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Sound Cleaner - Brokke System

1. 11 Frequency compensation Adaptive compensation algorithm Frequency compensation controls 12 Slowing 13 Clipping 14 Mu transformation 15 Impulse filtration Impulse filtration algorithm 0 Impulse filtration controls ooo 16 Dynamic processing Dynamic processing controls o ooo 17 Stereo processing Independent two channel processing StereoWave e o eee Adaptive stereo filtering o o o Stereo filtering in time domain Stereo filtering in frequency domain Processing composite stereo signals 18 Processing schemes Scheme widows 2 vlad Puget ie A geek ee we Creating and adjusting the scheme Duplication 2 aa 2 000020 e 8 CONTENTS Typical schemes Anasi agi a aa na a ee WE 92 Loading a typical scheme 93 Saving current scheme as typical 93 19 Sound Cleaner as Direct X plug in 95 20 Warranty 97 Tested and Approved o o 97 SU UPDOTE sun 24 vw ire Area Oe Se bee Paes 98 NECESSARY INFORMATION 9 Necessary Information This manual describes Sound Cleaner ver 5 x audio signal processing software Listed below are the telephone numbers that you may need if any questions or problems regarding the operation of Speech Technology Center products arise Tel 7 812 331 0665
2. 2 Fx ops DF Frame size Hamonic suppr i Compensation Time method Adaptation speed 13 Delay samples 1105 Figure 11 1 Frequency compensation window Frequency compensation toolbar buttons perform common func FREQUENCY COMPENSATION CONTROLS 67 tions Reset filter Fix filter Auxiliary options and Restore default filter settings To choose compensation mode use radio buttons at the right side of the screen Harmonic suppression allows to remove harmonic interferences pro vided its phase is relatively stable Compensation is used to suppress interferences in frequency domain Time method activates compensation in time domain Frame size is the most important compensation parameter in any mode It defines number of spectral bands and processed data block size The larger it is the more filter coefficients is used for compensa tion and therefore more spectral peaks may be removed In common cases greater frame size will lead to better compensation though it may also cause echo effects 512 2048 samples should be sufficient for most phonograms Adaptation speed slider sets filter adjustment time i e time the filter needs to tune itself to the variations of interference spectrum For common cases values around 20 25 are recommended If an interference spectral parameters change quickly you should try to increase adapta tion speed and vice versa Remember that large filter adjustment speed
3. Fax 7 812 327 9297 Mail Russia 196084 St Petersburg P O Box 515 Office 4 Krasutskogo str St Petersburg E mail info O speechpro com Web site http www speechpro com When requesting assistance you should have the following infor mation readily available e name of the product and version number e type of the computer and information about its configuration e name of OS being used and its version number e precise description of the problem Chapter 1 Overview Sound Cleaner software is designed to process and play audio signals in real time mode It may operate in any MS Windows environment Real time audio processing makes Sound Cleaner a powerful tool use ful for many different tasks such as noise cancellation during audio monitoring and so on It also drastically increases productivity and ad justment speed of processing tools The main tasks accomplished by Sound Cleaner are e Input of audio signals into PC memory and saving them on a hard drive e Real time noise cancellation and improvement of the signal qual ity e Ascertainment of speech fragments in poor quality phonograms Tf common Windows multimedia sound I O Card Sound Blaster is used In other cases see Installing sound I O hardware in chapter 2 for more information about OS supported 11 12 CHAPTER 1 OVERVIEW Delivery set Sound Cleaner delivery set includes e Sound Cleaner software e Wave Assistant signal editor
4. You may also save your own schemes as typical TYPICAL SCHEMES 93 Loading a typical scheme To load a typical scheme choose Load command from Typical schemes menu and then select a scheme Each of the schemes in the list is pro vided with a short pop up comment It specifies which type of noise or interference the scheme is designed to remove Saving current scheme as typical To save your own typical scheme activate Save as command in Typ ical schemes menu Save as typical scheme User Comment O O Config Name IE m Menu Text ss Example File PO zl Figure 18 2 Saving typical scheme dialog window In the dialog window Figure 18 2 you will have to fill four scheme information fields User Comment pop up comment to a scheme Configuration Name name of a file where the scheme will be stored without extension 94 CHAPTER 18 PROCESSING SCHEMES Menu Text Name of the scheme that will be displayed in the list of existing schemes Example file optional sound example file which will be loaded with the scheme Chapter 19 Sound Cleaner as Direct X plug in Direct X plug in mode is supported by Sound Cleaner version 5 10 or higher It does not affect any other program features As a Direct X plug in Sound Cleaner is proved to be compatible with e Adobe Audition 1 0 or higher e Sound Forge 6 0 or higher e WaveLab 4 0 Direct X mode is activated by pressing In this mode
5. Speech Technology Center Sound Cleaner Real time noise cancellation and speech enhancement software User Manual Operating Instructions Speech Technology Center St Petersburg Russia Sound Cleaner Real time noise cancellation and speech enhancement software User Manual Operating Instructions Copyright Copyright 1999 2005 by Speech Technology Center Limited STC Ltd All rights reserved Disclaimer Speech Technology Center accepts no liability whatsoever for any loss or injury incurred by the owner or by any third party while using this product and its user manual and specifically disclaims any warranties merchantability or fitness for any particular purpose The contents of the User manual are subject to change without notice Dear Customer Thank you for purchasing this product For optimum performance and safety please read these instructions carefully Contents Necessary Information o o o Overview Delivery Set aia aia dro WA woe bea ds Sound Cleaner capabilities Technical characteristics o o Preparing Sound Cleaner for operation System requirements o e e Installing sound VO hardware o Installing the software oo o Registration cams ra a Ge E Basic principles General information o o o Sound Cleaner main window and process
6. tends to impair speech signal quality Delay may be set for time 0 1000 msec and frequency from 0 to frame size compensation This slider sets an interval between the fragment where the interference is calculated and the beginning of compensation For frequency compensation delay should not be less than one half of the chosen frame size for time compensation 25 msec or more is recommended Otherwise speech signal may be greatly distorted Frequency compensation Auxiliary options window Figure 11 2 is opened with Ops toolbar button It contains Timbre correction set tings see chapter 9 Manual mode and Adaptation threshold slider 68 CHAPTER 11 FREQUENCY COMPENSATION Auxilary Options E Adap Threshold 1 m Timbre correction M ON OFF Speech from Hz 200 LF amplify dB o Speech to Hz 3600 HF amplify dB fo Cancel Figure 11 2 Frequency compensation auxiliary options window Adaptation threshold may vary within 1 32000 dB If signal am plitude exceeds this value necessary filter adjustments are made Thus you may set the threshold to keep filter unchanged during low signal level phonogram fragments for example when there is no interference Chapter 12 Slowing This process is used to increase or decrease signal playback speed without changing the voice pitch The window Figure 12 1 contains only playback speed adjustment slider Current speed coefficient in is displayed
7. HF and LF amplify parameters adjust sound level in high or low frequency area They work exactly the same as Q1 and Q2 sliders of the equalizer see Additional FC adjustment of the previous chapter For common cases it may be useful to reduce high frequencies setting HF amplify value to approximately 3 6 dB This is not a rule however in fact you may have to leave it unchanged or even raise this value There are also two parameters of the main window which are not 60 CHAPTER 9 ADAPTIVE BROADBAND FILTERING yet described Smoothness defines frequency smoothing of filter coeffi cients which like T smoothing is useful for suppressing musical tones which sometimes appear after processing Frame size auto tuning makes Sound Cleaner automatically choose the frame size according to sampling rate of the signal If you try to do it manually this flag will be automatically removed Model filtration If a noise you are going to remove with the broadband filter is station ary or close to it i e spectral envelope does not change in time and a phonogram contains a fragment of this noise without useful signal this fragment should be long enough for the filter to adjust itself then you may use this fragment as a model for filtration To do it find such fragment Wave assistant may be very helpful and run it in the loop mode Adjust the broadband filter so that it would completely suppress the signal noise in this case then press
8. USB device is supported in MS Windows 2000 XP Sound Blaster card will work on PC with any MS Windows OS installed If you use STC H216 external USB device or STC H189 sound I O card you will have to install it as well as necessary drivers according to instructions provided in respective manual and or device technical description INSTALLING THE SOFTWARE 17 Installing the software After you hardware is properly configured you should install Sound Cleaner software To do it place the included CD in your CD drive and run setup exe from Scleaner folder on this CD then follow instructions of the installation wizard Sound Cleaner and Wave Assistant will be installed on your computer Important If your PC operates under MS Windows 2000 or XP make sure you have access to the system register 1 e you have administra tor privileges before you start the installation and registration You won t need it after the registration is over Registration After the installation the program will run in a demo mode correspond ing to Premium version of the software i e all the possible filters and schemes included You have to complete the registration procedure in order to switch to work mode To do it run Sound Cleaner from Pro grams Speech Technology Center group If you launch the software for the first time you will be offered to register immediately or do it later by choosing Register from the Register menu In both cases ju
