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CARDINAL MiniLab Manual

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1. A a a a a d ii Lao Lo gt hk gt gt gt gt L pal h 2 2 gt 2 gt hkb 2 gt 3 a a ii ns e Output Level dE e Compression e Limiting Threshold Th r shold A a a a D 50 r0 B0 50 40 30 20 10 O Input Level dB Figure 123 Example LCE Curve In each section the LCE modifies the amplitude of the signal using a variable gain digital amplifier The amplitude is a rectified and smoothed version of the signal waveform as measured by a real time digital envelope detector So in the figure above the Input Level actually refers to the smoothed level envelope rather than the sample by sample instantaneous input level The operation of the envelope detector is governed by the Attack Time Release Time and Lookahead controls 88 Effect Gain Stage Presets Custom a T Sp GAIN STAGE Gain Mode i E Maximum Gain lt 26dBl Release Time Ms O dB mm 15 30 60 96 Output ESE Resample to 16kHz Averager Mode Num Awverages j Channel Select O Show Input Show Output Track Peak Apply Figure 124 Gain Stage Main Window AGC Mode Effect Gain Stage Presets Custom w Bi CARDINAL GAIN STAGE Gain Mode f Current Gain TE GEST Compression Threshold ETE Limiting Threshold 410 dB O dB m 8 z 13 a Expansion Ratio a i2 Compression Ratio
2. Averager Mode Num Averages Peak Freq 387Hz 37 22050Hz Sample Rate 44100Hz Figure 7 1 CH Adaptive Filter Main Window 4 2 MAIN PLUG IN WINDOW Figure 7 shows the plug in as it would be displayed if used in Adobe Audition The buttons above and below the plug in window will appear differently depending upon your audio editing environment Please consult the documentation on your specific audio editing environment for details of how to use these features This window provides access to all the functionality of the plug in A coefficient display and spectrum analyzer are also provided as aides in determining the characteristics of the target audio Details of the plug in are described in the following sections 4 3 FILTER CONTROLS 4 3 1 Prediction Span Prediction Span 19 Figure 8 Prediciton Span Sets the number of samples in the prediction span delay line Prediction span is indicated both in samples and in milliseconds Shorter prediction spans allow maximum noise removal while longer prediction spans preserve voice naturalness and quality A prediction span of 2 or 3 samples is normally recommended 4 3 2 Filter Size pi Size 1024 taps Figure 9 Filter Size Sets the number of FIR filter taps in the adaptive filter Filter size is indicated both in taps filter order and in milliseconds The maximum filter size is 8192 taps Small filters are most effective with simple noises such
3. 14 3 5 Recall Button Figure 155 Recall Button The Recall button will restore the entire filter chain configuration from a Filter Chain Settings FCS file that was preserved using the Store button Using this feature will wipe out the current Filter Chain configuration Figure 156 Clear Button The Clear button will clear all the adaptive filters in the chain It does not remove any filters from the chain 14 3 6 Clear Button 102
4. 46 8 3 1 2 Transition Slope pal Slope r Figure 40 Transition Slope Specifies slope at which frequencies above the Cutoff Frequency are rolled off in dB per octave Sharpest roll off occurs when Transition Slope is set to maximum while gentlest roll off occurs when Transition Slope is set to minimum Sharp rolloffs may cause the voice to sound hollow but will allow more precise removal of high frequency noises Note that the indicated value changes depending upon the Cutoff Frequency and Sample Rate 8 3 1 3 Stopband Attenuation Stopband Attenuation Figure 41 Stopband Attenuation Specifies amount in dB by which frequencies above the Cutoff Frequency are ultimately attenuated Stopband attenuation is adjustable from OdB to 120dB in 1 dB steps 8 4 HIGHPASS FILTER The Highpass filter is used to decrease the energy level lower the volume of all signal frequencies below a specified Cutoff Frequency thus reducing low frequency noises such as tape or acoustic room rumble from the input audio The Highpass filter is sometimes called a rumble filter The Cutoff Frequency is usually set below the voice frequency range somewhere below 300 Hz so that the voice signal will not be disturbed While listening to the filter output audio the Cutoff Frequency initially set to O Hz can be incrementally increased until the quality of the voice just begins to be affected achieving maximum elimination of low frequency noise
5. 10 3 1 Filter Mode Filter Mode Figure 79 Filter Mode The Filter Mode combo box allows the user to select either the Multiple Notch mode or the Multiple Slot mode When switching from one mode to the other the previous mode controls are remembered and preserved 10 3 2 Multiple Notch Filter Controls 10 3 2 1 Notch Frequency Frequency F501 Hz Figure 80 Notch Frequency Specifies frequency in Hertz which is to be removed from the input audio Minimum Notch Frequency is 10 Hz while maximum Notch Frequency depends upon the Sample Rate Notch Frequency is adjustable in 1 Hz steps Also sets the Base Frequency for a Notch Group 10 3 2 2 Notch Width j Notch Width hz i Figure 81 Notch Width Width of the generated notch in Hertz 10 3 2 3 Notch Depth 71 i Notch Depth Figure 82 Notch Depth Depth of the generated notch in dB 10 3 2 4 Notch Spacing i Notch Spacing Figure 83 Notch Spacing Notch Spacing defines where the other notches in the group are to be placed if the Base Notch frequency is F and the spacing is set to S then notches will be placed at frequencies F F S F 2S F 3S etc 10 3 2 5 Width Factor Width Factor 0 000 Figure 84 Width Factor Width Factor defines how wide the group notches should be Frequency variations often occur as a percentage of the frequency so the variation width in Hz is much larger at high frequencies The Width Factor defines a percentage width up
6. the total resulting output audio after the last stage of the filter chain Presets Custom FILTER CHAIN Averager Mode Num Averages a J COLO Lo ko ko ko Koko CI EI EI EI EI EI EIEI y y er M z Channel Select LeftOnty v O Show Input Show Output i z u mun mun aun mun Mo i Eesi ES ES ES E ual Radi Radi Radi Radi Wai m m alj e mail Bad Track Peak ao po Zoom Figure 150 Filter Chain Main Window 14 2 MAIN PLUG IN WINDOW Figure 150 above shows the plug in as it would be displayed if used in Adobe Audition The buttons above and below the plug in window will appear differently depending upon your audio editing environment Please consult the documentation on your specific audio editing environment for details of how to use these features This window provides access to all the functionality of the plug in A spectrum analyzer is provided as an aid in determining the characteristics of the target audio Details of the plug in are described in the following sections 100 14 3 FILTER CHAIN CONTROLS 14 3 1 Filter Select Figure 151 Filter Select Controls The Filter Select controls are a set of 8 combo boxes that allow you to select from all the available Cardinal MiniLab plug ins that are installed on your system Some filters are only available if the audio presented matches their criteria e g Reference Canceller is only available for stereo audio files The
7. Also notch groups can be added to cancel many harmonically related frequencies at once The Multiple Notch filter is synthesized from a frequency domain representation of the desired notch profile An inverse FFT builds FIR coefficients based on the frequency domain representation For this reason the notches in this filter are square notches rather than V notches Square notches mean that frequencies very close to the specified center frequency will be cancelled along with the center frequency However the square notches also mean that the Multiple Notch filter is able to tolerate moderate variances in the specified frequency such as those caused by wow and flutter effects Filters that use V notches include the Notch filter the Comb filter and the Parametric Equalizer To properly utilize the Multiple Notch filter you will first need to identify the noise frequencies The easiest way to do this is to use the included spectrum analyzer Once the noise frequencies have been identified add a notch for each frequency Notches are defined by three values the notch frequency the notch width and the notch depth The notch frequency is simply the frequency at which the notch should be centered The notch width defines the desired width of the square notch in Hz and the notch depth defines the desired depth in dB Often tonal noises include not only the fundamental frequency but also harmonic multiples of th
8. Filter Size 1024 taps 23 2200 ms Peak Tap 0 0 06250 Averager Mode 96 0dB Peak Freq OHz 96 0dB 22050Hz Sample Rate 44100Hz Sy alid Activation period remaining 30 days Please activate to install persistent lice machine Activate Close Figure 2 Plug in waiting for Activation To Activate simply click the Activate button This will send the necessary information to the DAC servers to activate the software on your PC The ONLY information sent and stored at DAC is the Serial Key and a hardware identifier usually a serial number from your System hard drive No other personally identifiable information is sent If the workstation you installed the software on does not have a network connection the plug in will display the Serial Key and HWID Hardware Identifier to you You must then contact DAC with this information and we will generate a valid License Key for you that you can then enter as shown in the figure below 12 Effect 1CH Adaptive Filter x Presets Custom Er a eee 1CH ADAPTIVE FILTER Vertical Zoom r Prediction Span 0 2268 ms tex Peak Tap 0 0 06250 Averager Mode 96 0 dB OHz 96 0dB Peak Freq OHz 96 0dB 22050Hz Sample Rate 44100Hz Contact DAC with this information Serial Key df64 9245 27a6 b5e5 542b eda8s 01f8 fdfa 00001 HWID 3159 Enter License Key here Figure 3 Plug in waiting for manual License Key entry Once activation is completed either over
9. The amount of volume reduction below the Cutoff Frequency can further be controlled by adjusting the Stopband Attenuation setting maximum volume reduction is 120dB The slope at which the volume is reduced from normal at the Cutoff Frequency to the minimum volume specified by Stopband 47 Attenuation can also be controlled by adjusting the Transition Slope setting Effect X Pass Filter Presets Custom Foi Filter Type Highpass F E Cutoff Frequency i Transition Slope 258 94 dB Octave E Stopband Attenuation 2120 dB i Sample Rate 14700Hz E Resample audio to 16kHz 0 0 dB Averager Mode Num Averages al ay Channel Select Let Ony e a A a iy Pat dni A A Puce oa ee 7 A 4 lo Ma aE LET A te a T Pro tb a hal e nk 7 O Show Input 1 O Show Output Ol Track Peak 96 0 dB 240 2068 F350Hz Close Figure 42 HIghpass Filter Controls 3 4 1 Filter Controls 8 4 1 1 Cutoff Frequency Cutoff Frequency Figure 43 Cutoff Frequency Specifies frequency in Hertz below which all signals are attenuated Frequencies 48 above this cutoff are unaffected Minimum Cutoff Frequency is O Hz no frequencies attenuated while the maximum Cutoff Frequency depends upon the Sample Rate Cutoff Frequency can be adjusted in 1 Hz steps 8 4 1 2 Transition Slope Transition Slope 258 94 dB Octave Figure 44 Transition Slope Specifies slope at which frequencies below the Cutoff Frequency are attenuated
10. the ASIF is adapting in response to incoming audio When the button is grayed the ASIF response is frozen The Clear button allows the user to re initialize the ASIF response and restart adaptation Note After a Clear operation or after re enabling adaptation there will be an adaptation period while the filter adapts to the current input signal The length of this adaptation period depends on the Adapt Rate control setting The Adapt Rate control allows the user to select the rate of adaptation for the spectral average on which the ASIF response is based The spectral averager uses an exponential average of the form Hi 1 a Xi i1 1 a Hi The value shown in the display box corresponds to the averaging constant a in the exponential average The lower the adapt rate value the slower the filter will respond to changes in the input audio Note Fast response sounds like a good thing so it can be tempting to set the adapt rate to a high value However the goal of the ASIF is not to remove transient noises but rather to reshape the long term spectral envelope of the signal If the adapt rate is too fast the filter will respond too quickly to transient audio characteristics which will produce artifacts in the output audio and will prevent the filter from settling on a good average solution For this reason most applications will work best with adapt rates at the low end of the available range If you hear tonal artifacts that co
11. Bandstop y p Lower Cutoff Frequency 2500 Hz i Upper Cutoff Frequency 3000 Hz i Transition Slope Stopband Attenuation 2120 dB Sample Rate 14700Hz Cy Resample audio to 16kHz 0 0 dB Averager Mode Exponential Num Averages a po Channel Select Left Onty C Show Input Show Output Fen tthe toot La J ag g Br l T m I ud 2 Ps cat A A a ia lg F h I E an oF H a JI a ant A T 1 at Ty Al if 4 y aniti 19 i aft XT 11 vie oF r iy h a iF i 1 We ha wy a a hey ho es Aly ae a hae iy Ww U ie ta G ate l b Track Peak 96 0 dB y 240 2d8 7350Hz Te Figure 51 Bandstop Filter Controls 8 6 1 Filter Controls 8 6 1 1 Lower Cutoff Frequency Lower Cutoff Frequency 2500 Hz Figure 52 Lower Cutoff Frequency Specifies frequency in Hertz below which no signals are attenuated Frequencies between this cutoff and the Upper Cutoff Frequency are attenuated Minimum Lower Cutoff Frequency is O Hz while the maximum Lower Cutoff Frequency is 93 10 Hz below the Upper Cutoff Frequency Lower Cutoff Frequency can be adjusted in 1 Hz steps NOTE The Lower Cutoff Frequency can never be set higher than 10 Hz below the Upper Cutoff Frequency 8 6 1 2 Upper Cutoff Frequency Upper Cutoff Frequency 3000 Hz Figure 53 Upper Cutoff Frequency Specifies frequency in Hertz above which no signals are attenuated Frequencies between this cutoff and the Lower Cutoff Frequency are attenuated Minimum
