Smart Mute is a tool that automatically searches for signal zero-crossings and adjusts the left/right boundaries of a selected block accordingly. If, despite adjusting the block edges, the results are not satisfactory, you can enforce smooth signal transactions at both sides of the block by local fading.
  1. Select a block containing the part of the signal you would like to mute. Do not worry about position of block edges at this stage- you can mark blocks on condensed signal plots if so convenient.
  2. Pull down the toolbox menu and click on the smart editing tools and open toolbox.
  3. Adjust the left or right block edges by moving them to the nearest zero-crossings situated to the left or right of the current position. You can audition your results by clicking on the preview button.
Smart Cut: Local cutting of unwanted material may be an efficient tool for elimination of disturbances. If the removed fragment is sufficiently short and the signal on its right hand side is a "natural continuation" of the one observed on its left hand side, our auditory system can be easily fooled and the disturbance is lost without a trace. Smart Cut is a tool that is capable of localizing such "natural continuations" by itself, saving a lot of time and effort.
  1. Select a block containing the part of a signal you would like to eliminate. If a signal has strong periodic components, try to make the block width commensurate with the signal period or its multiplicity.
  2. Open the toolbox menu and click on Smart Editing. Then choose Smart Cut.
  3. Adjust the left and/or right block edges position (the search is limited to 500 ms on the left and right side of each block edge). When you decide to move the right block boundary to the left or right of its current position, the signal just preceding the left block boundary will serve as a reference and vise versa. Use the accuracy setting to increase and decrease the accuracy of the fit.
Crossfade Cut: To avoid signal discontinuity at the joining edge, the signals observed at both sides of the removed fragment can be crossfaded into each other.
  1. Select the block containing the part of a signal you would like to eliminate. If a signal has strong periodic components, try to make the block width commensurate with the signal period or its multiplicity.
  2. Open the toolbox menu and click on the Smart Editing. Then choose Crossfade Cut function.
  3. Select the crossfade time to enforce smooth transition of the signal on the left side of the block into the signal on the right side of the block.
  4. You can audition the results by pressing preview.
  5. If you are satisfied with the results you can click on the process button. Otherwise, try adjusting the settings until you have what you want. Please note that for long recordings, cutting may be a very time consuming operation as it involves rewriting the entire audio file.
Note: At any time you may reduce the size of the crossfade cut dialogue box by pressing the minimize button situated in the upper corner.
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How to use the filters

Lowpass filter can be used to remove the high frequency components of the signal starting from a desired cutoff frequency.

Highpass filter allows you to remove all low frequency components of the signal up to a desired cutoff off frequency.

Bandpass filter can be used to remove from the analyzed signal both the low frequency components and the high frequency components.

Bandstop filter can be used to remove from the analyzed signal the mid frequency components. These are the frequencies between the high and low cutoff frequencies

Notch filter is designed to remove the components contained in a very narrow frequency band centered at a given notch frequency from the analyzed signal. It is very useful in removing narrow band interferences due to the power supply or electrical coupling. How to use these filters:
  1. Depending on what type of frequency you are wanting to surpress you will use one of these filters.
  2. Open up the sound file that you want to work with.
  3. Select the appropriate filter.
  4. Make the appropriate adjustments to the settings and audition the settings using the real time processing.
  5. Click on the process button
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How to change the sample rate

Resample:
Resample is used when you want to change the sampling rate of a sound file. For example, you might have a file that is in 22 kHz and you may want to change it to 44 kHz. You can use the resample tool to do this.

Change Speed:
Change speed allows you to change the playback speed of an audio file.

Adjust RPM Rate:
Adjust RPM Rate allows you to convert an audio file from one RPM rate to another. For example, if you do not have a turntable that supports 78 RPM recordings you can convert the audio file into 78 RPM after recording from the record player.

Trim Length:
Trim length allows you to speed up audio data of an entire compilation without audibly changing the sound of the music (changing the speed by more than 5% may damage the quality of an audio file). This is used when you have a number of songs that you want to write to CD, but you pass the limit by a minute or so. Trim length will speed up the audio just enough so that you can fit all of the songs on the CD that you want without audibly distorting the sound files.

CD Format:
CD Format converts all of your wave files into 44kHz and 16 bit which allows them to be written on to CD.

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How to use the Find Pattern Function

Find Pattern is a great tool to use when trying to detect noise disturbances throughout a file. Noises like thumps, clicks, pops, and scratches are commonly identical in their wave form and repetitive throughout the audio file. Find pattern can be used to locate these disturbances so superior restoration can be achieved.
  1. To use the find pattern function, select a block of audio, at least 50 samples and no more than 500 samples, that you want to locate throughout the audio file.
  2. Click on the toolbox menu and click on find pattern.
  3. When the find pattern dialogue box opens, click on the get button to load the pattern in to memory.
  4. Adjust the accuracy factors according to how discriminating you want the find pattern function to be when looking for identical bits.
  5. Use the arrows to search for the next available identical audio bit in the sound file.
NOTE: You can work with the audio file while the find pattern dialogue box is still open. Minimize the dialogue box when you need to work with the audio file
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