9. a common stereo file with all the available filters Chapter 18 Creating and editing processing schemes During noise cancellation you will once surely face tasks unaccom plishable by standard means and typical schemes That s why Sound Cleaner includes lots of possibilities for flexible tuning of processing procedure One of these is creating you own unique processing scheme or editing an existing one Scheme window Scheme window Figure 18 1 displays and manages signal processing sequence This window is usually located at the left edge of the main Sound Cleaner window You may open it with Project View scheme if the window is not displayed by default The scheme is displayed in the window as a sequence of icons Each one represents a process included in the current scheme Icons are con nected with black lines which show the way of a processed signal pass ing through the scheme Icon s background color indicates status of the process orange background marks active processes and gray passive 89 90 CHAPTER 18 PROCESSING SCHEMES CHANNELS TR K a Amel BREGE e al ES lle _ A A E EN I Lu a Figure 18 1 Scheme window ones Double clicking on an icon places its window on top in the Sound Cleaner window Icons of all the available processes are displayed in a column to the right of the scheme There are also up and down arrows for scrolling the window Note that left cl
10. in increasing X axis scale two times Zoom out showing all the spectrum from 0 Hz up to half of the sampling rate Reset filter placing all the adjustment sliders to zero Autoscale Y axis according to maximum and minimum spectrum values Pp Zoom out decreasing X axis scale two times Select a rectangular area to be displayed in the spectram r window Select it and press left button and see a dashed z rectangle appear then drag it to specify the area you wish to view and release the mouse button Selected area will be enlarged to fit the spectrum window onl 50 dB y CHAPTER 8 EQUALIZER Switch Y axis from linear to logarithmic dB scale When this button is pressed equalizer displays signal in logarithmic scale otherwise the scale is linear Turn spectrum accumulation on and off While this but ton is pressed the program accumulates signal calculating and displaying average spectrum Depress it to switch back to instant spectrum Build inverse or harmonic filter selected by user basing on current spectrum either instant or average if spectrum accumulation is on Pressing this button automatically stops spectrum accumulation Note that ei ther filter is calculated within bandpass borders only Contrast the filter See Equalizer options section for more information about filter contrasting Important If you have turned on filter contrasting in Options menu it will be automatical
11. of equalizer bands and more fine and precise adjustments may be made To select number of bands you should open Options dialog box described further Try to set largest possible number of bands to achieve best filtering quality and precision but remember that it increases system load as well FFT window size is strictly determined by number of bands in fact it is equal to number of bands multiplied by four 47 48 CHAPTER 8 EQUALIZER Equalizer controls Equalizer window may be displayed as standard default and large the only difference being its size and number of filter adjustment sliders Figure 8 1 shows standard equalizer window lt 3 gt EQUALIZER 8192 128 7248 A A Figure 8 1 Equalizer process window x You can see toolbar below the window header and a black window where current signal yellow and filter green spectra are displayed EQUALIZER TOOLBAR 49 X axis zoom bar is just under this window currently visible area is marked with blue Two red markers at the upper edge of the spectrum window indicate bandpass borders you may drag them to change the borders Lower half of the window is occupied with filter band adjust ment sliders Elastic mode and Additional FC controls Equalizer toolbar Equalizer toolbar contains most important and frequently used process controls Set maximum horizontal zoom i e each adjustment slider corresponds to a single filter band Zoom
12. right border respectively over the indicator bar in the middle Position of both ring borders will be indicated by black markers under the bar while whole looped fragment will be highlighted with blue At the same time Cur rent loop borders digital indicators will display border positions from the beginning of a phonogram as hh mm ss Press Loop mode on off button to turn this mode on and off When it is active you should see ON message appear at the bottom of the window next to RING Start input from sound I O device button will switch the window to input from a device see Input from a sound I O device ADC View file button will launch Wave Assistant signal editor software loading currently processed file and loop borders automatically Import borders button will do the opposite namely set currently selected in Wave Assistant borders as loop borders in Sound Cleaner Input from a file process window See Wave Assistant manual for more information on this software INPUT FROM A FILE 35 Sampling rate mono stereo input mode total duration and cur rent input position are indicated below the toolbar To the left of the input indicator bar you may see Move loop arrow buttons Press them to move both borders of the ring in any direction with a Step specified in the field in the bottom right corner of the win dow in seconds Finally STOP flag will if it is set enable Sound Cleaner to play the looped fragment only once and
13. then stop the input instead of doing it over and over again as loop mode is supposed to work Chapter 5 Audio output As with input there are actually two types of audio output available in Sound Cleaner playback i e output to sound I O device and save output to file Playback This process also called Speaker allows you to listen to either pro cessed or source signal as it is being inputted So if you wish to listen to a signal while it is processed ensure that Speaker module is in the scheme and active Playback window is shown in Figure 5 1 You may choose to play the sound as mono pseudo stereo or stereo In stereo mode you may also specify with a radio button which channel you wish to listen to L means only left channel R only right one and L R both Do not forget to choose two channel processing scheme if you are going to play a signal as stereo you may choose stereo sch standard scheme For stereo and pseudo stereo you may also set the inter channel delay within 0 20 msec range To adjust delay value move the slider at the right side of the window 37 38 CHAPTER 5 AUDIO OUTPUT lt 14 gt SPEAKER o fx MSR Inter Channel Delay msec 5 C Mono 7 Pseudo Stereo 1 20 a oF VOLUME Figure 5 1 Playback process window Playback Volume slider is located in the bottom part It is linked to MS Windows audio mixer Try to avoid setting huge volume level be cau
14. these windows unfold You may then arrange them as you wish minimize or hide those which you Sound Cleaner MAIN WINDOW AND PROCESS WINDOWS Toolbar Rig STC SOUND CLEAN Sound Options Typical scRajes Wi SIETE Program header y 5 020 PREMIUM Registered to 46 Active process window 2 a 5 x Minimized windows _ a 21 eng wav lt 14 gt SPEA CHANNELS rim gt m m slao C Mono Lett Right 025 He STEREO Total time 00 00 20 Cureryfime 00 00 04 9 i PERRE E E ciracia lil vel a E la T Hide Left T Hide Right el l tlo trlylal TA EJE SEERPEUm Figure 3 1 Sound Cleaner main window don t need You may not however change size of these windows except Equalizer Every window represents a single process module of the currently loaded processing scheme the only exception is Processing scheme window described later in the separate chapter Note however that if a module is included in the scheme this does not mean that it is actually used for signal processing because it may be inactive Each process window header holds additional Activate button except common Mini mize and Close XI If this button is greyed the module is inactive i e is not currently used for pro
15. windows Mensa a a ti iB es das Seago a Ade Sound 2 so8 et wie Se Be he he SU OPONE A Typical schemesS o o ooo oo WINDOWS urea rt rc De Project m jan de phe o dla Ae Doty Behe Ge as TEXT report ior ica Sc ea Bo ek Re ls 11 12 12 13 15 15 16 17 17 CONTENTS Register oaie sti ay OTe een gd Ae 26 Hel pss ci a Gee ek hw Se Ae a 27 Toolbar teu istiare i ea a to EY ES A 27 Processes os Dare Pee A Pe See oe Paes 28 Audio input 31 Input from a sound I O device ADC 31 Input froma Mecca ee ae BP he 33 Audio output 37 Playback ii s 20g be ee ee a eee 37 Pseudo stereo Mode o oo 38 Measure mode e 39 Saving signal to analog media 39 Save tofe iio cds BoP aoe A a A ae a a 40 Amplifier 43 Waveform 45 Equalizer 47 Equalizer controls o o o oo o 48 Equalizer toolbar o o o 49 Equalizer options o o e e 51 Zooming and scrolling o o 53 Adjusting filter PC cis Fe a 53 Elastic mode sa ca cacao 0 00002 eee ee ee 53 Additional FC adjustment o oo 54 Adaptive broadband filtering 55 Broadband filtering controls o 55 Automatic M0Ode o o a 57 Manual mode 57 CONTENTS Model filtration 10 Adaptive inverse filtration Adaptive inverse filtering controls