12. E stage 2 E stage Stage 3 E stage E Stage 4 E stage oo J mom in Figure 140 Current Stage The buttons in the Current Stage box allow the user to select the current equalizer stage for adjustment When a stage is selected its settings populate the Center Frequency Width Factor Boost Cut and Stage Active controls 13 3 2 Center Frequency Center Frequency G25 Hz Figure 141 Center Frequency The frequency at which the current stage s boost cut region is centered 13 3 3 Width Factor Width Factor ONES Figure 142 Width Factor A factor controlling the width of the current stage s boost cut region 97 13 3 4 Boost Cut Boost Cut F600 08 Figure 143 Boost Cut The amount of boost or cut to be applied by the current stage 13 3 5 Stage Active d Stage Active Figure 144 Stage Active If the indicator is lit the current stage is being applied to audio If the indicator is dark the current stage is bypassed 13 3 6 Output Gain Gutput Gain O dB Figure 145 Output Gain Amount of gain or attenuation applied to the audio after all active parametric EQ stages have been applied 13 3 7 Reset All Reset All Figure 146 Reset All Returns all stages back to their default settings which has no effect on the 98 audio Center Frequency OHz Boost Cut OdB Stage Inactive 13 3 8 Show Inactive En Show Inactive Figure 147 Show Inactive Displays an orange
13. HEE LiL Show Input Show Output OlTrack Peak 96 0 dB Fa 55 1d 22050Hz dom Figure 77 Multi Band Main Window Multiple Notch Mode 69 Effect Multi Band Filter Presets Custom ly Air MULTI BAND Filter Mode a 0 Multiple Slot _ w Sample Rate 44100Hz 0 Resample to 16kHz Add Slot Frequency 401 Hz Slot Spacing 400 Hz Add Group Slot Width Width Factor 0 000 M i Slot Gain 11 dB i Gain Factor Stopband Depth Upper Limit 4080 Hz _ 60 0 dB Aeran ERE Exponential Hum Averages O Channel Select Left Onty C Show Input GO Show Output AA Tii Pa a Aare A i i pane y ai AO pi Y paa pepino ly y Track Peak 96 0 dB 22050Hz Figure 78 Multi Band Main Window Multiple Slot Mode 10 2 MAIN PLUG IN WINDOW Figure 77 and Figure 78 above show the plug in as it would be displayed if used in Adobe Audition in Multiple Notch mode and Multiple Slot mode respectively The buttons above and below the plug in window will appear differently depending upon your audio editing environment Please consult the documentation on your specific audio editing environment for details of how to use these features 70 This window provides access to all the functionality of the plug in A spectrum analyzer is provided as an aid in determining the characteristics of the target audio Details of the plug in are described in the following sections 10 3 FILTER CONTROLS
14. Since it is an inherently two channel filter it is only available for stereo audio files The Primary channel MUST be on the left channel while the Reference channel consequentially MUST be on the right channel Future updates of this plug in will allow these channels to be switched if desired The Reference Canceller Filter comes equipped with the following features Filter sizes up to 8192 taps Delay line up to 32768 samples Auto Normalizing adapt rate Conditional Adaptation Reference Gain Selectable Filter Output Coefficient Display with max peak indicator Dual trace spectral analysis with max peak indicator and averaging 23 Effect Reference Canceller Presets Custom vv Hr Y CARDINAL REFERENCE CANCELLER Delay TS 4 9560 m5 a osa po Filter Size wr Gens MIN ILA B Figure 13 Reference Canceller Filter Main Window 5 2 MAIN PLUG IN WINDOW Figure 13 shows the plug in as it would be displayed if used in Adobe Audition The buttons above and below the plug in window will appear differently depending upon your audio editing environment Please consult the documentation on your specific audio editing environment for details of how to use these features This window provides access to all the functionality of the plug in A coefficient display and spectrum analyzer are also provided as aides in determining the characteristics of the target audio Details of the plug in are descr
15. Upper Cutoff Frequency is 10 Hz above the Lower Cutoff Frequency while the maximum Upper Cutoff Frequency depends upon the Sample Rate Upper Cutoff Frequency can be adjusted in 1 Hz steps NOTE The Upper Cutoff Frequency can never be set lower than 10 Hz above the Lower Cutoff Frequency 8 6 1 3 Transition Slope Transition Slope 2147 86 dB Octave Figure 54 Transition Slope Specifies slope at which frequencies above the Lower Cutoff Frequency and below the Upper Cutoff Frequency are attenuated in dB per octave Sharpest attenuation occurs when Transition Slope is set to maximum while gentlest attenuation occurs when Transition Slope is set to minimum Note that the indicated value changes depending upon the Cutoff Frequency and Sample Rates Also note that the Lower and Upper Transition Slopes always have different values this is because the frequency width of an octave is proportional to Cutoff Frequency 8 6 1 4 Stopband Attenuation 54 Stopband Attenuation Figure 55 Stopband Attenuation Specifies amount in dB by which frequencies above the Lower Cutoff Frequency and below the Upper Cutoff Frequency are attenuated 8 7 NOTCH FILTER The Notch filter is used to remove or notch out a narrow band noise such as a tone or a whistle from the input audio with minimal effect to the remaining audio The Notch filter works best with stable noise sources which have constant frequency if the frequency of the noise
16. applied to the signal when the input level falls in the Expansion Region The Expansion Ratio is expressed as a ratio 1 N Jumps in the output signal are N times larger than their corresponding jumps in the input signal For example with an Expansion Ratio of 1 3 a 10 dB jump in input level becomes a 30dB jump in output level 12 3 3 4 Compression Ratio p Ratio Figure 132 Compression Ratio Specifies the amount of compression to be applied to the signal when the input level falls in the Compression Region The Compression Ratio is expressed as a ratio N 1 Jumps in the output signal are N times smaller than their 92 corresponding jumps in the input signal For example with a Compression Ratio of 3 1 a 30dB jump in input level becomes a 10dB jump in output level 12 3 3 5 Attack Time pu Time Figure 133 Attack Time Controls how quickly the LCE responds to increases in input signal level For a more peak sensitive processor use a short Attack Time For a more average sensitive processor use a longer Attack Time For most speech applications a fast Attack Time of 2 5 milliseconds is recommended 12 3 3 6 Release Time p Time 199 ms Figure 134 Release Time Controls how quickly the LCE responds to decreases in input signal level Short Release Times lt 100 milliseconds can create an annoying pumping artifact as the level detector is too responsive to intra syllabic pauses Long Release Times gt 500 m
17. as tones and music Larger filters should be used with complex noises such as severe reverberations and raspy power hums A nominal filter size of 512 to 1024 taps is a good overall general recommendation CAUTION Large filter sizes gt 2048 taps will require large computing resources to maintain real time audio processing You may begin to hear skips in the audio during preview if your computer cannot keep up with the processing requirements However during render the audio will not contain any skips but may take longer than real time to process the file 4 3 3 Adapt Rate Adapt Rate Auto Normalize a 230 Figure 10 Adapt Rate Used to set the rate at which the adaptive filter adapts to changing signal conditions An adapt rate of 1 provides very slow adaptation while an adapt rate of 5884 provides fastest adaptation A good approach is to start with an adapt rate of approximately 100 200 to establish convergence and then back off to a smaller value to maintain cancellation Larger adapt rates should be used with changing noises such as music whereas smaller adapt rates are acceptable for stable tones and reverberations Larger adapt rates sometimes affect voice quality as the filter may attack sustained vowel sounds When Auto Normalize is turned on the specified adapt rate is continuously scaled based upon the input signal level This scaling generally results in faster 20 filter convergence without greatly increasing t
18. er rs AAA a Attack Time EEE Release Time 199 mo 60 A p 36 E Input Output Look Ahead Time 5ml ENT O Resample to 16kHz kecal Figure 125 Gain Stage Main Window LCE Mode 12 2 MAIN PLUG IN WINDOW Figure 124 and Figure 125 above show the plug in as it would be displayed if used in Adobe Audition in AGC mode and LCE mode respectively The buttons above and below the plug in window will appear differently depending upon your audio editing environment Please consult the documentation on your specific audio editing environment for details of how to use these features This window provides access to all the functionality of the plug in A spectrum analyzer is provided as an aid in determining the characteristics of the target audio Details of the plug in are described in the following sections 90 12 3 GAIN CONTROLS 12 3 1 Gain Mode Gain Mode Case y Figure 126 Gain Mode The Gain Mode combo box allows the user to select either the AGC mode or the LCE mode When switching from one mode to the other the previous mode s controls are remembered and preserved 12 3 2 AGC Controls 12 3 2 1 Maximum Gain Maximum Gain Figure 127 Maximum Gain Maximum Gain specified how much gain the AGC can apply in its attempt to bring the output signal up to the desired level The greater the Maximum Gain the lower the signal that can be brought up to the threshold level The Maximum Gain ran
19. filters The Coefficient Display displays the impulse response filter coefficients of the filter Vertical scaling of the filters coefficients for display is accomplished by clicking on the Vertical Zoom combo box control Supported zoom factors range from 1x to 32x The Track Peak feature displays a green vertical line in the graph display at the coefficient with the largest absolute value When this feature is enabled the maximum peak value and coefficient number is displayed in green text on the bottom right of the graph To enable this indicator click the button beside the Track Peak text so that the indicator light turns green The Channel Select control allows you to choose which channel s filter coefficients you wish to see displayed in the coefficient display window You can choose Left Only or Right Only Right Only is only available for stereo audio files 16 3 3 SAMPLE RATE CONVERSION ELO Resamele audio to aera Figure 6 Sample Rate Conversion All Cardinal MiniLab plug ins contain a resampling feature that utilizes a true Whittaker Shannon Interpolation to resample the incoming audio at 16kHz This method eliminates any distortion components commonly found in cruder resampling techniques such as linear interpolation Resampling the audio down at 16kHz sample rate primarily does two things for forensic audio 1 Bandlimits the audio to the forensic voice spectrum 200 5000Hz 2 Allows for larger filter
20. flutter effects To properly utilize the Multiple Slot filter you will first need to identify the noise frequencies The easiest way to do this is to use the included spectrum analyzer Once the noise frequencies have been identified add a slot for each frequency Slots are defined by three values the slot frequency the slot width and the slot gain The slot frequency is simply the frequency at which the slot should be centered The slot width defines the desired width of the square slot in Hz and the slot gain defines the desired amplitude in dB Often tonal noises include not only the fundamental frequency but also harmonic multiples of that frequency Instead of requiring the addition of an individual slot for each harmonic the Cardinal Multiple Slot filter allows the addition of Slot Groups to cancel harmonically related tones in a single action A Slot Group is defined in relation to its Base Slot The Base Slot is defined with a frequency width and gain just like a single slot Frequency width and gain of all other slots in the group will be calculated based on these parameters 68 Effect Multi Band Filter Ea Presets Custom Em Y Filter Mode Sample Rate 44100Hz D Resample to 16kHz Add Notch Itiple Notch Frequency Notch Spacing m W Notch Width 53 Width Factor Remove All Upper Limit 49 F E ers Recall 60 0 dB Averager Mode C Exponential y a H Ae Channel Select
21. in are described in the following sections 62 9 3 FILTER CONTROLS 9 3 1 Filter Controls Filter Controls i Fi j 1 a al Sas ES ee Pe ST a AS Figure 66 Filter Controls The Filter Controls combo box allows the user to select either the Noise EQ mode or the Noise Reducer mode displayed as N Reducer When switching from one mode to the other the previous mode s controls are remembered and preserved 9 3 2 Clear Figure 67 Clear Button Used to clear the spectral subtraction solution currently in memory and restart the algorithm from scratch 63 9 3 3 Filter Output Gain Filter Qutput O dB 15 30 60 Figure 68 Filter Output Controls The Output Gain control allows user to apply between O and 30dB of makeup gain to the processed output signal to maximize the signal level prior to final equalization AGC and listening recording The associated bargraph shows the actual output signal level after the gain has been applied 9 3 4 Noise EQ Controls The following sections apply only to the Noise EQ mode 9 3 4 1 Slider Controls B4 59 33 6 4 9 22 24 49 51 66 76 82 20 95 100 100 100 100 100 5581 1 7K 2 8K 3 9K 5 0K 61K 7 2K 8 3K 3 4K 10 5K 11 8K 12 7K 13 8K 14 9K 16 0K 17 1K 18 2K 15 3K 20 4K 21 5K Figure 69 Noise EQ Slider Controls The slider controls are used to specify the amount of noise reduction that the spectral subtraction attempts to apply to the input si
22. in dB per octave Sharpest attenuation occurs when Transition Slope is set to maximum while gentlest attenuation occurs when Transition Slope is set to minimum Note that the indicated value changes depending upon the Cutoff Frequency and Sample Rates 8 4 1 3 Stopband Attenuation E Stopband Attenuation 1208 Figure 45 Stopband Attenuation Specifies amount in dB by which frequencies below the Cutoff Frequency are ultimately attenuated 8 5 BANDPASS FILTER The Bandpass filter is used to decrease the energy level lower the volume of all signal frequencies below a specified Lower Cutoff Frequency and above a specified Upper Cutoff Frequency thus combining the functions of a Lowpass and Highpass filter connected in series into a single filter The signal region between the Lower Cutoff Frequency and the Upper Cutoff Frequency is called the passband region The Bandpass filter is useful for simultaneously reducing both low frequency rumble and high frequency hiss The Lower Cutoff Frequency is usually set below the voice frequency range somewhere below 300 Hz so that the voice signal will not be disturbed While listening to the filter output audio the Lower Cutoff Frequency initially set to O Hz can be incrementally increased until the quality of the voice just begins to be affected achieving maximum elimination of low frequency noise 49 The Upper Cutoff Frequency is usually set above the voice frequency range somewhere abo
