PCAP II Plus ENHANCED-CAPABILITY PERSONAL COMPUTER AUDIO PROCESSOR USER’S MANUAL DIGITAL AUDIO CORPORATION A DRI COMPANY
PCAP II Plus ENHANCED-CAPABILITY PERSONAL COMPUTER AUDIO PROCESSOR User’s Manual December 2004 Version 1.4 DIGITAL AUDIO CORPORATION A DRI COMPANY 4018 Patriot Drive One Park Center Suite 300 Durham, NC 27703 Phone: 919 572 6767 Fax: 919 572 6786 sales@dacaudio.com www.dacaudio.com Copyright © 2004 by Digital Audio Corporation. All rights reserved.
TABLE OF CONTENTS ACKNOWLEDGEMENT.......................................................................................................... vi WHAT’S NEW......................................................................................................................... vii FOREWORD........................................................................................................................... viii 1: . SYSTEM BASICS..............................................................................
4.4.6: 4.4.7: 4.4.8: 4.4.9: 4.4.10: 4.4.11: 4.4.12: 4.4.13: 4.4.14: 4.4.15: 4.4.16: 4.4.17: Bandstop Filter .................................................................................................. 62 Comb Filter........................................................................................................ 65 Notch Filter........................................................................................................ 68 Multiple Notch Filter.........................................
: DAC1024T AND DAC4096T ADAPT RATE SETTINGS................................................. 159 8: PCAP II SPECIFICATIONS ..............................................................................................
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LIST OF FIGURES Figure 1-1 PCAP II Plus Basic System Configuration.................................................................. 1 Figure 2-1 PCAP II Plus Master Control Icon............................................................................. 8 Figure 2-2 Master Control Panel ................................................................................................. 9 Figure 3-1 PCAP II Plus Connection Diagram...........................................................................
Figure 4-22 Highpass Filter Graphical Description..................................................................... 58 Figure 4-23 Bandpass Filter Control Window............................................................................ 59 Figure 4-24 Bandpass Filter Graphical Description .................................................................... 61 Figure 4-25 Bandstop Filter Control Window............................................................................
Figure 4-67: ASIF custom curve drawing window................................................................... 118 Figure 4-68: Audio trace view for ASIF Example 1................................................................ 122 Figure 4-69: Filter trace view for ASIF Example 1 ................................................................. 123 Figure 4-70: Filter response from Example 1 with Filter Amount 85%....................................
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ACKNOWLEDGEMENT The Enhanced-Capability Personal Computer Audio Processor, model PCAP II Plus, is a fourthgeneration PC-based digital audio filtering system. Like its predecessors, the original PCDF4096, PCAP, and MCAP products, this product was inspired by Mr. James “Jim” Foye of the United States Postal Service and Mr. John “Jack” Losinski of the Federal Bureau of Investigation.
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WHAT’S NEW? In addition to the base capabilities of the previous-generation PCAP II product, the PCAP II Plus system offers the following enhanced capabilities: • More Processing Power o Two additional general filter stages now available (increased to 6 from 4) o Two new broadband filter stages now available o 50% increase in adaptive filter taps (increased to 6144 from 4096) o New 1.8GHz floating-point processing core for implementing enhanced filtering functions (e.g.
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FOREWORD The PCAP II Plus Enhanced-Capability Personal Computer Audio Processor is the latest generation of the most popular forensic audio analysis and processing system used by the law-enforcement and intelligence communities.
The PCAP II Plus is applicable to a broad spectrum of voice and similar audio signals. It attacks a wide variety of noises in forensic applications. Body microphone, cassette, microcassette, telephone, broadcast, and hi-fi audio signals can all be processed efficiently, since the PCAP II can be set to operate at bandwidths of 3.2, 5.4, 6.5, 8.0, 11, and 16 kHz. The PCAP II is designed to replace an entire rack of audio processing equipment.
1: . SYSTEM BASICS 1.1: SYSTEM CONFIGURATION The basic configuration of the PCAP II Plus system is illustrated in the following (Figure 1-1): RS232 Interconnect Analog or Digital Input Source Audio Player PCAP II Plus Analog or Digital Output Computer Enhanced Audio Recorder Headphones Figure 1-1 PCAP II Plus Basic System Configuration For integration with a Digidesign Pro Tools system (e.g.