16. within a phono gram Sound Cleaner may input sound from DAT and WAV files DAT is an extension of audio files used by SIS software and Ikar Lab by STC WAV file formats supported are e 16 bit PCM e 8 bit PCM e u or A law compression 8 bit e EEE FLOAT 24 bit Input from a file window is shown in Figure 4 2 Slider bar in the center of the window displays current playback and input position Just below the header is the toolbar Loop controls and indicators are at the bottom of the window Some of the toolbar buttons are just common input and playback controls Open file Start Stop Rewind Fast Forward Other tools control specific input options After pressing Go to file time you may enter time hh mm ss from the beginning of a phono gram to start input and playback from Current input position slider will be placed accordingly You may also drag the slider with your mouse pointer or arrow keys You may select to play specific phonogram fragment in a cyclic loop mode To set beginning and end of the loop ring you may either place 34 CHAPTER 4 AUDIO INPUT Go to file Mark start end Loop mode Start input from of loop on off sound I O device Os nea A 00 00 05 gt STOP 1 Step Current loop Loop border i borders markers borders Figure 4 2 Input from sound file window the slider in a desired location and press Mark start end of loop but ton or simply click left or right mouse button for left and
17. AUDIO INPUT sound VO device and Save in a file and then process it Figure 4 1 shows process window Toolbar below the header con tains standard input control buttons Start input Pause and Stop in put Read sound file button at the right side of the toolbar will switch to Input from a file see Input from a file Z nia E gt C Mono Frequency i Stereo 11025 Current time 00 00 00 0 Lost 0 Figure 4 1 Input from sound I O device window Use radio button to select Mono or Stereo input format Remember that you have to create or load a scheme for two independent channels to process stereo signal Frequency droplist contains possible values of sampling rate for STC H189 you may enter any value within 8 48 kHz range Default value is 11025 Hz Important The higher you set input sampling rate the larger your sound file will be At 11025 Hz one second of audio takes 22 Kbytes of disk space while at 22050 Hz 44 Kbytes In case of stereo input necessary space is doubled INPUT FROM A FILE 33 Current time counter indicates duration of audio fragment currently being inputted while Lost counter below shows overall length of frag ments omitted during current input and or ADC If you encounter audio losses please consult Measure mode in chapter 5 Input from a file This process opens specified audio file passes the information acquired from it to the next active module and controls navigation
18. Audio input module will acquire sound data directly from a sound editor while Audio output will pass the filtered signal back to the editor Main advantage of this mode is that Sound Cleaner may now get audio data from the sound editor i e all the file types and formats sup ported by the editor may now be processed by Sound Cleaner Bypass mode of WaveLab is not supported 95 96 CHAPTER 19 Sound Cleaner AS DIRECT X PLUG IN To run Sound Cleaner as Direct X plug in please follow these step by step instructions 1 Start Sound Cleaner and enter Direct X mode by pressing 2 on the toolbar 2 Run your sound editor and activate STC Sound Cleaner plug in using this editor s Direct X plug in activation procedure Important If you are using Adobe Audition you will have to select Ef fects Refresh Effects before running Sound Cleaner plug in 3 Start playback of the fragment in the sound editor then switch to Sound Cleaner and adjust processing scheme as necessary 4 Return to sound editor window to stop the playback You should try to avoid running several sound editors at the same time as it may lead to Sound Cleaner failures Chapter 20 Warranty The developer guarantees that the software conforms to the technical requirements whereby the user observes the conditions and regulations of operation storage and transport for a period of 12 months from the date of sale Tested and Approved
19. FILTRATION CONTROLS 77 with respect to their duration Decreasing this coefficient will enable the program to detect short and weak impulses impairing however lo calization of longer impulses Contrast works approximately the same as threshold but performs more subtle filter tuning Generally increasing the contrast will enable Sound Cleaner to detect more impulses It is usually adjusted in the end after appropriate threshold and smoothing values are selected Except these settings impulse filter is also supplied with common Reset filter and Restore default filter settings buttons located on the toolbar Chapter 16 Dynamic processing Dynamic signal processing improves its intelligibility if the signal frag ments greatly differ in level in case of resonant knocks i e long im pulses and room noises Dynamic processing algorithms improve and unmask the signal suppressing powerful impulses and clicks and reduce the listener s fatigue for long phonograms It is in some way similar to mu transformation see chapter 14 Mu transformation but works a little slower On the other hand dynamic processing has a major ad vantage over mu transformation it brings no distortions into the signal Dynamic processing controls Figure 16 1 displays dynamic processing window Toolbar contains common Reset filter and Restore default filter settings buttons Threshold set by the slider in the bottom of the win dow is very important It is
20. Fix button on the toolbar Filter coefficients will be set and used for filtering the entire phonogram without any adjustment Chapter 10 Adaptive inverse filtration Adaptive inverse filtration process is based upon Adaptive Spectral Correction algorithm sometimes called also Adaptive Spectral Smooth ing Adaptive inverse filtration effectively suppresses strong periodic noises from electrical pick ups or mechanical vibrations thus recover ing speech signal and equalizing signal AFC It amplifies weaker signal components and suppresses the stronger ones at the same time The av erage spectrum therefore tends to approach the flat spectrum unmasking the speech signal and improving its intelligibility Broadband noises however usually become stronger making signal perception less com fortable It means that you should try to reach a compromise between noise reduction and speech perception Adaptive inverse filtering controls Inverse filtration window is shown in Figure 10 1 Its toolbar contains four common filter control buttons 61 62 CHAPTER 10 ADAPTIVE INVERSE FILTRATION g Reset filter fix Fix current filter settings Ops E ae Open Auxiliary options dialog window DF Restore default filter settings 7 Fix ops DF Time constant 3 Max amplification 20 Invers threshold 0 MV Harmonics suppr Figure 10 1 Adaptive inverse filter window Time constant slider sets the time which the fi
21. RINCIPLES Open Windows list window Windows Windows list Activate scheme window Windows View scheme ae Project View scheme Load a saved scheme Project Load scheme Es Save current scheme Project Save scheme Processes Switch to Direct X plug in mode Go to help contents Help Help topics Sound Cleaner provides consecutive signal processing i e modules are applied to a signal consecutively one after one This chain like sequence is called processing scheme and may be saved in sch file Remember that modules included in the scheme may be inactive On the other hand windows of some processes no matter active or not may be hidden or minimized Table below displays names of processes their icons and short descriptions Name Icon Description File input Input signal from a file Device input Input signal from I O device BEX S Clipping Clip the signal Waveform Signal waveform Slowing Changing signal playback speed PROCESSES 29 Amplifying Signal amplification Save Save signal to file Equalizer Parametric equalizer Mu transform Signal transformation u law Duplicate Splitting a signal in two Speaker Signal playback StereoWave Stereo waveform of a signal Freq compensation Frequency compensation Impulse filter Adaptive impulse filtration Inverse filter Adaptive inverse fi
22. Sound Cleaner software serial number conforms to technical requirements and documentation and is declared suitable for use Adjustment conducted by Date of issue 97 98 CHAPTER 20 WARRANTY Support Our developers are always ready to assist you In case of any questions please don t hesitate to contact us Tel 7 812 331 0665 Fax 7 812 327 9297 E mail info Ospeechpro com Web site http www speechpro com
23. aner and Wave Assistant System requirements Sound Cleaner s PC configuration demands are rather high due to real time audio processing For proper operation it requires e Pentium 111 1000 MHz or better CPU e At least 25 Mb of free disk space e MS Windows 95 98 2000 XP OS depending on sound I O device used see Installing sound I O hardware e At least 128 Mb or more RAM 15 16 CHAPTER 2 PREPARING Sound Cleaner FOR OPERATION e One free USB or PCI slot depending on sound I O device used e CD drive e Color monitor with 1024x768 dpi or better resolution e Keyboard mouse During the operation especially audio input and saving data in a file Sound Cleaner software occupies nearly all the system resources CPU time memory local BUS flow and DMA channels That s why we advise that you close all other applications while running Sound Cleaner in order to avoid possible loss of signal fragments Important Please note that Sound Cleaner has a high priority when ad dressing sound I O device This means that if there is some other application occupying the same device it may lose access to the I O device after you shut down Sound Cleaner Installing sound I O hardware Sound Cleaner may receieve signals from one of three different sound T O devices STC H189 STC H216 and Sound Blaster Each of these devices has certain OS limitations namely STC H189 operates only in MS Windows 98 while STC H216