23. room can cause the signal characteristics to change over the course of a recording For this reason simply freezing the filter once convergence is reached may mean that noise cancellation will degrade over time Instead of freezing the filter use Conditional Adaptation First allow the filter to converge in Always mode and then select If Normal Output lt Threshold and adjust the Threshold by observing the bargraph levels during pauses in speech Click on the Clear button if you desire the filter to completely readapt based upon the new Conditional Adaptation settings 5 3 5 Reference Gain 26 Figure 18 Reference Gain Ref Gain is used to add gain to the reference channel audio if necessary To achieve good cancellation it is important that the reference audio be at least as loud as the noise it is intended to cancel from the primary audio 5 3 6 Filter Output Figure 19 Filter Output Used to optionally listen to the rejected audio that is being cancelled by the adaptive filter Normal should almost always be selected but the Rejected setting can be useful when configuring the filter allowing the user to hear exactly what is being removed by the filter 5 3 7 Freeze Filter Used to enable or disable filter adaptation When Freeze is off the filter adapts according to its settings When Freeze is on the filter never adapts regardless of the other settings 5 3 8 Clear Filter Used to reset the coefficients of
24. sizes with less computational requirements For example a 1024 point filter at 48kHz will require roughly 3x the CPU resources that the same 1024 point filter will require at 16kHz Note After processing the plug in resamples the audio back to the original sample rate to return it to the audio editor environment If you process a 44 1kHz WAV file you will still have a 44 1kHz WAV file in the end but any information above 8kHz will have been eliminated 17 4 1 CH ADAPTIVE FILTER 4 1 OVERVIEW OF THE 1 CH ADAPTIVE FILTER The 1 Channel Adaptive filter is used to automatically cancel predictable and convolutional noises from the input audio Predictable noises include tones hum buzz engine motor noise and to some degree music Convolutional noises include echoes reverberations and room acoustics The 1 CH Adaptive Filter is a forensic plug in that has been specifically designed to work with VST host audio editing systems The 1 CH Adaptive Filter comes equipped with the following features Filter sizes up to 8192 taps Prediction Span up to 32768 samples Auto Normalizing adapt rate Conditional Adaptation Selectable Filter Output Coefficient Display with max peak indicator Dual trace spectral analysis with max peak indicator and averaging 18 Effect 1CH Adaptive Filter Presets Custom vy Aor Di ICH ADAPTIVE FILTER 444 CAY ne Vertical Zoom r Prediction Span r Filter Size 1024 taps 23 2200 ms
25. source varies the 1 Channel Adaptive filter is recommended To properly utilize the Notch filter you will first need to identify the frequency of the noise this is best done using the Spectrum Analyzer window Initially set the Notch Depth to 120 dB and the Notch Width to the narrowest possible value Next set the Notch Frequency to the noise frequency Fine adjustment of the Notch Frequency may be necessary to place the notch precisely on top of the noise signal and achieve maximum reduction of the noise This is best done by adjusting the Notch Frequency up or down 1 Hz at a time while listening to the Notch filter output on the headphones Often the noise frequency will not remain absolutely constant but will vary slightly due to modulation recorder wow and flutter and acoustic beating Therefore you may need to increase the Notch Width from its minimum setting to keep the noise within the notch For maximum noise reduction set the Notch Depth to 120dB It is best to adjust the Notch Depth up from 120 dB until the tone is observed then increase the depth 5 dB 55 Effect X Pass Filter Presets Custom Ei oan eae peer X PASS FILTER Filter Type Noten Notch Frequency Width 200 5 Hz E Notch Depth 120 dB l Sample Rate 14700Hz Cy Resample audio to 16kHz 0 0 dB Averager Mode Num Averages E El Channel Select Left Ony C Show Input Show Output Track Peak 96 0 dB 240 2
26. the Reference Canceller Filter Clearing a filter is useful when the audio characteristics change dramatically so that the filter can readapt to a new clean solution Clearing is also useful in the case of a filter crash when the filter coefficients diverge to an unstable state usually in response to a large and abrupt change in the signal coupled with a fast adapt rate 27 6 SPECTRAL INVERSE FILTER 6 1 OVERVIEW OF THE SPECTRAL INVERSE FILTER The Spectral Inverse Filter SIF has two modes an adaptive mode and a manual mode Thus it is actually a combination of the Adaptive Spectral Inverse Filter ASIF and the traditional Spectral Inverse Filter SIF Both are equalization filters that readjust the spectrum to match an expected spectral shape It is especially useful when the target voice has been exposed to spectral coloration i e muffling hollowness or tinniness but it can also be used to remove bandlimited noises The ASIF is much like the SIF except it continually updates the spectral solution whereas the SIF only updates the solution when it is built The filter maintains an average of the signal s spectrum and uses this information to implement a high resolution digital filter for correcting long term spectral irregularities The goal of the filter is to reshape the overall spectral envelope of the audio not to respond to transient noises and characteristics Several user controls are available for refine
27. then slots will be placed at frequencies F F S F 28S F 3S etc 10 3 3 6 Width Factor i Width Factor Figure 99 Width Factor Width Factor defines how wide the group slots should be Frequency variations often occur as a percentage of the frequency so the variation width in Hz is much larger at high frequencies The Width Factor defines a percentage width up to a maximum of 1 9 and each notch will be at least the width defined by that percentage For instance if a slot group has width factor 0 015 and one of the slots in that group is at 1000 Hz then the width of the 1000 Hz slot will be at least 1000 x 0 015 15 Hz NOTE The frequency domain representation used to build the Multiple Slot filter has an inherent minimum slot width Especially at the lower frequency slots the width specified by the Width Factor will often fall below that minimum width in which case the minimum width is used For this reason the effect of the Width Factor control may only be visible at the higher frequency notches 76 10 3 3 7 Gain Factor i Gain Factor all Figure 100 Gain Factor Gain Factor defines how much gain is applied to the group slots Many harmonic tonal noises have a 1 f volume profile where the lower harmonics are strong and higher harmonics are progressively weaker The Gain Factor controls the gain taper of the slots so that the slot gain can parallel the harmonic strength profile The base slot always has the
28. to a maximum of 1 9 and each notch will be at least the width defined by that percentage For instance if a notch group has width factor 0 015 and one of the notches in that group is at 1000 Hz then the width of the 1000 Hz notch will be at least 1000 x 0 015 15 Hz NOTE The frequency domain representation used to build the Multiple Notch filter has an inherent minimum notch width Especially at the lower frequency notches the width specified by the Width Factor will often fall below that minimum width in which case the minimum width is used For this reason the effect of the Width Factor control may only be visible at the higher frequency notches 10 3 2 6 Depth Factor 72 i Depth Factor Figure 85 Depth Factor Depth Factor defines how deep the group notches should be Many harmonic tonal noises have a 1 f volume profile where the lower harmonics are strong and higher harmonics are progressively weaker The Depth Factor controls the depth taper of the notches so that the notch depth can parallel the harmonic strength profile The base notch always has the specified Notch Depth while subsequent notches taper to smaller depths as frequency increases The higher the Depth Factor the more gradual the taper A Depth Factor of 0 0 produces the most severe taper and means effectively that there are no harmonics at all A Depth Factor of 1 0 means that notches have uniform depth at the Base Notch depth setting 10 3 2 7 Upp
29. trace on the graph to indicate what the transfer curve looks like if all the stages are enabled 13 3 9 Store Button NES Figure 148 Store Button This button allows the user to store the current configuration to a user specified disk file that will not be lost when the computer is turned off 13 3 10 Recall Button Recall Figure 149 Recall Button This button allows the user to recall a previously stored configuration from any of the saved disk files previously generated using the Store button 14 FILTER CHAIN 14 1 OVERVIEW OF THE FILTER CHAIN The Filter Chain plug in provides the ability to put up to 8 MiniLab plug ins in series and render the cumulative effect in one pass of the audio Normally each plug in must be brought up separately in the editor and rendered on the audio independent from one another This process can be time consuming and inefficient if you wish to apply more than one plug in With the Filter Chain it is possible to create a chain of plug ins modify their settings and listen to their cumulative effect before rendering the audio just once Some audio editors provide their own Effects Rack or Filter Chain but most do not If you wish to chain together Cardinal MiniLab plug ins it is recommended that you use the Filter Chain plug in to do it as it is designed specifically to work with our plug ins The spectrum analyzer displays the unmodified input from the audio editor and 99
30. Cardinal MiniLab Plug in Suite User s Manual eN Salient Sciences Clarity Revealed Cardinal MiniLab Plug in Suite User s Manual February 2015 ETA Salient Sciences Clarity Revealed Digital Audio Corporation d b a Salient Sciences 4018 Patriot Drive Suite 300 Durham NC 27703 Phone 1 919 572 6767 Fax 1 919 572 6786 sales salientsciences com www salientsciences com Copyright 2015 by Salient Sciences All rights reserved TABLE OF CONTENTS Cardinal Mmi Lab Ito UCA ci 9 2 OM Ware StA NAON liliana 10 2 SO Ware Installa tio sesan nea had neue vacnanesh eaa 10 202 MAGCNSINe Amd AC MV AMON eeann EEE EAEN 10 gt General Plug im CONC DIS iii iia 14 Sl Anc A iaes 14 S L Spectrum Ana YZEF as isciscczernandadswobasianonscanadmentcatansdatonseauadwaawsnsiaanessacadaes 14 322 COCINICICNL diS Pla o eee eee 16 Seo Dampe Rate CONVE ION ouire is 17 Ae de A A A 18 4 1 Overview of the 1 CH Adaptive Filter oooooccccnconccnccnnccnccnncnncnnncnncnnos 18 A2 Mata Pusa Wii OW iii ii 19 a Fiter CONOS osa 19 dl Prediction SpA il 19 Fo 2 ed e e O ZE 0 ROO NO E E R 20 AS O A APDO RAE di 20 4 3 4 Conditional Adaptation seeen A EEE N E aoi 21 Ao Fer 1A DU rei AAA 21 A A Memos E ENEA 22 esr lear T e resapan a e Sate ciid teccloeecSae eal SectSisidiste as eetecacl State 22 O Reterence Canceler Tiradas 23 5 1 Overview of the Reference Canceller Filter ooooococconccncconccnccnncnnooo 23 3 2 Mara Pl
31. Output O Track Peak Apply Figure 64 Spectral Subtraction Main Window Noise Reducer Mode 61 Effect Spectral Subtraction Presets Custo m CARDINAL SPECTRAL SUBTRACTION Filter Controls g4 59 33 6 4 9 22 34 43 51 68 78 82 90 95 100 100 100 100 100 Fliter Output Noise EQ y E Zero All Normalize Maximize All All Up 1 All Down 1 561 17K 2 8K 3 9K 5 0K 61K 7 2K 8 3K 3 4K 10 5K 11 6K 12 7K 13 8K 14 5K 16 0K 17 1K 18 2K 19 3K 20 4K 21 5K Gain BELTE Averager Mode Num Averages pea a tut s ie M f hin Channel Select F ky fha y ta A e fi n Lef Sny 1 i 1 y SAP aa it ao ait He Input Wy ly i at i ai Al af he l A ARA chow Outp ut if l Peka ty aa aA AA Ol Track Peak 96 0 dB ye Zoom Figure 65 Spectral Subtraction Main Window Noise EQ Mode 9 2 MAIN PLUG IN WINDOW Figure 64 and Figure 65 above show the plug in as it would be displayed if used in Adobe Audition in Noise Reducer mode and Noise EQ mode respectively The buttons above and below the plug in window will appear differently depending upon your audio editing environment Please consult the documentation on your specific audio editing environment for details of how to use these features This window provides access to all the functionality of the plug in A spectrum analyzer is provided as an aid in determining the characteristics of the target audio Details of the plug
32. Pen ee eR me 48 8 9 Bandpass Hille amp ii 49 Sn Ll Piter Cont Sinai 50 5 0 Bandas top PUE ES A EE Na 92 SO ker COn TOS e o dr a ON E 53 NOC EE F aa da a 55 Sl Per Controla dd deba dea 56 A A A E O A DS 57 OSs CONTO S aera T A 58 O Spectral Subtraction ter ia 60 9 1 Overview of the Spectral Subtraction Filter ooooccconconcccnconccnccnncnnos 60 Be Nat Ea WARIO es Seats salsas aya ented icon E 62 IAS AAN Rn nee men en nee ee 63 ol Fiter CONTO Sr os dios 63 SESION E EINEN 63 SRA O e AAA A A ren RS 04 D4 NOISE EOQ CONTO Saint ios 64 9 3 5 Noise Reducer COntrols iuVi enotienie E RE 66 10 Multi Band Filter irritado 67 10 1 Overview of the Multi Band Filter oooooocccccconconncnncnnccnncnncnnncnncnnoos 67 LOW A Overview o Multiple Not ca 67 IOTZ Overview Ol MU plea SOl eeoa A meine 68 10 2 Matt Plus WII OW tesco dan tse deci ee tea exsada Ada 70 10 3 Falter CORTOS 71 FOS Erer Mode ii AE NA 71 10 92 Multiple Notch Filter Controls sesno E 11 103 3 Multiple Sot CONOS ai den LD Mt AN 80 11 1 Overview or the Graphic HQualiZer oseanen aaeersaniinnateneae ios 80 11 Overview Or Grape EO dida 80 P11 2 Overview or MRES Grapnic Ni 80 LL Mi PS WII dd ive 82 1122 TG AZ r CONO Si o seh ies 83 2 Guan Zer MOdE ris sist reseae eich gine nna es 83 112 2 Grap nic EO CONTOS nadia 83 11 20 Hiekes Graphic Controls aceiaeoaeiod newness 84 PACET CAS E ahaa pa ea erences E E AR ONO 87 12 Overview Or he Gain State reee e
