With Normalize you can optimize the resolution of a sample after recording. Although we recommend you to always try to record with the maximum possible amplitude there are cases in which recording cannot be repeated with a better gain setting (e.g. live recordings from studio musicians). Then it is a good idea to optimize the sample's volume to its maximum in the digital domain. Normalize determines the loudest sample in the area and uses it to bring the sample to the absolute gain level specified with 'Maximum Peak'. When processing stereo samples both stereo channels are tied together so no relative change in volume will appear between channels. If you want to process both channels independently you have to split it into two passes and normalize both channels sequentially.
If you set 'Maximum Peak' to a value of 0 dB DFS all bits of the currently used resolution (8, 16 or 24 bit) are being used. This gives optimum audio quality.
Maximum Peak
This parameter sets the new maximum peak level for the sample data. You should set it to 0 dB DFS to achieve the maximum possible amplitude. The audio material is normalized to the value given here no matter how loud it was before. If you set Maximum Peak to -6 dB, the block will be normalized so that the maximum amplitude appears at -6 dB after processing.
Transpose
This function changes the pitch of your recording. It uses a high quality resampling algorithm with anti-aliasing to provide optimum quality. As pitch changes, the duration of the sound is also altered. If you want to change the basic pitch wihout affecting the length of your recording use the Pitch Scale or the Phase Vocoder function.
Transpose Coarse
Defines the amount of transposition in semitones.
Transpose Fine
Defines the amount of transposition in cents (1 cent = 1/100 semitone).
Keep file length
If this option is checked, the length of the file is not altered. Enable it if you are working with raw audio files in order to stay compatible with the information contained in the header.
Fade-In
Fade-Out
These two plug-ins can be used to generate a fade-in or fade-out with a linear ('ramp') fade-zone characteristic. The fade length is determined by the length of the selected area.
Reverse Selection
This plug-in reverses the playback direction of the material within the selected area. Try processing reversed samples with different effects and then reverse them again to regain the original playback direction. This can create really interesting effects.
Sample Rate Conversion
This plug-in converts the sample rate (number of samples per second, a number directly related to the audio quality of the sample). It uses a high quality resampling algorithm with anti-aliasing to provide optimum quality. Use this function to convert samples to the standard rates of 44.1 (CD) or 48.0 kHz (DAT) if you plan to digitally record them on CD or DAT. You may also use this function to lower the sample rate for a particular recording, e.g. in order to save disk space on your harddisk.
Please note that the Apple SoundManager converts the file's actual sample rate to the system sample rate during playback. Since it is using a simple linear interpolation for doing this, artifacts may occur when playing back recordings having a sample rate different from the one set in the control panel "Sound".
When using Sample Rate Conversion you should always convert the whole file (use 'Select All' and enable the 'Easy' checkbox next to the in/out pop-ups in the wave edit window) to avoid different playback speed and pitch change of portions of the file.
Keep file length
If this option is checked, the length of the file is not altered. Enable it if you are working with raw audio files in order to stay compatible with the information contained in the header.
Add channels
This function adds the signals of both inputs (inA, inB). To create a mono file from a stereo file, you may also use the 'Save as╔' command from the file menu to create a mono file from a stereo file.
A/B Input Balance
Controls the input balance of the two inputs (inA, inB). With this slider fully left, only input A will appear in the output. With the slider fully right, only input B will appear.
To avoid clipping, the volume is adjusted automatically.
Hall Reverb
This algorithm offers a studio quality room simulation with a very dense and smooth sound. It is mostly suited for large-sized rooms and creates a rich and brilliant sound great for vocals and percussion.
Wet/Dry Mix
Determines the wet/dry effect mix. A setting of 100 will only output the effect signal.
Reverb Decay Time
Determines the duration of the reverb. High values create larger halls.
Early reflection level
Determines the volume of the early reflections. Higher values will tend to sound resonant and are more typical for smaller rooms.
Keep file length
If this option is checked, the length of the file is not altered. Enable it if you are working with raw audio files in order to stay compatible with the information contained in the header.
Time Scale
This function changes the length of an audio file without affecting its pitch.
New Length Coarse/Fine
Determines the new length of the sample. A value < 100% means a shorter duration than the original, a value >100% means time stretching.
If you do not get satisfactory results for a particular audio recording or if you want to apply very high expansion ratios try splitting processing into different passes at lower expansion ratios. Bear in mind that all passes add up exponentially rather than additionally. That is, two passes at 124% each will give a total expansion of 154 (1.24*1.24*100%) rather than 148 (100% +24% + 24%).
The formula for getting the time stretching percent value from a bpm value is: percent = 100*old/new, with old and new meaning the old and new (desired) tempo of the audio recording. In the retail version of sonicWORX Artist, you can set the BPM values directly.
Keep file length
If this option is checked, the length of the file is not altered. Enable it if you are working with raw audio files in order to stay compatible with the information contained in the header.
Pitch Scale
This function changes the pitch of an audio file without affecting its length.
Semitones
Determines the transposition in semitones.
Cent
Determines the transposition in cents (1 cent = 1/100 semitone).
* Export Range to .WAV File
Exports the area between the markers as a MS-RIFF WAVE file. When yo hit the Execute button, you will be asked for a name and a location where the .wav file should be created. Type in the file name and hit 'Save'.
Export Stereo File
sonicWORX will always create a stereo file if this option is checked, no matter if the selected area is mono.
Phase Vocoder
This method for independent time and pitch scaling was introduced by Flanagan and Golden in 1966 and digitally implemented by Portnoff ten years later. It uses a Short Time Fourier Transform to convert the audio signal to the complex Fourier representation and alters the signal in the frequency domain. In short, the signal is regarded as consisting of a mixture of sines and cosines with varying amplitude and frequency (derived from phase), and the harmonic evolution of a sound is altered by stretching and shrinking the frequency and amplitude development of the individual components thus allowing independent manipulation of the speed and pitch of a sound. A Phase Vocoder works well with signals that are by themselves very 'sine-like', i.e. are well represented by a sum of sines mixture. In the case of percussive signals, due to the lack of localization of the sine and cosine basis functions in the time domain and the interpolation used in the process, the Phase Vocoder will cause audible signal smearing. This effect is common to all Phase Vocoder implementations and not specifically related to this software. Use the Time and Pitch Scale functions of Artist Basic if this effect gets too annoying.
Expansion/Compression Ratio
Determines the new length of the sample. A value smaller 1.0 means a shorter duration than the original, a value greater than 1.0 means time stretching.
The formula for getting the time stretching ratio value from a bpm value is: ratio = old/new, with old and new meaning the old and new (desired) tempo of the audio recording. In the retail version of sonicWORX Artist you can set the BPM values directly.
FFT Frame Size
Defines the basic frame size used for the Discrete Fourier Transform (DFT, more popular known as 'FFT' in its efficient form used in practical applications). Smaller sizes will produce less smearing (reverberation) but will resolve more poorly in the frequency domain. This means you will get more artifacts, especially when working with sounds that have a noticeable low-frequency component. Larger values will produce more smearing but will resolve better in the frequency domain.
Analysis Window Shape
Defines the window shape that is used for the analysis. Different shapes will work well with different material, depending on the spectrum of the input sound. Experiment to find the optimal setting in your case.
Transpose Coarse
Determines the transposition in semitones.
Transpose Fine
Determines the transposition in cents (1 cent = 1/100 semitone).
Keep file length
If this option is checked, the length of the file is not altered. Enable it if you are working with raw audio files in order to stay compatible with the information contained in the header.