24. appropriate process ing scheme Typical schemes Load Standard scheme of 2 channel signal processing or create a two channel scheme yourself Then you may use any of the described processes for each channel independently If you have used different filters or different parameters for these two channels you will have to include the Stereo Wave process into your scheme 81 82 CHAPTER 17 STEREO PROCESSING StereoWave This process is used to display audio data from both processed channels in a single window It is absolutely similar to Waveform see chap ter 7 The only exception are the Hide left Hide right checkboxes above the waveform which allows to hide a waveform of a signal ac quired through particular channel Adaptive stereo filtering As in the previous case you will have to create or load a scheme where one of the adaptive stereo filtration modules is present There are two appropriate typical schemes Typical schemes Load Stereo filtra tion for filtration in time and frequency domain The only difference between these two schemes is the type of active stereo filter using them together for the same signal makes no sense anyway Two channel adaptive filtration algorithms are designed to suppress both unstationary broadband background speech radio room noise and periodical vibrations power line pick ups noises The backbone of these methods is acquiring the information about the interference from the refer
25. at the right side 2137 SLOWING Figure 12 1 Slowing process window Speed adjustment is usually employed in the very end of the scheme just before the playback of processed sound 69 Chapter 13 Clipping This process is necessary to limit amplitude of a signal thus removing strong bursts and smoothing the signal s level On the other hand it does not significantly affect speech intelligibility Clipping is also very useful for rough clearing of harmonic interferences provided there are rather long fragments of interference and its level exceeds the level of useful signal lt 16 gt CLIPPING MIN 27249 Figure 13 1 Clipping process window 71 72 CHAPTER 13 CLIPPING To clip the signal you just have to point minimum and maximum signal level values using MIN and MAX sliders in the process window Figure 13 1 Current value is shown in a field above the slider Then all the values of signal level which are higher than maximum and lower than minimum value are discarded and made equal to respec tive value Chapter 14 Mu transformation Mu transformation is recommended if signal fragments greatly differ in level It happens for example if one of the speakers is located near the microphone and the other one quite far from it During the trans formation weak fragments are greatly amplified while strong ones are amplified insignificantly or not amplified at all This process is in fact similar to dynami
26. c processing described later in chapter 16 but works faster Mu M x 27645 Figure 14 1 Mu transformation process window 73 74 CHAPTER 14 MU TRANSFORMATION There are two parameters of mu transformation see Figure 14 1 Mu MAX defines highest possible level of signal after the transfor mation 1 32000 Contrast value sets the number of mu transformation coefficients and defines signal level range to be transformed with maximum ampli fication 1 value means that all the input signal will be evenly amplified and brought to maximum during transformation 1000 value will on the contrary greatly amplify the weak signals reducing at the same time the level of strong ones Chapter 15 Impulse filtration Adaptive impulse filtering automatically restores speech or musical fragments distorted and masked by various pulse interferences such as clicks radio noises knocks and so on Adaptive impulse filtering algorithms improve quality of the signal suppressing powerful signal impulses and thus unmasking useful audio signal and increasing its in telligibility Impulse filtration algorithm During impulse filtration Sound Cleaner substitutes impulses with smoothened and weakened interpolated signals If the program does not detect an im pulse it leaves the fragment unchanged It also does not suppress tonal interferences and broadband noises Impulse detection is based upon the information which the program has about differen
27. ces between useful signal and an interference Thus setting filtration parameters correctly is critical for effective processing 75 76 CHAPTER 15 IMPULSE FILTRATION Impulse filtration controls Impulse filter window is shown in Figure 15 1 lt 7 gt Imp Filter 7 OF Threshold m Contrast 0 S Smoothing Figure 15 1 Impulse filtration process window The process may work in two modes switched by the radio button at the bottom edge of the window Impulse hindrances in this mode short up to 100 samples im pulses will be smoothened and removed being substituted with values interpolated from the useful signal Amplitude distortions works for the the same short impulses as well as for distortions due to overload if for example ADC limita tions were violated during audio input In this mode the distortions are smoothened so you will not have to adjust any processing parameters Impulse filtration is configured with three sliders Threshold is necessary to detect and locate the impulses basing upon their energy Default value is 6 Decreasing detection threshold will make the program locate weaker impulses Setting higher threshold values will leave the weak impulses intact if the value you have set is too small then sudden changes of useful signal level will be considered impulse interferences and removed Smoothing slider controls detection and localization of impulses IMPULSE
28. cessing To activate the process press this button and see it turn red which indicates that the process is active Remember that all the windows no matter active or not may be also 22 CHAPTER 3 BASIC PRINCIPLES hidden closed minimized or displayed on the screen For more de tails on managing process windows see description of Windows menu below Menus Menu string is located just below the program header Menu commands control the program in general Sound Sound menu contains commands which manage input and output of audio data Open file Choosing this menu item will open standard open file di alog window Specify the file you wish to open and process as a signal source Save to file You may choose or create a file to save the processed audio signal into Start sound input Choose this item to start input from sound I O de vice Play file This command starts sound input from a specified file Stop Stops sound input Pause Continue Use this command to pause the playback and then continue audio input Global reset This command restores all the default filter process set tings and removes all the information accumulated in buffers Cancel saving file This item stops saving data in a file but does not affect other processes MENUS 23 Make composite stereo Choose this command to create a composite stereo 1 e stereo signal combined from two independent mono signals See Processing composi
29. e chosen with respect to one important rule primary channel interference should never anticipate the reference channel For exam ple if primary and reference signal are simultaneous setting positive delay will make correct filtering impossible Introducing on the con trary small negative delay will lead to reference signal anticipating the primary which is necessary for correct filter adjustment and effective processing In most cases delay should be close to 0 or a little bit less But if the reference signal is recorded from an interfering device output setting ADAPTIVE STEREO FILTERING 85 huge positive delay may be necessary Note that 1 count delay at 8000 Hz sampling rate is equal to 2 3 cm propagation difference i e difference between distances the signal has to cover in order to reach primary and reference microphone Thus 1 m propagation difference will demand about 40 counts delay for good processing results Stereo filtering in frequency domain Frequency spectral filtration differs from filtration in time domain only in processing methods and conditions Delay for the frequency filtration is set in centimeters not in counts It represents the difference of distances between microphones and interference source In all other aspects delay has the same meaning and is set the same way as for time filtering Frame size background output and adaptation speed also work exactly as described in previous section Frequency
30. ech FORMAT MEAN Mono Stereo File Writing 540672 Figure 5 2 Save process window Save sound to file button opens a dialog window where you choose file and path to save a phonogram into It will be displayed just below SAVE TO FILE 41 the toolbar After you have specified target file start the input and the signal will be saved To temporarily stop saving signal press Pause button Close file button will close the file so you won t be able to continue recording into it Finally you may view and edit this file with Wave Assistant by pressing Export file to Wave Assistant Chapter 6 Amplifier Amplifier enables you to increase or decrease signal level up to 60 dB Its window is shown in Figure 6 1 lt 10 gt AMPLIFYING Figure 6 1 Amplifier process window Gain coefficient is adjusted by the slider with 1 dB step current value is displayed at the right side of the bar 0 dB value means that signal is not amplified During signal processing its level may go down due to corrections of frequency characteristic and noise reduction It weakens some com ponents of a signal which may be then lost during further processing That s why it is reasonable to amplify a signal during audio processing For most effective amplifying Waveform module is very useful We advise that you place it in your scheme right after the Amplifier to con 43 44 CHAPTER 6 AMPLIFIER trol the gain If waveform of your signa
31. ected phonogram fragment in loop mode Play a phonogram in pseudo stereo mode with adjustable channel delay Optimize amplitude and frequency of a signal providing best pos sible audio perception Operate as Direct X plug in for most common sound editors Enter text using built in text editor Automatically create text report of signal processing Save processing parameters for later usage Technical characteristics Main technical parameters of Sound Cleaner are given in the table be low Number of processed channels 1 2 Processed signal format 8 or 16 bit PCM u law 24 bit float Input signal format 16 bit PCM Saved signal format 16 bit PCM Sampling rate 8 48 kHz chosen by user 14 CHAPTER 1 OVERVIEW Audio file types supported WAV DAT SDT Maximum phonogram duration Limited only by free HD space depends on signal sampling rate Largest possible number of adaptive filter coefficients 8192 Gain adjustment range From 60 to 60 dB Duration of fragment played in loop mode From 0 5 sec up to whole phonogram Playback speed coefficient 0 7 3 adjustment range Channel delay value range 0 20 msec for pseudo stereo playback Chapter 2 Preparing Sound Cleaner for Operation To prepare the software for operation you should 1 Install sound I O hardware and its drivers if necessary 2 Install Sound Cle