33. RUM ANALYZER Averager Mode E ida Hum Awerages a Z Channel Select Left Ontby O Show Input Show Output Track Peak Zoom Figure 4 Spectrum Analyzer Most plug ins have an integrated spectrum analyzer to display frequency content of the audio Most will look similar to the figure above The Spectrum Analyzer displays the frequency content of the input and or output audio The frequency axis of the Spectrum Analyzer plot goes from left to right with the lowest frequency at the leftmost side and the highest frequency at the rightmost side The energy loudness axis of the Spectrum Analyzer plot goes from top to bottom with stronger louder frequencies 14 indicated by higher peaks on the plot For example a signal consisting of a single tone will appear as a single peak located at the tone frequency A white noise signal which contains equal amounts of every frequency in the signal bandwidth will appear as an approximately flat line across the entire frequency range The spectrum analyzer can display the Input audio the Output audio or both To enable the Input and or Output trace click the corresponding LED button so that the indicator light is on The Input audio is the signal before the filter is applied The Output audio is the signal after the filter is applied When both signals are shown each signal is indicated by a line using yellow for the Input and blue for the Output The Sp
34. Show Input Show Output 120 0 de Figure 20 Spectral Inverse Filter Main Window in Adaptive Mode 29 Effect Spectral Inverse Filter Presets Custom ly Bi RDINSS SPECTRAL INVERSE CNY Filter Operation Output Shape Filter Amount 400 E Flat a Equalize Voice Pink o Aiea Ae Nae Output Gain E Auto Gain 0 0 dB E Voice Adaptive Mode Lower Voice Limit E Run Analyzer i Manual Upper Voice Limit BO00 Hz i Sample Rate 14700 Hz va Resample audio to 16kHz 60 0 db e Aa A il eS EnA jF TPF wT h bi a TT pa Num Averages Channel Select Let Only y O Show Input Show Output 120 0 de Figure 21 Spectral Inverse Filter Main Window in Manual Mode 6 2 MAIN PLUG IN WINDOW Figure 20 and Figure 21 show the plug in as it would be displayed if used in Adobe Audition in Adaptive mode and Manual mode repsectively The buttons above and below the plug in window will appear differently depending upon your audio editing environment Please consult the documentation on your specific audio editing environment for details of how to use these features This window provides access to all the functionality of the plug in A spectrum 30 analyzer is provided as an aid in determining the characteristics of the target audio Details of the plug in are described in the following sections 6 3 FILTER CONTROLS 6 3 1 Filter Operation Filter Operation Equalize Voice At
35. ZER The Parametric Equalizer consists of a variable number of IIR filter stages connected in series which can be used for boosting or peaking and cutting or nulling portions of the input signals frequency spectrum Each stage is described by a center frequency a frequency width and a boost cut amount and can be configured independently A common application of the parametric equalizer is to construct a precision notch filter which will perform nulling of the input signal at the specified center frequencies In the Current Stage block the eight available stages can be selected one at a time to adjust their individual configurations Individual stages can be toggled between Active and Inactive An active stage is applied to the audio while an inactive stage is bypassed When a stage is made inactive its settings are preserved HINT It is often helpful to activate only one stage at a time when adjusting the stage settings Then once satisfactory settings have been found for each individual stage all stages can be activated for audio processing When multiple stages are in use their effects can overlap so that the overall signal level is reduced or boosted more than expected For this reason an output gain control is available as part of the Parametric Equalizer allowing the user to compensate for overall level changes that may result from Parametric Equalizer filtering Advanced users may note that many Parametric EQ filters prov
36. ai E 87 124 LOVE view Ol AGC ah rire esas oie giles ace ein ee eee meee as 87 FLA OO VET VIS WO LEE dd idas 87 22 Mata Phe ii Wit OW land 90 AECA A A Sen cee tiene my eae nt RIE ene E eee 91 Res EE ols eae eee tT ree E reese 91 2 32 AGO CORTOS se 91 Eo GCE CONTOS di diia 92 to Paramete Equal ia 95 13 1 Overview of the Parametric Equalizer aoier ea e a a 95 AA E Ee 1 Wat OW AAA EA 96 ES Elite CONOS cs 97 PO OnE CULrren Stato aiii E R 97 99 20 Center F rC CNC iii A AS 97 Foo WIG EI PACO Sa 97 EA A A A eee 98 Sos Stage ACIIV Gc ici apia 98 ESAS O OUPUCGA PC UP 0 U PMI arode aban ailacnes Bleed tend oes 98 Ese RES CA A aoe hous 98 URC Wis Rowe AO INACTIVO ii adi 99 Sd 9 SLOPE BUTTON is 99 TS Oy RECAM dr decias 99 TA FIle Cds satan 99 14 1 Overview of the Filter Chain ccc ccc eee eeeccccecccccccccceccccveseeecs 99 14 2 Mata Plus WII Wa acca eevee aiid E das 100 143 Filter Chain Controls os adoos 101 NA ye o AA A AS a Eas 101 Eko Filter Control Button Urol Das 101 A A nite reich ali iidacines EAE EOE E EREE 102 E RS StOre Btn sth ca a etre nels tas ane hc heh eee oka 102 ES reel Reca BUTO 5 di ais 102 IES OC Ear BUON odas 102 1 CARDINAL MINILAB INTRODUCTION Thank you for purchasing the Cardinal MiniLab Plug ins The entire suite of plug ins includes the following 1 CH Adaptive Filter Comb Filter Filter Chain Gain Stage Graphic Equalizer Multi Band Filter Parametric Equalizer Refe
37. ain slot spacing gain factor width factor and upper limit of the slot group 10 3 3 11 DTMF Button DTMF Figure 104 DTMF Button Inserts 8 pre defined frequencies that make up the Dual tone multi frequency DTMF The version of DTMF used for telephone tone dialing is known by the trademarked term Touch Tone and is standardized by ITU T Recommendation 10 3 3 12 Remove Button Remove Figure 105 Remove Button Removes the currently selected slot or slot group from the filter 10 3 3 13 Remove All Button Remove All Figure 106 Remove All Button Removes all slots and slot groups from the filter 10 3 3 14 Store Button Figure 107 Store Button 78 Saves the filter s current configuration to a disk file 10 3 3 15 Recall Button Recall Figure 108 Recall Button Loads a previously saved filter configuration from a disk file 79 11 GRAPHIC EQ 11 1 OVERVIEW OF THE GRAPHIC EQUALIZER The Graphic EQ provides two different equalizers in a single plug in the Graphic EQ and Hi Res Graphic Only one equalizer can be operational at a time 11 1 1 Overview of Graphic EQ The 20 band Graphic Equalizer is an easy to use linear phase FIR digital filter that is used to reshape the spectrum of the final output signal Reshaping is accomplished with twenty vertical scroll bars also called slider controls which adjust the attenuation of each frequency band These controls are very similar to the s
38. at frequency Instead of requiring the addition of an individual notch for each harmonic the Cardinal Multiple Notch filter allows the addition of Notch Groups to cancel harmonically related tones in a single action A Notch Group is defined in relation to its Base Notch The Base Notch is defined with a frequency width and depth just like a single notch Frequency width and depth of all other notches in the group will be calculated 67 based on these parameters 10 1 2 Overview of Multiple Slot The Multiple Slot filter is used to isolate or slot single frequency noises such as tones or whistles in the input audio attenuating all other audio This is the exact opposite of the Multiple Notch filter function Single slots can be added one at a time and configured individually Also slot groups can be added to isolate many harmonically related frequencies at once The Multiple Slot filter is synthesized from a frequency domain representation of the desired slot profile An inverse FFT builds FIR coefficients based on the frequency domain representation For this reason the slots in this filter are square slots rather than V slots Square slots mean that frequencies very close to the specified center frequency will be cancelled along with the center frequency However the square slots also mean that the Multiple Slot filter is able to tolerate moderate variances in the specified frequency such as those caused by wow and
39. be the range over which speech frequencies are found Setting a Lower Limit above 300 Hz or an Upper Limit below 3000 Hz is not recommended in equalize voice mode as intelligibility may suffer When in Equalize Voice mode all frequencies outside the SIF region are assumed to be non speech and are therefore attenuated In Attack Noise mode the SIF region is typically chosen to bracket the bandlimited noise as closely as possible Frequencies outside the ASIF region will be passed through i e there will be little or no effect outside the SIF region except for a narrow transition band between the SIF region and the passbands Note Changing the Voice Limits does not require an adaptation period to arrive at a good solution Because a full average spectrum is maintained regardless of the Voice Limit settings the new Voice Limits will take effect instantaneously in both the output audio and the display traces However since the auto gain adapts based on the actual applied filter with voice limits taken into account there may be some adaptation time required to reach a stable auto gain value after the limits are changed 34 6 3 6 Adaptive Mode Adaptive Mode h Adapt Manual Adapt Rate MEET Figure 27 Adaptive Mode In Adaptive mode the above figure is displayed and gives the user access to the Adapt Clear and Adapt Rate controls This puts the plug in in the ASIF mode When the Adapt button is lit green
40. button next to each filter select control enables the filter The audio always runs through each filter however if the filter is disabled its audio output is discarded This allows adaptive filters to keep their solution in sync with the audio playback but prevents the output of the filter to continue down the chain 14 3 2 Filter Control Button Group EE Figure 152 Filter Control Group The Filter Control Group of buttons allows the user to move delete and edit the corresponding plug in The Move button will shift the plug in up and down in the chain without altering its settings 101 The Delete button will remove the corresponding plug in from the chain and shift any filters below it up to fill the gap The Edit button brings up the graphical interface for the corresponding plug in You can have multiple plug in windows open at one time 14 3 3 Bypass All gt Bypass All Figure 153 Bypass All The Bypass All control will disable all the filters in the chain This is useful to easily hear before and after the filter chain processing to see if you are achieving the desired result 14 3 4 Store Button Figure 154 Store Button The Store button will save the entire filter chain configuration including all filter parameters into a Filter Chain Settings FCS file for recall later These files can also be used by the Error Reference source not found to apply the settings to a large number of audio files in batch mode
41. can be operational at a time 12 1 1 Overview of AGC The Automatic Gain Control automatically attempts to boost low level output signals to a peak reference level 18dB bargraph level by gradually increasing output signal gain over a specified Release Time interval until either the proper level or Maximum Gain has been reached This compensates for near party far party conversations and for losses in signal level which may have occurred during the enhancement process If the output signal levels are at or above the 12 dB reference level the AGC will have no effect 12 1 2 Overview of LCE The Limiter Compressor Expander LCE is a three section signal level processor allowing manipulation of the overall dynamic range of a signal The LCE is typically used to correct for near party far party or quiet talker scenarios The three sections correspond to three types of level processing available limiting compression and expansion Limiting is applied to the loudest levels in a signal Compression is the middle region and expansion is applied to the quietest levels e In the Limiting region the output signal level is damped to the Limiting Threshold level When the input signal level is in the Limiting region attenuation is applied to keep the output level from exceeding the specified Limit Threshold e In the Compression region levels are adjusted so that output signal level changes are smaller than their corresponding inpu
42. cy 8 5 1 4 Stopband Attenuation l Stopband Attenuation NETO Figure 50 Stopband Attenuation Specifies amount in dB by which frequencies below the Lower Cutoff Frequency and above the Upper Cutoff Frequency are ultimately attenuated 8 6 BANDSTOP FILTER The Bandstop filter is used to decrease the energy level lower the volume of all signal frequencies above a specified Lower Cutoff Frequency and below a specified Upper Cutoff Frequency The signal region between the Lower Cutoff Frequency and the Upper Cutoff Frequency is called the stopband region The Bandstop filter is useful for removing in band noise from the input signal The Lower Cutoff Frequency is usually set below the frequency range of the noise while the Upper Cutoff Frequency is set above the frequency range of the noise While listening to the filter output audio the Lower and Upper Cutoff Frequencies can be incrementally adjusted to achieve maximum elimination of noise while minimizing loss of voice The amount of volume reduction in the stopband region can further be controlled by adjusting the Stopband Attenuation setting maximum volume reduction is 120dB The slope at which the volume is reduced from normal at each Cutoff Frequency to the minimum volume specified by Stopband Attenuation can also be controlled by adjusting the Transition Slope setting 92 Effect X Pass Filter Presets Custom Ei Sa inate a X PASS FILTER Filter Type C
43. d 248 048 048 048 048 048 048 048 0d8 2248 4548 74dB S4dB 10048100481004810048100d8 247 742 1 2K 1 7K 2 2K 2 7K 3 2K 3 7K 4 2K 4 7K 54K 6 7K 7 9K 9 4K 10 9K 12 4K 13 9K 15 4K 16 8K 18 3K Figure 112 Slider Controls The twenty vertical scroll bar slider controls are used to set the frequency response of the equalizer Each slider can set the gain of its frequency band to any value between OdB and 100 dB in 1dB steps 11 2 2 2 Zero All Zero All Figure 113 Zero All Button This button instantly moves the slider controls for all bands to OdB defeating the entire equalizer This is a useful feature when it is desired to reset all sliders from scratch 83 11 2 2 3 Normalize Button Normalize Figure 114 Normalize Button This button instantly shifts all slider controls up together until the top slider is at OdB After normalization the relative positioning of the sliders remains the same This allows the digital equalizer to implement the desired equalization curve with minimum signal loss 11 2 2 4 Maximize All Maximize All Figure 115 Maximize All Button This button instantly moves the slider controls for all bands to 100dB maximizing the attenuation for all bands This is a useful feature when it is desired to quickly adjust the sliders such that only a few bands are passed with all others rejected 11 2 2 5 AllUp1 All Up 1 Figure 116 All Up 1 Button This button shifts all sliders up by 1dB