Note that in this configuration the digital OUTPUT RATE should be configured for AutoSYNC for proper digital audio interaction with Pro Tools. The PCAP II Plus Master Control program is written to be run on Windows® 98 Second Edition, Me, NT 4.0, 2000, and XP.
1.2: EXTERNAL PROCESSOR CAPABILITY The PCAP II Plus EXTERNAL PROCESSOR unit is a high-performance, self-contained digital signal processor and contains 38 DSP microprocessors, which are allocated as follows: • 24 FIR filter processors which can be configured as 1 to 6 independent audio processors. These flexible processors may be combined into mono and stereo configurations.
1.3: EXTERNAL PROCESSOR FRONT PANEL The front panel controls are arranged into three logical groups: headphone MONITOR controls, STAND-ALONE controls, and INPUT LEVEL controls. The MONITOR controls allow the user to listen to either the INPUT or OUTPUT signals with a pair of stereo headphones connected to the 1/4" PHONES jack. Switching the headphones between the INPUT and OUTPUT signals does not alter the signal flow to the LEFT OUT and RIGHT OUT line output RCA connectors.
1.4: EXTERNAL PROCESSOR REAR PANEL The rear panel of the PCAP II Plus external processor appears as follows (Figure 1-): Figure 1-4 PCAP II Plus Rear Panel DC power is provided to the unit through the external POWER jack by either the supplied external AC adaptor or direct connection to a 9 - 18 VDC source. The POWER switch must be switched to the ON position in order for the unit to operate.
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2: INSTALLATION INSTRUCTIONS 2.1: CAUTIONS TO USER To install the PCAP II Plus hardware and software, the user must have a good working knowledge of IBM PC-compatible computers and the Microsoft Windows operating environment. Particularly, the user must know which RS232 COM ports are COM1, COM2, COM3, etc.
2.2: INSTALLATION PROCEDURE 1. Carefully remove the PCAP II Plus external processor from the shipping container. Confirm that the AC power adapter, RS232 cables (2), demonstrator audio CD, and software disk(s) are included. Also confirm that any optional accessories are included. 2. Connect the AC power cord to the PCAP II Plus rear panel POWER connector. Keep the POWER switch OFF for now. 3. With the PCAP II Plus POWER switch OFF, plug the AC power adapter into an AC outlet. 4.
Double click on this icon now to run the program. A screen similar to Figure 2-2 should appear: Figure 2-2 Master Control Panel If an error message is displayed, it is possible that the software is not configured for the correct COM port (software defaults to COM1). If you know to which COM port the PCAP II Plus external processor is connected, configure the Master Control program for the correct COM port by following the procedure in Section 6.1: .
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3: GETTING STARTED Operation of the PCAP II Plus system is highly intuitive; most operators can quickly learn while using. The Fast Start procedure in Section 3.1: should allow first time users to quickly begin processing audio and utilizing the basic enhancement capabilities of the PCAP II Plus. For a lesson in operating the PCAP II Plus controls, it is recommended that the user also complete the Tutorial in Section 3.2: . Section 3.
DIGITAL I/O RCA jack to the corresponding digital input jack of your digital Enhanced Audio Recorder. 3. Connect your stereo headphones to the PHONES jack on the PCAP II Plus external processor front panel as shown in Figure 3-1. Make sure that the MONITOR pushbuttons are set to OUTPUT, LEFT, and RIGHT (the three buttons are “out, in, and in”, respectively). Turn the phones VOLUME control to MIN.
If you wish at this point to discontinue the Fast Start procedure and experiment with the PCAP II Plus Master Control Panel on your own, please feel free to do so - you will not damage anything. If, however, you still feel unsure of what to do, please continue the Fast Start procedure as follows: 8. From the PCAP II Plus Master Control Panel menu bar, click on File. This will cause the following pulldown menu to appear: Figure 3-2 Fast Start File Pulldown Menu 9.