32. ence channel and the using it to remove this interference from the primary channel Stereo filtering has however a serious limitation it may be effec tively applied only if following conditions were met during the audio record e Stereoscopic base 1 e distance between the microphones should not be less than 10 cm e Primary and reference channel microphones should be located at different distance from the signal source The same is true for the source of interference ADAPTIVE STEREO FILTERING 83 e Both useful and interfering signal received by microphones should come directly from the source not be reflected Stereo filtering in time domain Time filtration allows to suppress unstationary broadband noise speech music and periodical interferences vibrations power line pick ups from some point like source e g radio set Its process window is shown in Figure 17 1 lt 14 gt Time stereo SO O FX F L R Frame size 400 Z 500 Background output 600 z Adaptation speed 20 30 26 Ref channel delay cnts 30 1000 161 Figure 17 1 Time filtration process window Toolbar contains common Reset filter Fix filter and Set default filter values buttons There are also two additional L and R buttons necessary to specify which channel should be considered primary L stays for left channel and R for right Frame size should be selected from the list and may vary between 50 and 2000 counts We reco
33. eport There is only one Create option in this menu If you check it the pro gram will offer you to store processing parameters and current configu ration in a text file each time you stop the playback If you agree current scheme and configuration will also be saved in the same folder Register There are two available items How to register This command will display information explaining how to register your copy of Sound Cleaner software TOOLBAR 27 Register Launches registration wizard see Registration in Chapter 2 for details Help Help topics This command will open Sound Cleaner help file About Sound Cleaner Brings forth brief information about Sound Cleaner About STC Displays brief information about Speech Technology Cen ter Toolbar Sound Cleaner toolbar buttons are shortcuts to most frequently used menu commands Start audio input from I O device same as Sound Start sound input Start audio input from a file Sound Open file Open two mono files and combine them into a composite stereo signal Sound Make composite stereo Save Load program configuration Options Save Op tions Load Restore default parameters for all modules and clear all buffers Sound Global reset Pause continue signal processing Sound Pause Continue y Qe Start stop direct listening to source audio signal without RK processing T Open text editor window Windows Text edit 28 CHAPTER 3 BASIC P
34. ering works best if you have to get reliable results very fast It also does not demand from the user any special noise filtering skills and experience All you have to do in this mode is specify desired noise reduction level soft standard or hard Each level has also its own inner scale so in total you have 10 grades of noise cancellation Just choose appro priate radio button and the program will automatically select optimum values of noise suppression Intensity and Depth Note that filter needs some time to tune itself to a specific noise so you should try to avoid making considerable instant changes of noise reduction level You may also choose Additional depth of noise reduction within 15 15 dB range and enable Frame size auto tuning These settings will be described later in this chapter Manual mode For experienced users who are not completely satisfied with results of automatic processing there is a possibility to tune all the necessary parameters manually To do it just select Manual settings in Noise reduction degree group This will automatically open Auxiliary op tions window Figure 9 2 Main and most important parameters of broadband filtering are sup pression Intensity and Depth located in Manual settings group Suppression intensity adjusts inflection point of SNR suppression curve 1 40 Suppression depth may be set within 1 80 range it deter mines largest possible suppression of spectral components of a signa
35. g scheme types Single channel schemes are used for mono signals Two channel schemes are designed to process signals in left and right channel independently Stereo filtering scheme is very effective for stereo signals in most cases To help you master Sound Cleaner faster and easier we have in cluded a set of standard noise cancellation schemes in it These schemes provide good results for typical noises and interferences in most cases In case if Sound Cleaner modules won t suffice and you will need additional tools you may wish to employ Wave Assistant signal editor supplied with Sound Cleaner Currently processed signal will be auto matically loaded in Wave Assistant if you launch it during processing The programs are linked so that if you edit a fragment in Wave Assis tant the changes will be transferred to Sound Cleaner Input module and vice versa To learn more about Wave Assistant signal editor refer to its manual Sound Cleaner main window and process windows Figure 3 1 displays main window of Sound Cleaner program As you see it is a standard window its header displays program name and version number There is a common menu string just below the header and a toolbar which contains shortcut buttons to most fre quently used menu items There is also the messages window at the very bottom while all the main part of the window is occupied with process windows As Sound Cleaner loads you will see all
36. gnal speech quality Except frequency compensation this process also provides adaptive compensa tion in time domain It also suppresses harmonic interferences provided their phase is more or less stable Adaptive compensation algorithm Adaptive compensation enables user to remove both narrowband sta tionary interferences as well as regular ones vibrations power line pick ups electrical device noises steady music room car and water noises reverberation and so on One channel adaptive frequency compensa tion recovers the speech suppressing tonal interference by 20 40 dB Main advantage of this process is its capability to preserve the speech signal much better than other filters usually do It happens because the interference is in this case subtracted from a signal and not multiplied by 0 In some cases only a part of periodic interference may be removed 65 66 CHAPTER 11 FREQUENCY COMPENSATION so you may use adaptive compensation in your scheme more than once Primary compensation parameters which affect noise suppression level are the number of filter coefficients defined by frame size and delay value Increasing number of coefficient allows to suppress more spectral interference peaks at the same time lowering filter adjustment speed Delay should not be set lower than half the number of filter coefficients Frequency compensation controls Figure 11 1 displays Frequency compensation window lt 4 gt Fr Compens
37. he scheme will open its context menu which contains a list of available operations with scheme 92 CHAPTER 18 PROCESSING SCHEMES elements Activate De Activate These commands make the process active or pas sive Delete Remove the process from current scheme Show Hide Minimize Standard process window operations Create Delete connection After you select one of these commands the cursor will change to N Then just click it over the icon to be connected or disconnected This operations may be necessary 1f automatic connections were made in a wrong way Transpose inputs Swap incoming channels for a selected process Duplication Sometimes it is necessary to duplicate the signal and process it in two different modules independently Most evident example is saving pro cessed signal to file playing it at the same time to evaluate processing results For such cases there is an additional Duplicate module Being placed in a scheme it receives a signal and then passes it to two different modules Typical schemes Typical schemes included in Sound Cleaner demonstrate its performance when dealing with different types of noises and interferences It may also help an unexperienced operator to master the program As you choose the scheme the program will automatically load preset filter set tings and an example of signal to be processed You may configure the program so that no example will be loaded see Options in Basic principles
38. icking on these arrows will scroll the scheme and right clicking available icons At the right edge of the window there is a column of scheme control CREATING AND ADJUSTING THE SCHEME 91 buttons They correspond to commands of the system menu you may open it by clicking over the icon in the window header El Load scheme Choose an existing scheme and load it El Save scheme Save current scheme a Change scale View the scheme in smaller scale It is s useful for large and complicated schemes which do not fit the regular window Redraw scheme Automatically redraw the scheme opti A mizing its structure Before you redraw the scheme ensure that there are incoming connections for all the processes except sound input Erase scheme Clears currently loaded scheme Change style This button minimizes scheme window hid ing all the icons and control buttons To return to default window style activate system menu command B Ly ms Swap input channels Swaps incoming channels for a stereo signals Creating and adjusting the scheme Current scheme is displayed in the scheme window To add a new pro cess find its icon in the column of available processes and drag it to desired location in the scheme The program will automatically connect the process to its neighbors in the scheme Do not forget that there may be only one sound input and one output process in any scheme Clicking right mouse button over an icon in t