44. dB Figure 56 Notch Filter Controls 8 7 1 Filter Controls 8 7 1 1 Notch Frequency Notch Frequency Figure 57 Notch Frequency Specifies frequency in Hertz which is to be removed from the input audio Minimum Notch Frequency is 10 Hz while maximum Notch Frequency depends upon the Sample Rate Notch Frequency is adjustable in 1 Hz steps 96 8 7 1 2 Notch Width Notch Width Pe 200 5 HZ Figure 58 Notch Width Width of the generated notch in Hertz NOTE Notch Width varies with the Sample Rate 8 7 1 3 Notch Depth Depth Figure 59 Notch Depth Depth of the notch that is generated 8 8 SLOT FILTER The Slot filter is used to isolate or slot a single frequency signal such as a tone or a whistle in the input audio attenuating all other audio This is the exact opposite of the Notch filter function NOTE The Slot filter has very little use in speech enhancement applications the main value is in isolating other types of signals that are non speech in nature To properly utilize the Slot filter you will first need to identify the frequency of the signal to be isolated this is best done using the Spectrum Analyzer window Once the frequency of the signal has been identified initially set Stopband Attenuation to 120 dB and the Slot Width to the narrowest possible value Next set the Slot Frequency to the signal frequency Fine adjustment of the Slot Frequency may be necessary to place th
45. dardized by ITU T Recommendation 10 3 2 11 Remove Button Remove Figure 90 Remove Button Removes the currently selected notch or notch group from the filter 10 3 2 12 Remove All Button Remove All Figure 91 Remove All Button Removes all notches and notch groups from the filter 10 3 2 13 Store Button 74 Figure 92 Store Button Saves the filter s current configuration to a disk file 10 3 2 14 Recall Button Recall Figure 93 Recall Button Loads a previously saved filter configuration from a disk file 10 3 3 Multiple Slot Controls 10 3 3 1 Slot Frequency i Frequency Figure 94 Slot Frequency Specifies frequency in Hertz which is to be enhanced in the input audio Minimum Slot Frequency is 10 Hz while maximum Slot Frequency depends upon the Sample Rate Slot Frequency is adjustable in 1 Hz steps 10 3 3 2 Slot Width i Slot Width 125 H7 i Figure 95 Slot Width Width of the generated slot in Hz 10 3 3 3 Slot Gain i Slot Gain mann Figure 96 Slot Gain The gain of the base slot in dB 15 10 3 3 4 Stopband Depth j Stopband Depth Figure 97 Stopband Depth Specifies amount in dB by which frequencies other than the Slot Frequency are attenuated 10 3 3 5 Slot Spacing j Slot Spacing Figure 98 Slot Spacing Slot Spacing defines where the other slots in the group are to be placed if the Base Slot frequency is F and the spacing is set to S
46. e Win OW a ds 24 Ow Fiter OM EPO lS PANA ena o O AR A Reseni 24 So LDE AV e O E E RN E 24 A E 25 OO Adap RIC ea EE N E E O sans 25 5 3 4 Conditional Adaptation oocccccccnconccnncnnconccnncnncnnnonnnnnconnnnnnnncnnnonnnnnos 26 0 OD IRCIETE MCE A nents E E eeacctiet desde 26 IO NSE OMPI AA 21 Die Freeze PINOT sin cikd taste On uu O nates 27 AS e mA T EEa 27 OrSpectral verse Utero iS 28 6 1 Overview of the Spectral Inverse Filter oooococcocccnnconccnccnncnnccnncnncnnoos 28 62 Marino Plas W GOW rastas 30 Dro HIS OO Sutra tea and S 31 Oo Biter Operation prada ino T EO OOT REO 31 ESA OTP ONAE ra a a a E ous one anna E eee 32 Boo Butter Ammon od 32 673 4 Outp t Gairand Acto analisis 33 6 3 5 Lower and Upper Voice LIME speaigvnsiensopmmmasurxenonnoimiaagsignderseommasasevt 34 653 06 Adapuve MOE AS E aii 35 o Mana MOE E A EET NA A asa 36 Oro DISPLAY estrada tata a des oa dad os 36 A A E ROS 38 7 1 Overview of the Comb Filter 2 0 cece cece cence ce ceccccccccccceencnecceeececeeeees 38 Toh M in Plug in Wal GOw vis a AE ee tay A A A a 40 TES ia sock tertile EN 41 EAN noo coa AAN Zn E e att nern tener werner 41 FD INOUC He AAAs scr a adan 41 IR PP E PE EVNENE 41 TDR NO CA ATEOS da 42 BD PAO LA sit 42 A A A A A are MRE Ren MT entre Sone 43 Sl Overview of the X pass Mitra td dd dados 43 8 2 Mair Plugin Wind Winds 44 So LOW ASS Ue ica asedio tocas oa 45 8 3 L Aero e es T 46 Sib Highpass Pitra di 47 O ao AAA EAE en
47. e slot right on top of the signal This is best done by adjusting the Slot Frequency up or down 1 Hz at a time while listening to the Slot filter output on the headphones Usually the signal frequency will not remain constant but will vary slightly due to modulation recorder wow and flutter and acoustic beating Therefore you 57 may need to increase the Slot Width from its minimum setting to avoid having the signal move in and out of the slot To optimize background noise reduction for your application set the Stopband Attenuation to 120dB If however you wish to leave a small amount of the background noise mixed in with the isolated signal adjust the Stopband Attenuation to the desired value Effect X Pass Filter Presets Custom Er aah wees X PASS FILTER Filter Type pa Frequency 856 HZ j Width EE E Stopband Attenuation 08 Sample Rate 14700Hz aa Resample audio to 16kHz Averager Mode Mum Averages Channel Select C ketony Te 0 Show Input Gj Show Output Track Peak Close Figure 60 Slot Filter Controls 3 8 1 Filter Controls 8 8 1 1 Slot Frequency 98 Frequency Figure 61 Slot Frequency Specifies frequency in Hertz which is to be enhanced in the input audio Minimum Slot Frequency is 10 Hz while maximum Slot Frequency depends upon the Sample Rate Slot Frequency is adjustable in 1 Hz steps 8 8 1 2 Slot Width pa Width Figure 62 Slot Width Width o
48. ectrum Analyzer provides a cursor marker to help in identifying specific frequency values At any time when the Spectrum Analyzer is activated clicking in the graph area displays a red vertical marker at the frequency location clicked The frequency in Hz of the location clicked is displayed in red text on the bottom left of the graph The input level in dB is given in yellow text and the output level in dB is given in blue text The Track Peak feature displays a green vertical line in the graph display at the strongest loudest frequency When this feature is enabled the maximum peak value in dB and frequency in Hz is displayed in green text on the bottom right of the graph To enable this indicator click the button beside the Track Peak text so that the indicator light turns green Use the Averager Mode control to select what type of averaging is done to the signals displayed No Average applies no averaging and the traces will move very rapidly Exponential applies averaging to the trace values based on the Num Averages value Peak Hold will hold the maximum value for each data point of the trace Trace data will never leak down in this mode This is useful for determining peak energy for a given frequency Use the Num Averages control to increase or decrease the number of averages applied to the frequency spectrum A small Num Averages value allows a more accurate snapshot of the spectrum at a given time but the trace values w
49. er Limit i Upper Limit Figure 86 Upper Limit Upper Limit defines how many notches there are in the group If a harmonic tonal noise only extends up to a certain frequency it may be undesirable to notch out all multiples of the base frequency when only a few are needed In this case set the Upper Limit just above the highest frequency where a notch is desired notches will be added up to that limit and no notches will be added above the limit 10 3 2 8 Add Notch Add Notch Figure 87 Add Notch Button Adds a new single notch at the frequency indicated in the Notch Frequency box and by a marker on the visualization axis The notch is added with default settings and the user is presented with controls to adjust the frequency width and depth of the notch 10 3 2 9 Add Group 73 Add Group Figure 88 Add Group Button Adds a new notch group with its base notch at the frequency indicated in the Notch Frequency box and by a marker on the visualization axis The notch group is added with default settings and the user is presented with controls to adjust the frequency width depth notch spacing depth factor width factor and upper limit of the notch group 10 3 2 10 DTMF Button DTMF Figure 89 DTMF Button Inserts 8 pre defined frequencies that make up the Dual tone multi frequency DTMF The version of DTMF used for telephone tone dialing is known by the trademarked term Touch Tone and is stan
50. ets Custom Er COMB FILTER i Comb Frequency E Notch Limit Harmonics Notch Depth MORET E Auto Tracking Oy All Even C Odd Sample Rate 44100Hz E Resample audio to 16kHz Awerager Mode Num Averages Zj Channel Select Show Input Show Output J Track Peak Figure 30 Comb Filter Main Window Manual mode 39 Effect Comb Presets Custom Elm Y COMB FILTER E Notch Limit OUT Harmonics Notch Depth MET Oy Auto Tracking ey All i Even cy Odd Sample Rate 44100Hz Resample audio to 16kHz rior A 0 0 dB Averager Mode Num Averages a Z Channel Select C ketony Je O Show Input Show Output J Track Peak Figure 31 Comb Filter Main Window Auto Tracking 7 1 MAIN PLUG IN WINDOW Figure 30 and Figure 31 above show the plug in as it would be displayed if used in Adobe Audition in Manual mode and Auto Tracking mode respectively The buttons above and below the plug in window will appear differently depending upon your audio editing environment Please consult the documentation on your specific audio editing environment for details of how to use these features This window provides access to all the functionality of the plug in A spectrum analyzer is provided as an aid in determining the characteristics of the target audio Details of the plug in are described in the following sections 40 7 2 FILTER CONTROLS 7 2 1 Comb Frequency Co
51. f the generated slot in Hertz NOTE Slot Width varies with the Sample Rate 8 8 1 3 Stopband Attenuation f Stopband Attenuation NET Figure 63 Stopband Attenuation Specifies amount in dB by which frequencies other than the Slot Frequency are attenuated 99 9 SPECTRAL SUBTRACTION FILTER 9 1 OVERVIEW OF THE SPECTRAL SUBTRACTION FILTER The Spectral Subtraction filter is a frequency domain filter that implements automatic noise reduction over the entire frequency spectrum It operates by continually measuring the spectrum of the input signal and attempting to identify which portions of the signal are voice and which portions are non voice or noise All portions determined to be noise are used to continually update a noise estimate calculation this is used to calculate the equalization curve that needs to be applied to the input signal to reduce each band s energy by the amount of noise energy calculated to be in that band The net result is an output signal that has all non voice signals reduced in level as much as possible thereby polishing the enhanced voice signal as much as possible prior to final equalization and AGC The Spectral Subtraction filter has two modes the Noise Reducer mode and the Noise EQ mode Operation in the Noise EQ mode is governed by 20 control sliders each representing a frequency band Adjusting the control sliders allows the user to precisely control the amount of noise reduction being a
52. from their current position no slider however will be allowed to go above OdB This button allows the user to shift the entire equalizer curve up so that there will be room to move one or more sliders down relative to the others 11 2 2 6 All Down 1 All Down 1 Figure 117 All Down 1 Button This button shifts all sliders down by 1dB from their current position no slider however will be allowed to go below 100dB This button allows the user to shift the entire equalizer curve down so that there will be room to move one or more sliders up relative to the others 11 2 3 Hi Res Graphic Controls 11 2 3 1 Hi Res Graphic Mini Tutorial The smoothing curve is graphed by the user using control points These control 84 points are seen in Figure 110 as large circles on the graph Control points represent a point on the curve where the slope of the line changes Users can manipulate these control points in one of three ways e Add a control point e Delete a control point e Move a control point To add a control point simply click on the graph where you want it to be The control point will immediately appear and you will hear the audio change immediately To delete a control point click on an existing control point it will turn red and then click the Delete button except the first and the last points they cannot be deleted This will remove the control point and the curve will snap back between the control points on either s
53. g in will also be copied to the default Steinberg VST plug in folder normally C Program Files Steinberg VstPlugins If this folder is not found it will be created Most audio editors will look in this directory by default for new plug ins 2 2 LICENSING AND ACTIVATION When you first launch your plug in it will be running in Demo mode This is normal operation until you enter the Serial Key you received from Digital Audio Corporation when you purchased your plug in NOTE In Demo mode the plug in will insert 1 second of silence for every 10 seconds of audio processed There is nothing wrong with your plug in you simply need to enter your Serial Key to unlock the plug in from Demo mode The plug in itself provides a field to enter the Serial Key as shown in the following figure 10 Effect 1CH Adaptive Filter aA El ON ICH ADAPTIVE FILTER Span TESTS al eaten P1024 taps pen A ILA B Enter Serial Key here Figure 1 Plug in in demo mode Simply enter your Serial Key in the black text field provided in the lower right of the screen and press Enter The Serial Key MUST be entered exactly as provided by DAC If the Serial Key is valid it will be registered with your system and you will now have 30 days to activate the plug in with DAC 11 Effect 1CH Adaptive Filter E Presets Custom Er A ICH ADAPTIVE FILTER M CAY ib Vertical Zoom r Prediction Span tex r
54. ge is 0 100dB For most near party far party applications around 10dB is recommended Settings greater than 10dB may elevate background noise to an objectionable level during pauses in speech A soft AGC using of 5dB is often useful even when large voice level differences are not present 12 3 2 2 Release Time Release Time ms Figure 128 Release Time Release Time controls how quickly the LCE will respond to decreases in input signal level The shorter the Release Time the more quickly the AGC will react For most voice applications a release time of about 200 milliseconds in recommended Release Time settings less than 200 milliseconds may result in annoying pumping sounds as the AGC changes gain during rapid fire 91 conversations 12 3 3 LCE Controls 12 3 3 1 Compression Threshold Threshold i Figure 129 Compression Threshold The level above which compression is applied to the signal The specified compression ratio is applied to the input signal whenever the input level is between the Compression Threshold and the Limit Threshold 12 3 3 2 Limiting Threshold j Limiting Threshold 1008 Figure 130 Limiting Threshold The level above which the signal is damped For instance if the Limit Threshold is 20dB all signal levels above 20dB will be attenuated to 20dB 12 3 3 3 Expansion Ratio E Expansion Ratio NO E Figure 131 Expansion Ratio Specifies the amount of expansion to be