10. The Setup Files box should contain a list of all PCAP II Plus setup files on your hard disk. The installation procedure in Section 2.2: should have installed several setup files which have the .DAC extension onto your hard disk. Included in these setup files are: File Description PHONE.DAC “Landline” telephone audio enhancement GSMBUZZ.DAC GSM mobile phone RF interference noise reduction BODYWIRE.DAC “Body wire” receiver static noise reduction and voice enhancement MOTOR.
3.2: PCAP II TUTORIAL This brief tutorial should allow the user to quickly learn the basic operation of the PCAP II Plus controls. It should require about 1 hour for completion. NOTE: A subset of the PCAP II Plus functions are utilized in this tutorial. Refer to Chapter 4: for detailed information on all of the PCAP II Plus functions. Tutorial Steps: Figure 3-4 Tutorial Master Control Panel 1. With the installation procedure in Section 2.
2. Use the mouse* to set the Input HPF, Limiter, Output HPF, AGC, System Bandwidth, Input Mode, Number of General Filter Stages, Adapt Enabled, and Processor Enabled buttons as they are shown in Figure 3-4. Do not set up the Filter blocks and Equalizer blocks at this time. 3. Connect the LEFT and RIGHT outputs (AUDIO OUT jacks) of the audio player to the LEFT and RIGHT ANALOG INPUTS jacks on the PCAP II Plus external processor; make sure that the INPUT SELECT switch is set to ANALOG.
11. Switch the Output HPF in and out by clicking on the Active button in the Output HPF block. You should hear the low frequency effects of this control. Restore the button such that the Output HPF is in. 12. Click on the Config button in the Limiter block to activate the Limiter Setup Window. The following window (Figure 3-5) will appear: Figure 3-5 Tutorial Limiter Setup Window When the limiter is active, red markers will appear in the Input Level bargraphs on the Master Control Panel.
Restore the Active button to the out position (LED unlit) and restore the INPUT LEVELS controls to normal level. 17. Click on the AGC Setup button in the Output Levels block. The following window (Figure 3-6) will appear: Figure 3-6 Tutorial AGC Setup Window Use the mouse to set the Release Time to 200 milliseconds and the Maximum Gain to 20dB, as shown in Figure 3-6. Click on OK when done. 18. Reduce the INPUT LEVEL until the Output Level Bargraphs indicate a peak signal level of approximately -36dB.
Figure 3-7 Tutorial Filter Selection Window 21. Click on the Select button in the Filter 2 block and select Lowpass. 22. Click on the Control button in the Filter 1 block. The window in Figure 3-8 will appear: Figure 3-8 Tutorial 1-Channel Adaptive Filter Control Window This is the 1-Channel Adaptive Filter Control Window. It is used to select Filter Size, Adapt Rate, and Prediction Span. It also permits configuration of the filter for Conditional Adaptation. 23.
unlit). You should notice a reduction in background noise whenever the filter is in. Press the Clear button to cause Filter 1 to readapt to the input signal. 25. Restore the Filter Active button to the out state (LED unlit) and click on OK to exit the 1Channel Adaptive Filter Control Window. 26. Click on the Control button in the Filter 2 block. The window in Figure 3-9 will appear: Figure 3-9 Tutorial Lowpass Filter Control Window This is the Lowpass Filter Control Window.
Figure 3-10 Tutorial Equalizer Selection Window Use the mouse to click on the 20-Band Graphic. The Equalizer block should now indicate that the selected mode is 20-Band Graphic. 33. Click on the Control button in the Equalizer block. The window in Figure 3-11 will appear. This is the 20-Band Graphic Equalizer Control Window. The twenty vertical scroll bars, or "sliders", are used to set the equalizer attenuation for each of the 20 frequency bands.
Figure 3-12 Tutorial Store 20-Band Graphic Equalizer Window 36. Enter a the filename “Test1.tbg” (the OK button will be enabled when a valid filename is entered). (We will recall these settings later.) Once you have clicked the OK button, the system will automatically return to the 20-Band Graphic Equalizer control window. 37. While listening to the OUTPUT audio with the headphones, adjust the sliders and listen to how each affects the sound.
