Features
- Sound input from line output and microphones
or from sound files and processing
- *.wav format
- Real-time noise suppression, changing
filtering settings without playback stop
- Automatic report on sound processing for
expert's conclusion
- Preset filtering schemes for typical noises
- Saving filters parameters for further usage
- DirectX support
| Highlights
- Efficient suppression of all types of noises and
distortions
- Adaptive algorithms of filtering
- Easy to use
- Filters can be combined to process the noisy
record simultaneously
- Low quality records processing for further instrumental
ID analysis: "weak" formants enhancement
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Filters:
The EQUALIZER module contains several filter and display functions. Automatic (one button), Semi Automatic and Manual filter adjustment. Filtering is available in the form of a graphic equalizer, inverse filtering, harmonic filtering, and spectrum correction. User controls include setting number of equalizer bands (128, 256, 512, 1024, 2048), control of X and Y display axis, averaging, linear and log display, notch filter characteristics and elastic control of sliders.
The EQUALIZER can be displayed in a large mode for finer detail of the spectrum. The active signal is displayed with the characteristics of the applied filter displayed as an underlying trace. This feature allows the user to see the filter in use and make adjustments for the best results.
The BROADBAND noise filter is an adaptive frequency subtraction filter ideal for use to suppress broad band noise and periodical noise. Simple to use setting choices are available or more in depth manual controls for the experienced user control suppression depth, intensity, fine tuning of the coefficients, adapt time, and frame time (points / sample rate). The filter can be applied to broadband noise associated with radio transmissions, tape noise.
The BROADBAND AUXILIARY window provides controls of the band pass filter (ON/OFF) and the cut off frequencies. Additional controls include Suppression Depth which controls the reduction of spectrum components, Suppression Intensity which controls the efficiency of the filter, Time Constant which controls the adapt time, T-Smoothing which controls smoothing of the filter (fine tuning), and Frame Size which controls filter spectrum resolution (spectrum resolution, number of spectrum bands, size of data block)
The DYNAMIC filter is an audio compression filter which permits the user to control the volume of the loud speaker in relation lower volume speaker. A threshold level can be set using the slider bar to determine when the filter is active.
The MU-TRANSFORMATION filter works much like an automatic gain control. Low level audio is amplified while high level audio is slightly or not amplified. Controls set the level at which the signals are amplified.
The ADAPTIVE filter provides three methods of filtering. Harmonic, frequency and time domain filters. Filter size can be set between 128 to 8192. In addition standard adapt speed and threshold controls can be set. This filter can be applied for typical room noises, vibration noise, tones and harmonics, slow music, reverberation / echo type noises.
The FREQUENCY STEREO filter is a powerful filter working in the frequency domain. Filter size from 64 to 8192 with control over adapt speed, delay measured in centimeters. Point Source filter uses a linear reference noise source such as a radio or TV signal. Reference Noise filter uses a noise signal from a microphone positioned beside the noise. The Ambient Noise Filter works to remove noise that is not common in two adjacent microphones. Additional controls include adapt speed, cancellation depth, and signal delay.
The TIME STEREO filter is a two channel filter for noise cancellation where the second channel is used for a source for the interfering noise. Time Domain filters provide some advantages over Frequency Domain in particular applications. Controls set adaptation speed, channel delay, choice of inverse output signal (that which is removed), and filter size.
The IMPULSE filter is designed to deal with transient signals. Two types of impulse filters are available. IMPULSE filter will remove transient signals. AMPLITUDE filter will reduce transient signals. In addition, threshold levels and fine tuning based on speed / duration of the transients can be set.
The INVERSE filter is works off the application of a flat audio spectrum. The filter will decrease strong components and increase weak components to form a flat spectrum. Controls include adapt time, maximum applied gain to the signal, inverse filter threshold in order to control what segments of the spectrum the filter is applied to. In addition, strong tones and harmonics can be included in the activation of the INVERSE filter.
| The INVERSE FILTER AUXILIARY window provides controls of the band pass filter (ON/OFF) and the cut off frequencies. In addition, a range of gain is provided for each of the limits from +24 dB to -24 dB. (amplify to attenuate) |
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| The AMPLIFIER can be inserted as needed to a gain range between - 60 dB to + 60 dB. |
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| The SLOWING module enables you to change the speed at which a file is played back while maintaining intelligibility |
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| The INPUT module is used to input a signal from outside the computer into the sound card. The module will set mono / stereo, sample rate for signals that are input into the computer and saved as a file on the computer. |
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| The FILE INPUT module functions as a standard WAV player. Features include Play, Stop, Rewind, Fast Forward, Jump to End, Jump to Beginning, Loop Start, Loop End, Loop On / Off. |
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| The SAVE module enables you to save the processed data as a sound file. Choice of mono or stereo. |
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| The SPEAKER module enables you to choose the type of output and volume to the output section of the sound card. In addition, to increase comfort in listening delay can be added to create an artificial stereo output (Pseudo). In addition, the module will measure (MSR) the ability of the given computer to perform the desired processing. |
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| The WAVEFORM window provides a visible display of the signal at the point where the window is inserted. Show in this image at the beginning of the filter scheme. The range of the Y axis can be increased or decreased and is referenced to the input signal level. |
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| The WAVEFORM window is typically inserted at the end of the filter scheme where it is possible to see the resulting output signal. When the use of a filter is in doubt. the waveform window can be inserted to provide a look at the effects of a given filter. |
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| The STEREO WAVEFORM window is the LEFT and RIGHT / Channel 1 and 2 in a single window. Two colors are used to differentiate the two channels. |
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| The FILTER SCHEME window is a combination filter display and "tool" pallet. The filters are assembled from Input at the top to the Output at the bottom. Filters in use are highlighted in orange. The tool pallet provides a list of icons of the available filters which can be dragged and dropped into the desired positions. The tool pallet also includes one button icons for typical requirements from cleaning up / redrawing the window (removing spaces), starting with an empty layout, loading saved schemes, saving schemes, switching left and right channels, changing the size of the window. |
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