39. l Generally speaking increasing both these values will lead to better noise reduction but useful signal may also be suppressed and speech 58 CHAPTER 9 ADAPTIVE BROADBAND FILTERING Auxilary Options m Manual Settings Frame size Suppression Intensity 1 40 15 Suppression Depth 0 80 24 Time constant 3 T Smoothing 0 9 5 Harmonics H Suppress E Timbre correction JV ON OFF Speech from 200 LF amplify dB o Speech to Hz 3600 HF amplify dB 1 0 Cancel Figure 9 2 BB filter auxiliary options window quality and intelligibility reduced We advise that you adjust these pa rameters very carefully and always control achieved effects Time constant field sets the time of filter adjustment to signal spec trum variations In most common cases 3 4 sec is recommended for non stationary noises the value should be slightly decreased to 1 2 sec onds Setting less than 1 second time constant will in most cases greatly decrease speech quality T smoothing parameter controls time smoothing of filter coefficients It usually helps to remove musical noises which sometimes appear af ter broadband processing For large SNR values and signal sampling rate less than 11025 0 1 values should suffice while bad SNR and huge sampling rates may demand T smoothing of approximately 5 Default BROADBAND FILTERING CONTROLS 59 value is 1 In the Frame size list you may choose s
40. l nearly fills the waveform win dow in Y direction the gain is set correctly Too low gain may lead to gaps in useful signal while over amplified signal will be distorted during playback If the gain value you set is too high overflow indicator ls will appear to the left of the slider bar Click left mouse button over this indicator to remove it Note that over amplifying will result in discomfort during playback and in some cases may even cause ear injury especially if a listener uses headphones for signal playback So you should avoid making sig nificant gain adjustments instantly Also remember that deactivation of processing modules may sometimes greatly increase signal level For such cases there is Dependent option in Amplifier s system menu De pendent amplifier will become active only if the process which stays in the scheme directly before the amplifier is active Otherwise amplifier will be automatically rendered inactive You may open it by clicking on Microsoft icon in the window s header Chapter 7 Waveform This process is designed to view signals in waveform and monitor their level It is usually employed in a scheme at least twice first time to view just inputted signal and then after all the processing to control it before playback 227 WAYEFORM Figure 7 1 Waveform process window 45 46 CHAPTER 7 WAVEFORM On Figure 7 1 you may see a waveform of a signal displayed in the process window C
41. lose the window saving all the changes Cancel will discard them ZOOMING AND SCROLLING 53 Zooming and scrolling To zoom signal spectrum window in and out you may use appropriate toolbar buttons see Equalizer toolbar lt gt buttons and horizontal zoom bar located below the spectrum area Blue marked fragment of this bar indicates which part of the whole spectrum is currently dis played If the whole bar is blue then all the spectrum is displayed if for example button was pressed Click left or right mouse button over the bar to set respective borders of displayed spectrum You may also scroll it with arrow buttons to the left of the bar or with arrow keys In the latter case you have to set the focus on the bar Pressing the button or key moves the displayed area 1 16 1 32 for large window part of the whole signal area to the right or left Adjusting filter FC There are 16 spectrum adjustment sliders in standard window and 32 in the large Current signal level adjustment value in controlled band is indicated above each slider in dB Highest and lowest possible values 20 72 dB which correspond to extreme slider positions are given near the left edge of the window There you may also see a number of spectral bands which are currently controlled by a single slider If you zoom in and out this number will change reaching 1 at maximum X axis zoom level Elastic mode Elastic mode enables you to sim
42. losely 30 70 cm located microphones from a nearby less than 1 5 m source removing reflected noise of a distant source air condi tioner orchestra and so on It works best if the interference source is located symmetrically relative to the pair of microphones If not you may correct the symmetry with appropriate delay setting PROCESSING COMPOSITE STEREO SIGNALS 87 Processing composite stereo signals In Sound Cleaner there is a possibility to create a composite stereo signal by combining two different mono signals into one It may help you to apply adaptive stereo processing to signals recorded by two different microphones To be merged in a composite stereo mono signals must be of the same type wav dat and recorded with the same sampling rate If these signals have different length then duration of resulting composite stereo will be equal to the length of the shorter signal To create a composite stereo press on the toolbar of the main Sound Cleaner window The program will display a dialog window Figure 17 3 Making Composite Stereo Signal b xi FILE 1 C Program Files Speech Technology Center Sound Cleaner SPECIAL S Browse FILE 2 C Program Files Speech Technology Center S ound Cleaner SPECIALAS OK Cancel Figure 17 3 Composite stereo creation window You will have to select files to be included in a composite stereo If they meet all the necessary conditions you may load the signal and process it as
43. lter needs to adjust to variations of the spectrum within 0 1 10 seconds range In common ADAPTIVE INVERSE FILTERING CONTROLS 63 cases 3 4 sec values are recommended for non stationary noises de creasing this value to 1 2 seconds may be useful Max amplification limits weak spectrum components gain for any given frequency It is necessary to avoid raising the level of noise during the pauses 20 30 dB values are recommended Inversion threshold value marks certain level of the signal which is considered the border between weak and strong i e all the com ponents below that level are amplified while those that exceed it are weakened Ops toolbar button opens Auxiliary options dialog window Fig ure 10 2 You may choose frame size from the list box and config ure Timbre correction parameters Timbre correction settings are de scribed in Manual mode section of Adaptive broadband filtering Auxilary Options 3 Frame size m Timbre correction MV ON OFF Speech from 200 LF amplify dB le Speech to Hz 3600 HF amplify dB 10 Cancel Figure 10 2 Adaptive inverse filter auxiliary options window Chapter 11 Frequency compensation Adaptive compensation process uses Widrow adaptive filtering algo rithm for one channel adaptive compensation It is most effective for narrow band stationary and regular interferences The filter adjusts it self smoothly maintaining good useful si
44. ltering Broadband Filter Broadband filtering of a signal Dynamic processing LCD le E 8 Le Los Lo De Signal dynamic processing Frequency stereo Frequency stereo filtration Time stereo a Time stereo filtration Chapter 4 Audio input This chapter describes audio input process There are two types of input process depending on signal source Input from sound I O device and Input from a file Since Sound Cleaner may work only with one signal source at a time only one kind of audio input may be active and they are united in the single Input module Process window will change according to type of signal source selected Obviously Input should always be placed first in any processing scheme Input from a sound I O device ADC Main task of this module is to input audio signal from an external sound source To do it you should connect linear output of playback equip ment to linear input of sound I O device or if sound is received from a microphone connect it to the microphone input of the device During the input signal is converted into digital form and passed di rectly to Sound Cleaner Processed signal is transferred to linear output of the I O device and may be at the same time saved via Save in a file process But much more effective way is to save the unprocessed signal on a hard drive or any other media to do it activate only Input from 31 32 CHAPTER 4
45. ly done during inverse filter calculation and or au tomatic filtration In this case pressing this button will make Sound Cleaner to contrast the filter once more Automatic filtration button turns off spectrum accumula tion and then inverse or harmonic filter calculation filter type is selected by user You may set time of automatic spectrum accumulation in Options menu Switch between standard and large equalizer window EQUALIZER OPTIONS 51 This indicator button turns red if a signal was over amplified which caused an overflow during equalizer out put Press the button to bring the indicator back to passive state until the next overflow Equalizer options If you click your left mouse button over Microsoft sign in the header of equalizer window its standard system menu will appear It has how ever additional Options entry that opens Additional options dialog window see Figure 8 2 Bands Number Filter Contrasting Options Discrimination Lewel 0 1 os Analysis Window Width Hz 70 High Intensity Iv Accumulation time sec f Filter Contrasting Iv Graphics Drawing Iv OK Cancel Figure 8 2 Equalizer options window There you may choose number of bands for the equalizer Gener 52 CHAPTER 8 EQUALIZER ally speaking setting greater number of bands will increase filter per formance as well as your PC load Filter contrasting options field occupies the center of the wi