40. Click on the Recall button to bring up the Recall 20-Band Graphic Equalizer window. The window in Figure 3-13 will appear. Figure 3-13 Tutorial Recall 20-Band Graphic Equalizer Window 41. Select the file “Test1.tbg” that was saved in Step 36 to recall the slider settings previously stored. Once you have clicked this button, the system will automatically return to the 20-Band Graphic Equalizer control window and the stored slider settings will be restored*. 42.
Figure 3-14 Tutorial Spectrum Analyzer Window 44. Try adjusting the Number of Averages control using the + and - buttons in the Averager block. Note that as the Number of Averages parameter is increased, the spectrum traces react more slowly and smoothly to the input signal. Return the Number of Averages setting to 8. 45. Click on the Run button in the Averager block until GREEN LED is no illuminated. You should see the spectrum waveform stop updating.
47. Click the Find Peak button to automatically move the marker to the largest magnitude displayed. Adjust the Gain controls for both Trace 1 and Trace 2 and note how the indicated magnitude for each trace is increased as gain is increased. 48. Set the Gain controls for the Yellow Trace to 40dB and the Blue Trace to 0dB.
55. Access the Save Setup File feature by clicking on File in the menu bar. The pulldown menu in Figure 3-15 will appear. Figure 3-15 Tutorial File Pulldown Menu 56. Click on Save Setup File to bring up the Save Setup File window as shown in Figure 3-16. For now, do not change the Look In or Folder Selection settings; just keep in mind that you could change these settings to allow setup files to be stored to any directory of any drive.
Figure 3-17 Tutorial Store in File Name Text Box 59. Click on OK to store your first setup file and return to the Master Control Panel. 60. Click on File to access the File Pulldown Menu as in step 56. Click on Open Setup File to bring up the Open Setup File window similar to Figure 3-3, but with mysetup.dac added to the files list. For now, do not change the path settings; just keep in mind that you could change these settings to allow setup files to be recalled from any folder of any drive. 61.
65. Open your original filter settings by repeating steps 60 and 61 to bring up the Open Setup File window. Next, click on mysetup.dac in the list box, then click on OK to return to the Master Control Panel. Figure 3-18 Tutorial Report Generator Pulldown Menu This completes the PCAP II Plus Tutorial. Please feel free at this point to experiment with the onscreen filter settings. Remember you can always recall filter settings that will likely work using the Open Setup File feature.
3.3: TRAINING CD RECORDINGS The PCAP II Plus Training CD supplied with the PCAP II Plus is a useful training tool. Each of its 13 noise sample recordings is accompanied by a several suggested setup files (“.dac” files), installed on your computer along with the other software by the installation procedure. Table 3.1 summarizes the contents of the training CD. The SETUP column contains the name of the setup files installed on the disk as part of the software installation procedure (Section 2.2: ).
Track Time Noise Description 1 02:30 500 Hz tone 2 02:20 3 01:11 Swept tones 4 02:21 High frequency hiss 5 00:44 6 01:14 7 02:40 8 02:24 9 02:34 10 01:11 11 02:26 12 02:10 13 04:52 Setup Comments Training CD Track 1 Solution A.dac Adjust Notch Freq to null tone Training CD Track 1 Solution B.dac 400 Hz and 1100 Hz Training CD Track 2 Solution A.dac tones Training CD Track 2 Solution B.
4: PCAP II PLUS SOFTWARE REFERENCE MANUAL This portion of the user's manual is designed as a reference guide to which the user may refer for more detailed information regarding the PCAP II Plus Master Control program graphical user interface (GUI). It is assumed in this section that the user has a good working knowledge of Microsoft Windows, has properly installed the PCAP II Plus hardware and software using the installation instructions in Section 2.2: , and has either completed the Tutorial in section 3.
