Filter

input file : source soundfile

output file : destination soundfile (when marked)
or result.wav (for unregistered files)

processing mode : local or global

You have at your disposal five different filters that can be used to change the frequency content of analyzed signals :

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, namely, all components up to a desired cutoff frequency.

Bandpass filter can be used to remove from the analyzed signal both the low frequency components (those placed below the lower cutoff frequency) and the high frequency components (those situated above the higher cutoff frequency).

Bandstop filter can be used to remove from the analyzed signal the mid frequency components (those placed between the lower and higher 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. You can use it, for example, to remove narrow-band interferences due to the power supply or electrical coupling.

By altering the harmonic or timbral contents of a recorded sound you can correct certain types of problems that may have occurred during the recording or transfer process. For example, a direct transfer of a musical recording from old 78 RPM (revolutions per minute) disks to a wideband playback system will be highly noisy due to the limited bandwidth of the old record. To reduce this noise, a bandpass filter with a passband matching the bandwidth of the old record can be utilized. Some older recordings can be made more pleasing by adding a broad high-frequency peak in the 5 to 10 kHz range and by filtering out some of the lower frequencies. The notch filter is particularly useful in removing 60 Hz power supply hum.

Attenuation

You can determine the 'sharpness' of each of the filters described above, by choosing the attenuation coefficient (expressed in decibels per octave) - the larger the value of this coefficient, the steeper the 'cut' at the cutoff frequency or the narrower the notch. Sharper filters are more complex, and hence they result in longer processing times.

Linear phase

The frequency shelving Butterworth filters are the basic 'building blocks' used to design various types of filters provided by DART XP Pro. Belonging to the class of recursive infinite impulse response (IIR) filters, the Butterworth filters guarantee fast processing (which is an obvious advantage) but also introduce nonlinear phase distortions, i.e. they do not preserve the shape of the input signal. This sounds like a major drawback but whether it is or is not depends strongly on the purpose and context of filtering. The point is that our auditory system is practically insensitive to phase distortions introduced by filters, i.e. despite the apparent differences in output signals yielded by linear phase (shape-preserving) filters and their nonlinear phase counterparts, the sound we hear is exactly the same.

On the other hand, whenever filters are used for local processing only, preservation of the shape of an input signal is desirable - otherwise some discontinuities are likely to appear at the boundaries of the selected block.

Whenever you would like the filter to preserve the waveform shape, check the Linear phase box - the signal will be filtered twice : first in the forward, than in the backward direction. Such bi-directional processing allows for the linear phase condition to be fulfilled, though it will slow down the computation.

NOTICE

When a two-pass filtering procedure is used, the filter characteristics are automatically adjusted so that the overall signal attenuation in the stopband region is equal to the declared one, e.g., if you set the attenuation to 20 dB/oct, the filter with 10dB/oct attenuation will be used for the purpose of combined forward/backward processing.

Previewing results of filtering

To perform on-line tuning check the On-line box and select the processing range (entire file, from cursor, local, block). Then press the Play result button. Working in the on-line mode DART XP Pro will allow you to listen to the filtering results while changing the program settings. Each time you modify any of the settings the green Ready light situated next to the On-line box will go off - it will be switched on again as soon as the results obtained under the new settings are available. At any time during the test you can press the Play source button to bypass filtering and listen to the original recording.

If your computer is too slow to perform tuning in the on-line mode leave the On-line box unchecked. In cases like this all experiments will be performed off-line on a selected fragment of the recording.

NOTICE

If, during the on-line testing, the values of processing parameters are chosen incorrectly (e.g. the lower cutoff frequency of a bandpass/bandstop filter exceeds the higher cutoff frequency) the corresponding items are displayed in red. The inappropriate settings are ignored by the system until they are defined correctly (the last correct settings are used instead).

Local filtering

All filters can be operated locally, i.e. in the selected area only. In such a case:

1. If the output file does not exist yet or is not opened, a new soundfile, identical with the input file, is created prior to processing.

2. The filtering algorithm is run in the selected area and the corresponding block of samples in the output file is modified.

If the output file already exists and is opened, only the second part of the above procedure is realized.

NOTICE

When the results of local filtering are pasted over into the output file, some discontinuities may appear at the boundaries of the selected block. Usually this problem can be avoided, provided that the corresponding filters are operated in the linear phase mode; if not - try using DART XP Pro tools to enforce a smooth transition.