46. mands of this menu manage Sound Cleaner process windows Next Activate next window Previous Activate previous window Windows list Opens Windows list dialog window Figure 3 2 AAA Process Properties lt 2 StereoWave Select 3 gt EQUALIZER 8192 T Minimized 4 EQUALIZER 8192 Delete p 5 Inv Filter 2048 Iv Visible lt B gt Inv Fiter 2048 Replace alero lt 7 gt B B Filter 512 lt 8 gt BB Filter 512 9 AMPLIFYING Insert lt 10 gt AMPLIFYING lt 11 gt StereoWave 14 SPEAKER SIES SAVE Place All Cancel x x x XK Xx ikete Figure 3 2 Windows list dialog window MENUS 25 In the left half of the window there is a list of processes currently included in the scheme Active modules are marked with an aster isk Left click on any module to select it Process properties field at the right side will display information about its current status You may place a flag there to minimize hide or activate the module There are also control buttons in the centre They perform vari ous operations on the selected process Select button will unfold the window of selected module and place it atop all others you may do the same if you double click on any process in the list Delete button will remove the process from the scheme Pressing Replace or Insert will open a list of available modules Double click on any one of them to place it in the process list either in stead
47. mmend you to listen to results after ev 84 CHAPTER 17 STEREO PROCESSING ery frame size change and then make further adjustments if necessary Frame size defines number of filter coefficients large values usually slow down filter adjustment but provide better filter performance espe cially when removing reverberation noises Background output is used to monitor and control filter perfor mance If you place this flag the program will play back the signal which with current filter settings is considered noise and removed If all the settings are correct background output should contain only in terference signal Nevertheless if you hear useful signal in the back ground there may be two reasons for it 1 You may have swapped primary and reference channel In this case all you have to do is just choose the other channel as primary with L R toolbar buttons 2 Useful signal is partially removed It means that reference chan nel microphone receives not only interference but also useful sig nal To solve this problem try moving reference channel micro phone if possible to minimize useful signal received by it Adaptation speed slider defines filter auto adjustment speed For unstationary noises 26 29 values are recommended while for slow vary ing ones 22 25 should suffice Reference channel delay is a parameter critical for effective stereo processing It is set in counts and lies within 30 1000 borders Delay should b
48. ndow Contrasting means that Sound Cleaner will automatically detect narrow gaps in the filter FC and then broaden and deepen them This operation may considerably improve filtering quality especially if there are clear local noise peaks in the spectrum of a signal To enable filter contrast ing check the respective flag in the bottom part of the window Then adjust contrasting options e Discrimination level parameter represents relation between filter value in the gap and on its edge The program uses it to determine which filter gaps are to be contrasted 1 value means that it will contrast all the local minimums 0 filter won t change Values around 0 5 will make the program to skip small usually natural minimums at the same time contrasting large and distinct gaps e Analysis window width value determines maximum width of a gap in Hz which will be considered narrow and therefore will be subject to contrasting Default value is 70 Hz e High intensity flag when checked will slightly broaden the gaps in addition to common contrasting effect At the lower part of the screen you may see a group of additional controls Accumulation time value is a duration of spectrum accumu lation for automatic filter calculation Filter contrasting flag turns on respective operation as was mentioned earlier Graphics drawing may be turned off to disable drawing of signal spectrum enhancing perfor mance of weak PCs OK button will c
49. of the selected one Replace or before it Insert Finally Place All button will return all the windows to default position and status If you have used any of these commands the dialog box will be automatically closed Otherwise press Cancel button to return to Sound Cleaner main window Place All Restores default position and status of the process windows Hide inactive This command will hide windows of all currently inac tive modules Minimize inactive This item will minimize all windows of currently Inactive processes Hide minimized Use this command to hide minimized windows Text edit Opens a built in text editor window View scheme This item activates sound processing scheme window 26 CHAPTER 3 BASIC PRINCIPLES There is also Options submenu where you may set default oper ations with windows Possible choices are Hide inactive Minimize inactive Hide minimized and Windows standard Project Commands of Project menu control and adjust signal processing schemes Load scheme Choose and load existing processing scheme Save scheme Save current scheme in a file specified by user View scheme Activate Scheme window Load scheme config Save scheme config These commands load and save processing scheme and configuration file associated with this scheme Activate all Activates all the modules in the currently loaded scheme Deactivate all Makes all the modules in the current scheme inactive Text r
50. pectral resolution 1 e num ber of spectral bands and size of a signal processed audio block As this value is increased more different sounds are mixed together in a single block so there are fewer spectral bands containing no speech signal Larger frame size tends to produce considerable echo effect On the other side smaller values lead to weaker and less precise noise reduction Available Frame size range is 128 4096 with 512 being the default value In fact it depends on sampling rate of a signal but you may check Frame size auto tuning in the filter main window to make Sound Cleaner to choose frame size automatically Harmonic suppression flag turns on additional suppression of har monic noises Timbre correction is necessary to remove unnecessary low and high frequency components of a signal i e configure the bandpass so that it would suit human speech frequency range In this case speech becomes more intelligible and comfortable to listen to You may turn timbre correction on and off with ON OFF flag and adjust its parame ters Speech from sets bandpass lower border and Speech to the upper border For 10 11 kHz sampling rate 200 Hz lower and 3600 Hz upper borders are recommended In fact these values depend on signal quality and sampling rate As a rule you should somewhat lift the upper border value If you encounter strong broadband noise hisses rumbles it is reasonable to decrease this value to 2900 3200 Hz
51. se this will usually lead to considerable signal distortions it is much more reasonable to adjust signal level with Amplifier see chapter 6 There is also MSR button above the Delay slider Use it to activate Measure mode and calculate current PC load Pseudo stereo mode In pseudo stereo mode a signal is played in one of two channels and then the same signal is played in the other one with a little delay This makes audio perception more comfortable and increases signal intelligi bility moreover listeners tend to tire slower when pseudo stereo mode is used It is especially useful for determining text contents of very long phonograms In such case you should also try use different processing types and change signal level from time to time If you wish to find more useful information about the properties of PLAYBACK 39 human hearing important for perception of noisy speech phonograms as well as about main principles of determining contents of phonograms refer to STC Noise Cancellation and Text Decoding of Low Quality Speech Recordings Practical Approach Measure mode As you turn on measure mode audio playback is paused and automat ically calculated V value is displayed next to MSR button This value represents time spent for signal processing compared to total phonogram duration in percents V monitors PC load if its value exceeds 100 it means that your PC can not process the signal in real time which in turn may res
52. signed to suppress broadband and periodic noises due to electric pick ups or mechanic vibrations room and street noise communication channel or record equipment interferences You may hear these noises as hum rumbling hisses or roars Note that this algorithm performs at SNR not worse than 5 Hz It is nearly impossible to remove such noises with other methods such as one channel adap tive filtration spectrum smoothing or equalizer because they are spread across the whole spectrum and intersect with speech signal Broadband filtering controls Broadband filter window is shown in Figure 9 1 There are four process control buttons located on the toolbar 55 56 CHAPTER 9 ADAPTIVE BROADBAND FILTERING 4 gt B B Filter 512 2 Fx ops DF Noise Reduction Degree Soft Standard Hard eee EJE oe wee Manual Settings Additional Depth dB p Smoothness m Frame size auto tuning Figure 9 1 Broadband filter process window z Reset filter Fix current filter settings Ops di iO Open Auxiliary options dialog window DF E Restore default filter settings Noise reduction degree group enables you to either specify level of automatic filtering or use Manual settings for more subtle filter adjust BROADBAND FILTERING CONTROLS 57 ment Additional depth available only in automatic mode Smooth ness and Frame size auto tuning fields will be described later Automatic mode Automatic noise filt
53. software e STC Sound I O device optional Sound Cleaner capabilities This manual describes Sound Cleaner software For details on Wave Assistant software and sound I O devices see respective manuals and or technical descriptions Sound Cleaner software is capable to e Input and save audio signals from microphones and line inputs of playback devices e Suppress different kinds of noises and compensate various inter ferences including gt stationary and slowly varying additive complex narrow band polyharmonic and broadband noises gt slowly varying amplitude frequency interferences record or transmission channel AFC inconsistency gt short pulse interferences gt signal jumps gt any kinds of additive noises provided that audio signal is received through two channels Instead of Sound Blaster card or other windows compatible multimedia device the software may be delivered with STC H216 external USB device If Sound Cleaner is a part of STC Ikar Lab complex it will use STC H189 sound I O card TECHNICAL CHARACTERISTICS 13 Process signals in real time mode Waveform and instant spec trum of a signal before and after the processing are continuously displayed You may always adjust any of the processing param eters no matter which kind of filter is used instantly monitoring and listening to the effect of changes you have made Adjust playback speed smoothly without affecting sound pitch Play sel