55. gnal within each of 20 separate groups of frequency bands Within each band adjustment range is O no attenuation to 100 maximal attenuation in 1 increments 64 9 3 4 2 Zero All Zero All Figure 70 Zero All Button This button instantly moves the slider controls for all bands to 0 defeating the entire equalizer This is a useful feature when it is desired to reset all sliders from scratch 9 3 4 3 Normalize Normalize Figure 71 Normalize Button This button instantly shifts all slider controls up together until the top slider is at 0 After normalization the relative positioning of the sliders remains the same This allows the filter to implement the desired equalization curve with minimum signal loss 9 3 4 4 Maximize All Maximize All Figure 72 Maximize All Button This button instantly moves the slider controls for all bands to 100 maximizing the attenuation for all bands This is a useful feature when it is desired to quickly adjust the sliders such that only a few bands are passed with all others rejected 9 3 4 5 All Up 1 All Up 1 Figure 73 All Up 1 Button This button shifts all sliders up 1 while maintaining the desired curve This is useful when the desired shape is found but you want the filter to be less aggressive as a whole No slider will be allowed to go less than 0 65 9 3 4 6 All Down 1 All Down 1 Figure 74 All Down 1 Button This button shifts all slider
56. he risk of a filter crash It ts recommended that Auto Normalize be enabled for most speech signal processing 4 3 4 Conditional Adaptation Figure 11 Conditional Adaptation For advanced users only Novice users should keep Conditional Adaptation set to Always The threshold setting has no effect when Always is selected Conditional Adaptation allows the adaptive filter to automatically Adapt Freeze based upon signal bargraph levels This can be very useful in situations where there are pauses or breaks in the speech being processed Hint Conditional adaptation is useful for maintaining adaptation once the filter has converged Recording environment factors such as air temperature and motion in the room can cause the signal characteristics to change over the course of a recording For this reason simply freezing the filter once convergence is reached may mean that noise cancellation will degrade over time Instead of freezing the filter use Conditional Adaptation First allow the filter to converge in Always mode and then select If Normal Output lt Threshold and adjust the threshold by observing the bargraph levels during pauses in speech Click on the Clear button if you desire the filter to completely readapt based upon the new Conditional Adaptation settings 4 3 5 Filter Output Figure 12 Filter Output Used to optionally listen to the rejected audio that is being cancelled by the adaptive filter Normal should a
57. ibed in the following sections 5 3 FILTER CONTROLS 5 3 1 Delay 24 Figure 14 Delay Sets the number of audio samples by which the selected channel should be delayed Adjusting the Delay allows the alignment of the Primary and Reference channels to be adjusted Minimum Delay is 1 sample but can be set to as high as 32768 samples Specifies whether the delay line is to go into either the Primary channel Delay Pri or the Reference channel Delay Ref For most applications a slight delay typically 5 msec is placed in the Primary channel For applications with long distances between the microphone and radio TV a delay in the Reference channel may be required Extreme caution should be exercised when using reference channel delay allowing the reference to lag the target noise in the primary signal will result in poor cancellation 5 3 2 Filter Size pas Size 1024 taps ae Figure 15 Filter Size Sets the number of filter taps in the adaptive filter Filter size is indicated both in taps filter order and in milliseconds The maximum filter size is 8192 taps Normally larger filters sizes are used in the Reference Canceller adaptive filter CAUTION Large filter sizes gt 2048 taps will require large computing resources to maintain real time audio processing You may begin to hear skips in the audio during preview if your computer cannot keep up with the processing requirements However during render the audio will not co
58. ide To move a control point click on an existing control point it will turn red and drag it with the mouse Control points can only be moved vertically which adjusts the gain at that point Control points cannot be moved horizontally in an attempt to change the frequency at which the control point exists 11 2 3 2 Store Button Figure 118 Store Button This button allows the user to store the current configuration to a user specified disk file that will not be lost when the computer is turned off 11 2 3 3 Recall Button Recall Figure 119 Recall Button This button allows the user to recall a previously stored configuration from any of the saved disk files previously generated using the Store button 11 2 3 4 Normalize Button Normalize Figure 120 Normalize Button The Normalize button allows the user to shift the entire filter curve up until the highest point is at OdB 85 11 2 3 5 Clear All Clear All Figure 121 Clear All Button The Clear All button will clear all graph control points and reset the filter to a passthrough 11 2 3 6 Delete Button Delete Figure 122 Delete Button The Delete button will delete the current control point highlighted in red in the graph 86 12 GAIN STAGE 12 1 OVERVIEW OF THE GAIN STAGE The Gain Stage provides two different gain stages in a single plug in the Automatic Gain Control AGC and Limiter Compressor Expander LCE Only one gain stage
59. ide an input attenuation control so that fixed point saturation can be avoided Since Cardinal uses a floating point implementation saturation is not a concern so only output level adjustment is provided 95 Effect Parametric EQ Presets Custom Elm Y Current Stage Cy Stage 1 E Stage Stage 2 i Stage Stage 3 a Stage E Stage 4 E stage Boost Cut 60 0 dB Show Inactive HE E Stage Active Width Factor Channel Select Left Only C Show Input Show Output A 1 1 A k i et a T ET Le i PeF a i A i pe Y Fi N A Ta 1 1 r 1 Aha T y a i Lal y 4 fey Y AA Y I it wl E dy po hal a cae taal N 1 F F aiak Met ie al i pA ley Af o af rAr pifia C Track Peak 80 0 dB 240 2d8 22050Hz ee Figure 139 Parametric Equalizer Main Window 13 2 MAIN PLUG IN WINDOW Figure 139 above shows the plug in as it would be displayed if used in Adobe Audition The buttons above and below the plug in window will appear differently depending upon your audio editing environment Please consult the documentation on your specific audio editing environment for details of how to use these features This window provides access to all the functionality of the plug in A spectrum 96 analyzer is provided as an aid in determining the characteristics of the target audio Details of the plug in are described in the following sections 13 3 FILTER CONTROLS 13 3 1 Current Stage Current Stage E Stage 1 stage
60. ill change rapidly making longer term spectral characteristics difficult to see Larger Num Averages values result in a smoother spectral plot that represents stable frequency characteristics well but does not accurately show rapidly time varying signal characteristics This control only has an effect when Exponential is selected for the Averager Mode The Channel Select control allows you to choose which channel of audio you wish to see displayed in the spectrum analyzer window You can choose Left 15 Only Right Only or Mix The Mix selection sums the two audio channels together before performing the FFT analysis Right Only and Mix are only available for stereo audio files The analyzer also has three Zoom controls Zoom In Zoom Out and Zoom Full When zooming the graph will always try to center the graph on wherever the red user marker is located Each level of zoom halves the current frequency span of the display For instance if the current sample rate is 44100Hz the first level of zoom will cause the analyzer to display 11025Hz instead of 22050Hz worth of spectral data Zoom Full will always cause the analyzer to reset back to the full bandwidth of the signal 3 2 COEFFICIENT DISPLAY Wertical Zoom Channel Select Left Only Figure 5 Coefficient Display A few filters contain a Coefficient Display particularly the 1 CH Adaptive Filter and Reference Canceller Filter This graph can be useful in setting up these
61. illiseconds may fail to respond to breath group pauses and exchanges between speakers For most speech applications a Release Time of 200 400 milliseconds is recommended 12 3 3 7 Look Ahead Time i Look Ahead Time ms Figure 135 Look Ahead Time Lookahead controls the alignment of the envelope detector with the output signal Since the envelope is a smoothed version of the signal waveform level changes in the envelope will lag corresponding changes in the signal itself The applied LCE gain depends on the envelope level so the same lag is reflected in the applied gain The Lookahead control adjusts an internal delay that compensates for this lag The larger the Lookahead setting the earlier the gain adjustments will be shifted For most speech applications a Lookahead of 1 5 milliseconds is recommended 93 12 3 3 8 Reset Button Figure 136 Reset Button The Reset button resets all the LCE parameters to their default values which are 12 3 3 9 Store Button Figure 137 Store Button This button allows the user to store the current configuration to a user specified disk file that will not be lost when the computer is turned off Recall Figure 138 Recall Button This button allows the user to recall a previously stored configuration from any of the saved disk files previously generated using the Store button 12 3 3 10 Recall Button 94 13 PARAMETRIC EQUALIZER 13 1 OVERVIEW OF THE PARAMETRIC EQUALI
62. io the Cutoff Frequency can be incrementally lowered from its maximum frequency until the quality of the voice just begins to be affected achieving maximum elimination of high frequency noise The amount of volume reduction above the Cutoff Frequency can further be controlled by adjusting the Stopband Attenuation setting maximum volume reduction is 120dB The slope at which the volume is reduced from normal at the Cutoff Frequency to the minimum volume specified by Stopband Attenuation can also be controlled by adjusting the Transition Slope setting 45 Effect X Pass Filter Presets Custom Fir Sa inate a X PASS FILTER Filter Type Low pass Di Cutoff Frequency 4500 Hz i Transition Slope 7729 64 dB Octave E Stopband Attenuation 120 dB Sample Rate 14700Hz Cy Resample audio to 16kHz 0 0 dB Averager Mode Num Averages E El Channel Select Left Ony C Show Input Show Output A A in EP 1 A A eo no nr ki nt thy to aol iy 5 A pe pe ai ty a J 5 Ue Ds i ni a hfs a ye 4 1 Y t H Ta Track Peak 96 0 dB y 240 2d8 Zoom Figure 38 Lowpass Filter Controls 8 3 1 Filter Controls 8 3 1 1 Cutoff Frequency Cutoff Frequency Figure 39 Cutoff Frequency Specifies frequency in Hertz above which all signals are attenuated Frequencies below this cutoff are unaffected Maximum Cutoff Frequency depends upon the Sample Rate Cutoff Frequency can be adjusted in 1 Hz steps
63. le a microphone acoustically modified as a result of concealment a compensation filter that reshapes the audio to a normal spectral shape might be desirable The Hi Res Graphic Filter is essentially a 460 band graphic equalizer however instead of having 460 separate slider controls it allows the user to precisely 80 draw the desired filter shape on the computer screen using the mouse with as much or as little detail as desired Once the filter shape has been drawn a linear phase digital filter is constructed in the PC and transferred to the external processor The Normalize button allows the user to shift the entire filter curve up until the highest point is at OdB A Store and Recall capability is also provided to allow the user to store commonly used filter shapes to disk memories so that they can be recalled later Effect Graphic EQ Es Presets Custom ITKE 22K 27K 3 2K 27K 4 2K 47K 54K BTE 739K 9 4K 10 5K 124K 13 9K 15 4K 16 8K 18 3K Sample Rate 44100Hz A Resample to 16kHz 12 0 dB Averager Mode ds Exponential Num Averages Channel Select Left Only O Show Input W Show Output Ay HP Po FA Tiel Te DA ae ca Track Peak 54 0dB 22050Hz Figure 109 Graphic EQ Main Window Graphic EQ Mode 81 Effect Graphic EQ Presets Custom li Use the Spectrum Analyzer Below to Build a Hi Res Graphic Filter 1 To add a point click inside the graph 2 To delete a point click the poi
64. lider controls found on analog graphic equalizers found on many consumer stereo systems and thus should be very familiar to even the novice user However unlike analog graphic equalizers this digital equalizer has some very powerful additional capabilities For example the Normalize button allows the user to instantly move all slider controls up until the top slider is at OdB The Zero All button instantly sets all the sliders to OdB while the Maximize button instantly sets all the sliders to 100dB The All Down 1dB button instantly moves all sliders down in 1dB increments while the All Up 1dB button moves all sliders up in 1dB increments None of these functions are available in an analog graphic equalizer Notice also that the 20 sliders are spread across the selected Bandwidth and that the frequency spacing is optimized for voice processing Additionally since a computer with a disk drive operates the equalizer a Store and Recall capability is available This allows the user to store commonly used slider configurations in disk memories so that they can be instantly recalled later whenever they are needed without having to manually adjust the slider controls 11 1 2 Overview of Hi Res Graphic In some applications it may be necessary to precisely reshape the spectrum of input audio prior to passing it through successive filter stages For example if the audio is from a microphone which has an unusual frequency response curve for examp