54. st follow the instructions of registration wizard You will be asked to enter your User name and Registration code both of them are to be found in register inf file in Sclean folder on the CD If there is no User name specified you may enter your own name in this field Chapter 3 Basic principles of Sound Cleaner operation General information The program operates much like a mosaic this means you may design a scheme of signal processing combining any of the provided mod ules processes in a desired sequence starting from sound Input Each process included in the scheme receives a signal form the previous one applies his own algorithm and then passes the processed signal to the next module in the chain Number of modules used consequently in a single scheme is limited only by capabilities of your PC Each module is represented by an icon in the Scheme window located in the left part of the Sound Cleaner main window see Figure 3 1 and has 1ts own dialog window where all the information and necessary controls are located Name of the process is indicated in the header of its dialog window The header also holds standard minimize and close see Figure 3 1 buttons as well as activate button This button is used to apply the module to a signal activate it or stop using it for signal processing deactivate 19 20 CHAPTER 3 BASIC PRINCIPLES Depending on signal format and type it may be processed along one of the three processin
55. stereo filtration window is shown in Figure 17 2 Process toolbar is similar to time filtration window the only differ ence is Options button which opens frequency filtering auxiliary op tions dialog window In this window you may configure standard tim bre correction settings and set maximum suppression depth There are three available modes of Frequency filtration They are selected with the radio buttons Point source noise mode works approximately the same as time filtration but with better speed because of larger frame size values On the other hand the filter adjusts itself to an interference a little slower It is very effective when the reference signal is recorded from a line output of interfering device e g radio set In this case you should set appropriate positive delay because the source is actually located in some distance Reference channel noise mode removes noise of some distance source working engine background speech from the primary chan 86 CHAPTER 17 STEREO PROCESSING Fx ops DF L R Frame size Cancellation of 256 a Point source noise 312 Ref channel noise C Ambient noise Background output Adaptation speed 20 30 25 Figure 17 2 Frequency filtration process window nel provided the reference microphone is set next to this source You will also have to set an appropriate positive delay Ambient noise mode allows to clear useful signal if it is received by two c
56. te stereo signals on page 87 for more details Switch input channels Swaps left and right channels of incoming stereo signal Options Commands in this menu set some additional parameters of saving and loading files in Sound Cleaner Save Use this command to specify a file where current Sound Cleaner configuration will be saved Load There you may choose a file containing saved program configu ration and load it Details submenu contains three items Save file input options when saving to cfg file If you check this option with the flag the program will store name of cur rently loaded audio file and loop settings in configuration file Ignore file input options when loading from cfg file Check this item to ignore audio file saved in configuration file Load sound file example together with typical scheme If this option is checked Sound Cleaner will automatically load a sound sample associated with particular scheme Typical schemes This menu contains three submenus for managing typical schemes 24 CHAPTER 3 BASIC PRINCIPLES Load You may choose a typical processing scheme from a list and load it Save as This item saves current scheme and configuration as a typ ical scheme See Typical schemes section on page 92 for more details on creating typical schemes Delete item Choose this command to delete a typical scheme as well as sound sample and configuration file associated with it Windows Com
57. ult in phonogram fragments being omitted If you click on the displayed value calculation will be started over from this particular instant To decrease PC load and minimize audio loss probability if neces sary you should try to e Deactivate modules which provide visual presentation of a signal Waveform Equalizer e Decrease number of bands in Equalizer and frame size for adap tive filters e Decrease sampling rate e Use more fast and productive PC Saving signal to analog media To save processed signal or its fragment on an audio tape or other ana log medium you should first connect record device tape recorder line This brochure is delivered with STC Ikar Lab It is also available for other Sound Cleaner users please contact us for details 40 CHAPTER 5 AUDIO OUTPUT input to line output of your I O device Then adjust all the processing parameters and signal level Finally you should simultaneously press Start input button and record button of a tape recorder Record will go on until you stop or pause it Save to file To record processed signal in a file you should use Save module It may be saved as WAV or SIS file in mono or stereo 16 bit PCM format with a sampling rate previously used for signal input Stereo signals are saved as WAV files only Toolbar of the Save process window Figure 5 2 contains common record control buttons lt 15 gt SAVE 8 x le n m P C Program Files Spe
58. ultaneously adjust several sliders as if they were bound together with an elastic thread In this mode it is much easier to change filter FC smoothly 54 CHAPTER 8 EQUALIZER To include sliders in an elastic group you should click left and right mouse button on the bar below the sliders thus setting left and right border of the group Selected group will be marked with blue indicator in the bar Note that only sliders not signal bands may be grouped and bound together This means that if you zoom in or out those sliders which you have previously included in a group will control other signal bands x button to the left of the elastic mode bar turns linearization smoothing on and off Outlines of a filter built in elastic mode will be smooth not jagged if this mode is on Additional FC adjustment Sliders Q1 and Q2 located at the very bottom of the screen provide additional filter FC adjustment which are added to values set by FC adjustment sliders This extra adjustment makes speech sound more natural and be more comfortable for the listener s perception Q1 adjusts FC convexity within 100 800 Hz frequency band Q2 changes FC increase decrease for every 1000 Hz starting from 1000 Hz Both sliders work within 18 18 dB range with current value indicated to the left of it Chapter 9 Adaptive broadband filtering Adaptive broadband filter is based upon adaptive frequency algo rithm This algorithm is de
59. urrent time from phonogram start is displayed along the X axis while Y represents signal amplitude in samples within 20 33000 range To zoom Y axis in and out there is a scalebar to the right and and buttons Blue area of the scale represents currently displayed amplitude range You may set new upper and lower border by clicking on the bar with left or right mouse button respectively Current borders are indicated with black triangle markers at the right side of the bar and their numerical expressions X axis is not scalable i e this process always displays instant waveform of a signal Note that there is also Stereo waveform module similar to Waveform the only exception being that the former displays waveforms of signals in both channels Chapter 8 Equalizer This process displays signal spectrum and enables user to correct it via inverse filtering and filter contrasting Equalizer may work in automatic or semi automatic mode it is also possible to tune the filter manually to make fine spectrum adjustments This module may suppress any station ary components of a signal regardless of their frequency and location it also may be used to raise the amplitude in a chosen spectral band This filter works well for phonograms containing considerable station ary noises such as power line noise mechanical and engine noises and so on Number displayed in the window header is current FFT window size The larger it is the larger is number
60. used to distinguish strong and weak signals If you choose to weaken the strong signals they will be also brought down exactly to the threshold value while weak signals will be ampli fied to one tenth of this threshold Radio buttons in the middle set separate dynamic processing modes 79 80 CHAPTER 16 DYNAMIC PROCESSING lt 9 gt Dynamic Processing o m Strong signal C Weaken Remain Weak signal C Weaken C Remain ol Amplify Threshold Figure 16 1 Dynamic processing window for strong and weak signals Strong signals may remain at their ini tial level or be weakened which is usually done to bring loud speech down to threshold value or eliminate strong and long more than 20 sec impulses knocks You may also amplify weak signals to balance the level of speech for two speakers leave them unchanged remain or weaken them which may be useful for suppressing the noise in pauses between loud speech fragments Chapter 17 Stereo processing There are two different kinds of stereo processing available in Sound Cleaner two channel signal processing and adaptive stereo filtering In the first case the sound in each channel is processed independently while in the second case data acquired from one channel called refer ent are used for filtering the signal in the second one primary chan nel Independent two channel processing Before you start stereo processing you should load

Sound Cleaner - Brokke System

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