65. lmost always be selected but the Predicted setting can be useful when configuring the filter allowing the user to hear 21 exactly what is being removed by the filter 4 3 6 Freeze Filter Used to enable or disable filter adaptation When Freeze is off the filter adapts according to its settings When Freeze is on the filter never adapts regardless of the other settings 4 3 7 Clear Filter Used to reset the coefficients of the 1 CH Adaptive Filter Clearing a filter is useful when the audio characteristics change dramatically so that the filter can readapt to a new clean solution Clearing is also useful in the case of a filter crash when the filter coefficients diverge to an unstable state usually in response to a large and abrupt change in the signal coupled with a fast adapt rate 22 S REFERENCE CANCELLER FILTER 5 1 OVERVIEW OF THE REFERENCE CANCELLER FILTER The Reference Canceller adaptive filter is used to automatically cancel from the Primary channel any audio which matches the Reference channel For example the Primary channel may be microphone audio with desired voices masked by radio or TV noise The radio TV interference can be cancelled in real time if the original broadcast audio usually available from a second receiver is simultaneously recorded to the Reference channel The Reference Canceller Filter is a forensic plug in that has been specifically designed to work with VST host audio editing systems
66. mb Frequency Figure 32 Comb Frequency Specifies fundamental frequency in Hertz of comb filter Notches are generated at multiples or harmonics of this frequency When Auto Tracking is turned on this control does not appear 7 2 2 Notch Limit Notch Limit F655 Hz Figure 33 Notch Limit Specifies frequency in Hertz above which no notches are generated Minimum Notch Limit is half the Comb Frequency while maximum Notch Limit depends upon the sample rate 7 2 3 Notch Depth Notch Depth P 120 08 Figure 34 Notch Depth Specifies the depth of notches that are generated Notch Depth is adjustable from O dB to 120 dB in 1 dB steps 41 7 2 4 Notch Harmonics Harmenics a All Even gt Odd Figure 35 Notch Harmonics Specifies whether notches will be generated at All Odd or Even multiples or harmonics of the Comb Frequency If for example the Comb Frequency is set to 60 0 Hz then selecting All will generate notches at 60 Hz 120 Hz 180 Hz 240 Hz 300 Hz etc Selecting Odd will generate notches at 60 Hz 180 Hz 300 Hz etc Selecting Even will generate notches at 120 Hz 240 Hz 360 Hz etc 7 2 5 Auto Tracking i Auto Tracking Figure 36 Auto Tracking Enables Auto Tracking for the Comb filter When enabled the Comb Frequency control is no longer available and the filter will determine the fundamental frequency to use When disabled the last calculated frequency will then be loaded int
67. me and go in the output audio or if the filter trace display coefficients seem to be changing rapidly you probably need to reduce the adapt rate 35 6 3 7 Manual Mode Adaptive Mode E Run Analyzer E Manual Build Figure 28 Manual Mode In Manual mode the above figure is displayed and gives the user access to the Run Analyzer Clear and Build controls This puts the plug in in the SIF mode The Clear button is used to zero the averager memory and causes the averaged spectrum to be recalculated anew The Run Analyzer button allows the user to start or stop the update of the averaged spectrum The Build button builds the spectral inverse filter based on the original input audio spectrum and the SIF control settings Once the filter build is complete the calculated spectral inverse filter curve will be displayed as a green trace in the Filter Display area Hint Before clicking the Build button it is recommended that the spectrum analyzer be stopped using the Run Analyzer button to allow experimentation with the control settings for the same input spectrum 6 3 8 Display Num Averages a a Channel Select LeftOnly w O Show Input D Show Output 120 0 de Figure 29 SIF Display 36 The display trace is used to view the filter input and output audio and the SIF filter response The input audio is always shown in yellow the output trace in blue and the filter trace in green The L
68. ment of SIF operation The user can specify the expected spectrum so that the output audio is reshaped to a flat pink or voice like curve An adapt rate setting controls the update rate for the spectral average which in turn determines how quickly the filter responds to changes in the input audio Upper and lower limit controls allow the user to specify the range over which equalization is applied and a mode setting controls whether frequencies outside the equalization range are attenuated or left unaffected The amount of spectral correction is adjustable using the Filter Amount control The user can enable the auto gain functionality to ensure that the output audio level is maintained at approximately the same as the input audio level If the user disables the auto gain an output gain slider is available to manually boost the level of the output signal As an aid to visualizing the filter operation the user can view the input and output audio traces as well as the filter coefficient trace 28 Effect Spectral Inverse Filter Presets Custom ly Aor CARDINAL SPECTRAL INVERSE Filter Operation Output Shape Filter Amount 400 l E Flat a Equalize Voice 0 Pink D Attack Noise l Output Gain E Auto Gain 0 0 dB E Voice Adaptive Mode Lower Voice Limit i Manual Adapt Rate 01282 i Upper Voice Limit 5000 Hz Sample Rate 14700 Hz va Resample audio to 16kHz 60 0 db Num Averages Channel Select Let Only y O
69. nt and then click the Delete button 3 To modify a point click the point and then drag it to change the decibel level Clear All Sample Rate 44100Hz Cy Resample to 16kHz Averager Mode Mum Averages a Channel Select Left Only 0 Show Input O Show Output A i a 7 mw T P i Tey eaa an pt EE vi Pat Utd a Wan a O Track Peak 96 0 dB 72050Hz Close Figure 110 Graphic EQ Main Window Hi Res Graphic Mode 11 1 MAIN PLUG IN WINDOW Figure 109 and Figure 110 above show the plug in as it would be displayed if used in Adobe Audition in Graphic EQ mode and Hi Res Graphic mode respectively The buttons above and below the plug in window will appear differently depending upon your audio editing environment Please consult the documentation on your specific audio editing environment for details of how to use these features This window provides access to all the functionality of the plug in A spectrum 82 analyzer is provided as an aid in determining the characteristics of the target audio Details of the plug in are described in the following sections 11 2 EQUALIZER CONTROLS 11 2 1 Equalizer Mode Figure 111 Equalizer Mode The Equalizer Mode combo box allows the user to select either the Graphic EQ mode or the Hi Res Graphic mode When switching from one mode to the other the previous mode s controls are remembered and preserved 11 2 2 Graphic EQ Controls 11 2 2 1 Slider Controls GMB 28
70. ntain any skips but may take longer than real time to process the file 5 3 3 Adapt Rate Adapt Rate 7 Auto Normalize a Figure 16 Adapt Rate Used to set the rate at which the adaptive filter adapts to changing signal conditions An adapt rate of 1 provides very slow adaptation while an adapt rate of 5884 provides fastest adaptation A good approach is to start with an adapt rate of approximately 100 200 to establish convergence and then back off to a smaller value to maintain cancellation When Auto Normalize is turned on the specified adapt rate is continuously scaled based upon the input signal level This scaling generally results in faster 25 filter convergence without greatly increasing the risk of a filter crash It ts recommended that Auto Normalize be enabled for most speech signal processing 5 3 4 Conditional Adaptation i Sj o f Sj Figure 17 Conditional Adaptation For advanced users only Novice users should keep Conditional Adaptation set to Always The threshold setting has no effect when Always is selected Conditional Adaptation allows the adaptive filter to automatically Adapt Freeze based upon signal bargraph levels This can be very useful in situations where there are pauses or breaks in the speech being processed Hint Conditional adaptation is useful for maintaining adaptation once the filter has converged Recording environment factors such as air temperature and motion in the
71. o the Comb Frequency control NOTE It ts advised that you adjust the Comb Frequency manually to get it as close as possible to the desired fundamental frequency before enabling Auto Tracking 42 8 X PASS FILTER 8 1 OVERVIEW OF THE X PASS FILTER The X Pass Filter provides six different bandlimiting filters in a single plug in The following filters are included po Filter _ Application Decrease the energy level of high frequency noises Decrease the energy level of low frequency noises Bandpass Decrease the energy level of noises above and below a passband region Bandstop Decrease the energy level of noises within a stopband region Isolate a narrow band signal Only one filter can be operational at a time While they all share similar controls they perform different functions as detailed below Whichever filter you select its transfer function or envelope is displayed in the Spectrum Analyzer as a light gray trace 43 Effect X Pass Filter Presets Custom Mos CARDINAL X PASS FILTER Filter Type f Lower Cutoff Frequency B50 Ha f Upper Cutoff Frequency 5000 Hz i Transition Slope 202 11 dB Octave l f Stopband Attenuation 420 dB i Sample Rate 14700Hz 5 Resample audio to 16kHz Averager Mode Num Awverages a Z Channel Select C ketony Je O Show Input GO Show Output J Track Peak Figure 37 X Pass Main Window 8 2 MAIN PLUG IN WINDOW Figure 37 above show
72. ode is changed 31 6 3 2 Output Shape Output Shape Flat Pink i Voice Figure 23 Output Shape In this block the user can select the target spectral shape that the filter attempts to achieve The SIF has an inherent spectral flattening effect on the audio The selected spectral shape is applied to further reshape the audio spectrum The following output shapes are available e Flat no additional shaping after ASIF flattening e Pink 3 dB per octave rolloff above 100 Hz is applied in addition to ASIF flattening e Voice 6 dB octave rolloff above and below 500 Hz in addition to ASIF flattening Note Changing the output shape does not require an adaptation period to arrive at a good solution Because a full average spectrum is maintained regardless of the output shape setting the new output shape takes effect instantaneously in both the output audio and the display traces However since the auto gain adapts based on the actual applied filter with the shaping curve taken into account there may be some adaptation time required to reach a stable auto gain value after the shaping curve is changed 6 3 3 Filter Amount Filter Amount 100 Figure 24 Filter Amount This setting controls the degree to which the SIF can affect the signal with 0 corresponding to no filtering and 100 corresponding to full filtering In general it is best to use the minimum Filter Amount setting that produces the desired resul
73. ower and Upper Voice Limits are shown in red The controls for this display are identical to the controls for the Spectrum Analyzer except for the analyzer is always in the Exponential averaging mode hence no choice is given to change the averaging mode 37 7 COMB FILTER 7 1 OVERVIEW OF THE COMB FILTER The Comb filter is used to remove or notch out harmonically related noises noises which have exactly equally spaced frequency components such as power line hum constant speed motor generator noises etc from the input audio The filter response consists of a series of equally spaced notches which resemble a hair comb hence the name Comb filter Adjust the Comb Frequency to the desired spacing between notches also known as fundamental frequency Set the Notch Limit to the frequency beyond which you do not want any more notches Set the Notch Depth to the amount in dB by which noise frequency components are to be reduced Normally the Notch Harmonics option will be set to All causing frequencies at all multiples of the Comb Frequency within the Notch Limit to be reduced However certain types of noises have only the odd or even harmonic components present In these situations set the Notch Harmonics option to either Odd or Even The Auto Tracking feature allows the filter to automatically hone in on the exact fundamental frequency and continually track it even if it shifts slightly 38 Effect Comb Pres
74. pplied within each of 20 distinct groups of frequency bands offering much more precise control of the spectral subtraction than is available in the Noise Reducer mode though it does take more time to setup In the Noise Reducer mode all 20 control sliders are linked together and are adjusted using the first slider Adjusting the master slider allows the user to precisely control the amount of noise reduction being applied to all frequency bands the greater the value the more aggressive the operation of the Noise Reducer Because large amounts of noise reduction invariably create audible birdy noise artifacts in the output audio due to the nature of adaptive frequency domain processing the user should always try to minimize the amount of noise reduction being applied to achieve the best balance between maximal noise reduction and minimal audible artifacts Finally for convenience an Output Gain control and Output level bargraph are provided to enable the user to adjust the processed output signal to maximum level for better listening and recording 60 Effect Spectral Subtraction Presets Custom Er Y SPECTRAL SUBTRACTION Filter Controls N Reducer w 551 17K 2 8K 3 5K 5 0K 61K 7 2K 8 3K 3 4K 10 5K 11 6K 12 7K 13 8 14 9K 16 08 17 1K 18 2K 13 3K 20 4K 21 5K EEE resample to 16kHz Averager Mode Exponential w Num Averages E EJ Channel Select O Lett Only e C Show Input Show
75. rence Canceller Filter Spectral Inverse Filter Spectral Subtraction Filter X Pass Filter 6 filter set The MiniLab Suite includes all the above filters with the exception of the Reference Canceller and Filter Chain plug ins The Advanced Suite contains all the MiniLab plug ins plus an additional stand alone application called Cardinal Batch Utility 2 SOFTWARE INSTALLATION 2 1 SOFTWARE INSTALLATION The software requires that a host audio editing environment that supports the VST standard be installed The computer on which the audio editing environment is installed must meet all the requirements of that audio editing environment for the plug in to work Please consult the documentation of your audio editing environment for its requirements NOTE Because of the processing requirements of the plug in the filters will benefit from a faster computer To start the installation insert the CD containing the plug in software into your PC If the installation does not begin automatically run the file Setup exe located in the root folder of the CD You can run this file by clicking on the Start menu then on Run and typing X Setup exe where X is the drive letter of your CD ROM drive Follow the instructions on the screen to complete the installation If you have any questions or problems with the installation please contact DAC for further assistance In addition to its own installation directory the VST plu
76. s down 1 while maintaining the desired curve This is useful when the desired shape is found but you want the filter to be more aggressive as a whole No slider will be allowed to go more than 100 9 3 4 7 Store Recall Recall Figure 75 Store and Recall Buttons These buttons allow the user to store and recall a slider configuration to a user specified disk file that will not be lost when the computer is turned off 9 3 5 Noise Reducer Controls The following sections apply only to the Noise Reducer mode 9 3 5 1 Slider Control Figure 76 Noise Reducer Slider Controls When in Noise Reducer mode all the slider controls are linked to the first slider control This slider is used to specify the amount of noise reduction that the spectral subtraction attempts to apply to the input signal across all frequency bands Adjustment range is O no attenuation to 100 maximal attenuation in 1 increments 66 10 MULTI BAND FILTER 10 1 OVERVIEW OF THE MULTI BAND FILTER The Multi Band filter provides two different filters in a single plug in the Multiple Notch and Multiple Slot Only one filter can be operational at a time 10 1 1 Overview of Multiple Notch The Multiple Notch filter is used to remove or notch out single frequency noises such as tones or whistles with minimal effect on signal frequencies other than the notch frequency Single notches can be added one at a time and configured individually
77. s the plug in as it would be displayed if used in Adobe Audition The buttons above and below the plug in window will appear differently depending upon your audio editing environment Please consult the documentation on your specific audio editing environment for details of how to use these features The Filter Type control selects which of the six filters is currently engaged Only one filter is operational at a time If you desire to apply more than one of the filters on a given piece of audio you must either use multiple instances of the plug in in an effects chainer not available in all audio editor environments or render the audio multiple times using a different filter each time When a filter is selected that filter s controls are displayed in the main window 44 This window provides access to all the functionality of the plug in A spectrum analyzer is provided as an aid in determining the characteristics of the target audio Details of the plug in are described in the following sections 3 3 LOWPASS FILTER The Lowpass filter is used to decrease the energy level lower the volume of all signal frequencies above a specified Cutoff Frequency thus reducing high frequency noises such as tape hiss from the input audio The Lowpass filter is sometimes called a hiss filter The Cutoff Frequency is usually set above the voice frequency range so that the voice signal will not be disturbed While listening to the filter output aud
78. specified Slot Gain while subsequent slots taper to smaller gains as frequency increases The higher the Gain Factor the more gradual the taper A Gain Factor of 0 0 produces the most severe taper and means effectively that there are no harmonics at all A Gain Factor of 1 0 means that slots have uniform gain at the Base Slot gain setting 10 3 3 8 Upper Limit i Upper Limit 4080 Hz l Figure 101 Upper Limit Upper Limit defines how many slots there are in the group If a harmonic tonal noise only extends up to a certain frequency it may be undesirable to slot out all multiples of the base frequency when only a few are needed In this case set the Upper Limit just above the highest frequency where a slot is desired slots will be added up to that limit and no slots will be added above the limit 10 3 3 9 Add Slot Add Slot Figure 102 Add Slot Button Adds a new single slot at the frequency indicated in the Slot Frequency box and by a marker on the visualization axis The slot is added with default settings and the user is presented with controls to adjust the frequency width and gain of the slot 10 3 3 10 Add Group 77 Add Group Figure 103 Add Group Button Adds a new slot group with its base slot at the frequency indicated in the Slot Frequency box and by a marker on the visualization axis The slot group is added with default settings and the user is presented with controls to adjust the frequency width g
79. t When Equalize Voice mode is used a lower Filter Amount can reduce artifacts that result from a fast adapt rate so the Filter Amount can be used to help strike a balance between responsiveness and stability When Attack Noise mode is used to reduce bandlimited noise a lower Filter 32 Amount setting will often be a better choice to prevent the elevation of background noises Note Changing the Filter Amount setting does not require an adaptation period to arrive at a good solution Because a full average spectrum is maintained regardless of the setting the new filter amount setting takes effect instantaneously in both the output audio and the display traces However since the auto gain adapts based on the actual applied filter with filter amount taken into account there may be some adaptation time required to reach a stable auto gain value after the filter amount is adjusted 6 3 4 Output Gain and Auto Gain Qutput Gain Auto Gain QE Figure 25 Output Gain and Auto Gain These controls provide two options for adjusting the level of the SIF output When Auto Gain is enabled the SIF automatically monitors the input and output levels and applies a gain value that matches the output level to the input level When Auto Gain is disabled the user can use the Output Gain setting to specify the amount of boost applied to the SIF output If Auto Gain is enabled then the Output Gain slider is ignored The Auto Gain is an adap
80. t signal level changes Thus the LCE decreases the dynamic range of the signal for levels in the Compression region As an example a 2 1 compressor would produce an output level change of only 10 dB when the input signal changes by 20 dB Compression is often used to correct near party far party level differences boosting the lower level far party speech relative to the louder near party speech Compression also eases listening especially for noisy audio Compressors are generally preferred over AGCs since input signal level differences are more closely preserved 87 e In the Expansion region levels are adjusted so that output signal level changes are larger than their corresponding input signal level changes Thus the LCE increases the dynamic range of the signal for levels in the Expansion region Expansion is the opposite of compression For example a 1 3 expander would produce an output level change of 30 dB when the input signal changes by 10 dB A 1 2 expansion would restore a signal s dynamic range following a 2 1 compression Expansion is also used to attenuate objectionable low level background noise that is below the voice level Figure 123 shows an example LCE curve In this example the Limiting Threshold is set at 20dB and the Compression Threshold is set at 60dB The Compression Ratio is 2 1 and the Expansion Ratio is 1 3 una ic IRE Te oa Eon o is ecisite ebaiaio Expansion Compression Limiting Region Redion Redion
81. tack Noise Figure 22 Filter Operation In this block the user can select the operational mode of the filter If the filter is being used to correct spectral coloration the Equalize Voice mode should be selected If the filter is being used to remove bandlimited noise the Attack Noise mode should be selected Note The Filter Operation mode selection only affects the behavior of the filter outside the range selected by the upper and lower limits In Equalize Voice mode the frequency ranges outside the limits are attenuated In Attack Noise mode the frequency ranges outside the limits are left unaffected subject to a transition region near the limits If the auto gain is disabled and the manual gain is set to O dB frequencies outside the limits and transition regions will be unaffected However if gain is applied the gain will be reflected over the entire frequency range See the section on Upper and Lower Voice Limits for more information on selecting the range Note Changing the filter operation mode does not require an adaptation period to arrive at a good solution Because a full average spectrum is maintained regardless of the mode setting the new mode takes effect instantaneously in both the output audio and the display traces However since the auto gain adapts based on the actual applied filter with operational mode taken into account there may be some adaptation time required to reach a stable auto gain value after the m
82. the network or manually a persistent license will be installed on the machine If the software is not activated within 30 days it will revert back to Demo mode You must uninstall the software which will deactivate it as well before installing the software on another machine If you have purchased one of the suites you will only need to enter your serial key one time and activate it All the other plug ins in the suite will recognize the license and run normally If you purchase plug ins individually then you will have to enter a separate serial key for each plug in you purchased 13 3 GENERAL PLUG IN CONCEPTS 3 1 LAUNCHING A PLUG IN The Cardinal MiniLab suite of plug ins are not stand alone applications They are plug ins that execute from within a host environment such as Acon Digitals Acoustica Adobe s Audition Sony s SoundForge and other similar audio editors Plug ins are launched from host applications in various ways depending on the host Usually the plug in can be found in the Effects Tools or VST menu If the currently loaded audio file is not compatible with a plug in the plug in choice will usually be grayed out disabled In general upon startup the host application will scan a well known VST plug in folder for valid VST plug ins Usually new folders can be added to the scanned list Please consult your audio editor manual to determine how to recognize and launch a VST plug in 3 1 SPECT
83. tive value whose rate of change depends on the same Adapt Rate slider setting that controls filter coefficient averaging This means that when the filter response changes rapidly and dramatically the auto gain will take some time to catch up to these changes In particular the output audio may clip when user settings are changed in a ways that have a boosting effect such as switching from a pink to a flat shaping curve adjusting the filter amount or increasing the size of the ASIF region in Equalize Voice mode so that some frequencies that had been heavily attenuated are now present While these settings changes will take effect immediately the Auto Gain may take some time to adapt to the change For this reason when the user expects to be making many changes in the settings it is often better to disable Auto Gain and instead choose a manual gain setting that avoids clipping 33 6 3 5 Lower and Upper Voice Limits Lower Voice Limit Upper Voice Limit QO Hz Figure 26 Lower and Upper Voice Limit The Lower and Upper Voice Limits allow the user to specify the frequency range or SIF region over which the SIF is applied Two red markers on the Spectrum Analyzer indicate where the lower and upper voice limits are located Viewing audio on the display trace while manipulating the markers is an easy way to identify where your SIF region limits should fall In Equalize Voice mode the SIF region is typically chosen to
84. toff Frequency is O Hz while the maximum Lower Cutoff Frequency is 10 Hz below the Upper Cutoff Frequency Lower Cutoff Frequency can be adjusted in 1 Hz steps NOTE The Lower Cutoff Frequency can never be set higher than 10 Hz below the Upper Cutoff Frequency 8 5 1 2 Upper Cutoff Frequency F Upper Cutoff Frequency B00 Ha Figure 48 Upper Cutoff Frequency Specifies frequency in Hertz above which all signals are attenuated Frequencies between this cutoff and the Lower Cutoff Frequency are unaffected Minimum Upper Cutoff Frequency is 10 Hz above the Lower Cutoff Frequency while the maximum Upper Cutoff Frequency depends upon the Sample Rate Upper Cutoff Frequency can be adjusted in 1 Hz steps NOTE The Upper Cutoff Frequency can never be set lower than 10 Hz above the Lower Cutoff Frequency 8 5 1 3 Transition Slope Transition Slope 202 11 dBiOctave Figure 49 Transition Slope Specifies slope at which frequencies below the Lower Cutoff Frequency and above the Upper Cutoff Frequency are attenuated in dB per octave Sharpest attenuation occurs when Transition Slope is set to maximum while gentlest 91 attenuation occurs when Transition Slope is set to minimum Note that the indicated value changes depending upon the Cutoff Frequency and Sample Rates Also note that the Lower and Upper Transition Slopes always have different values this is because the frequency width of an octave is proportional to Cutoff Frequen
85. ve 3000 Hz so that the voice signal will not be disturbed While listening to the filter output audio the Upper Cutoff Frequency initially set to its maximum frequency can be incrementally lowered until the quality of the voice just begins to be affected achieving maximum elimination of high frequency noise The amount of volume reduction outside the passband region can further be controlled by adjusting the Stopband Attenuation setting maximum volume reduction is 120dB The slope at which the volume is reduced from normal at each Cutoff Frequency to the minimum volume specified by Stopband Attenuation can also be controlled by adjusting the Transition Slope setting Effect X Pass Filter Presets Custom Ty Ah K CARDINAL X_PASS FILTER Filter Type Bandpass v Lower Cutoff Frequency ERTE j E Upper Cutoff Frequency 5000 Hz j Transition Slope 202 11 dB Octave f Stopband Attenuation er i Sample Rate 14700Hz Resample audio to 16kHz Averager Mode Num Averages j Zj Channel Select C Le ony 0 Show Input GO Show Output J Track Peak D Zoom Figure 46 Bandpass Filter Controls 8 5 1 Filter Controls 90 8 5 1 1 Lower Cutoff Frequency E Lower Cutoff Frequency 2250H i Figure 47 Lower Cutoff Frequency Specifies frequency in Hertz below which all signals are attenuated Frequencies between this cutoff and the Upper Cutoff Frequency are unaffected Minimum Lower Cu

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