4.1: MASTER CONTROL PANEL When the PCAP II Plus Master Control program is run from Microsoft Windows, the Master Control Panel appears. The Master Control Panel features are shown in Figure 4-1. From this screen, all audio processing capabilities can be accessed. The Master Control Panel is organized logically from left to right, with audio input controls at the far left, digital signal processing controls in the middle, and audio output controls at the far right.
simultaneously with as many as nine stages of digital signal processing; each stage is identical for both the Left and Right channels. Stereo, Independent is the same as Stereo, Linked, except that the processing stages may be setup and controlled independently for the Left and Right channels. General Filter Stages Box Selects the number of General Filter Stages that are to be used. Between 1 and 6 of these stages are available, depending upon the specified System Bandwidth and Input Mode.
Adapt Enabled Button Toggle control used either to freeze all adaptive filters (LED not illuminated) or to allow non-frozen filters (as specified by the Adapt button for each Filter) to adapt (LED illuminated GREEN). Clear All Button Clears the filter coefficient solutions of any adaptive filters selected. If no adaptive filters are present, then the Clear All button has no effect.
4.2: INPUT AND OUTPUT PROCESSORS 4.2.1: Input Level Measurement/Adjustment Application: It is critical that the input signals coming into the PCAP II Plus be adjusted in level such that the Input Levels bargraphs on the Master Control screen are maximized in intensity without indicating an overload condition (display fully pegged in the RED zone). Therefore, the Input Levels controls are provided.
illustrated in Figure 4-3; interpretation of the external processor front panel “tricolor” LEDs is illustrated in Figure 4-4.
4.2.2: Digital Input Channel Swapping Application: Particularly on occasions where a digital audio tape recorder (DAT) is used to record “Primary” (e.g., microphone) audio and “Reference” (e.g., TV) audio simultaneously onto the same tape, it is possible that the channels may be swapped; e.g., the Primary audio may be recorded onto the Right audio channel and the Reference audio recorded on the Left audio channel.
Figure 4-6: Channel Swap Feature Indication 4.2.3: Input and Output Highpass Filters (HPFs) Application: The Input HPFs are used to remove rumble or other low-frequency noises which occur below 100 Hz to 500 Hz (adjustable) from the input signals before they enter the digital processors. Since very little speech information is lost in bandlimiting below these frequencies, this filter is recommended for voice enhancement.
For fast access* to the Input and Output HPFs, pressing or may also be used to toggle the Input or Output HPFs, respectively, in or out of the process. Adjustment of the Input HPF cutoff frequency is accomplished by clicking on the Input HPF Config button. When this button is pressed, the window shown in Figure 4-8 appears. Figure 4-8 Input HPF Config Window Specify the cutoff frequency in 10 Hz increments using the slider control.
4.2.4: Digitally-Controlled Limiter Application: The dual-channel input Limiter automatically protects the input circuits from overload distortion by reducing input signal levels whenever loud sounds, such as door slams, exceed a specified Threshold. When the overload goes away, the Limiter returns the input signal levels to their original settings over the specified Release Time interval. If no loud sounds exceed the specified Threshold, the Limiter will not affect the input signals.
Figure 4-10 Limiter Config Window Threshold specifies the input bargraph level which is considered to be an overload condition; any time an input signal exceeds this level, the gain will be decreased for that input until it no longer exceeds the Threshold. The Threshold options are -6dB (highest), -9dB, -12dB, and -15dB (lowest). The -9dB setting is recommended for most applications.
4.2.5: Digitally-Controlled AGC Application: The dual output 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.
Figure 4-12 AGC Config Window Maximum Gain specifies how much gain the AGC can apply in its attempt to bring the output signal up to the proper level. The greater the Maximum Gain, the lower the output signal level that can be brought up to proper level. The Maximum Gain options are 5dB, 10dB, 15dB, 20dB, 25dB, 30dB, and 35dB. For most near party/far pary applications, the 10dB setting is recommended. Maximum Gain settings greater than 10dB may elevate background noise during pauses in speech.
4.3: 4.3.1: DIGITAL PROCESSING SELECTION General Filter Stages Filter Selection To select the type of filtering to be applied by any of the General Filter Stages, use the filter select combo box (or “drop-down menu”) shown in Figure 4-13. To select the filter, first click on the small arrow to the right of the combo box and then, using the menu that appears, click on the desired filter.
4.3.3: Equalizer Stage Processing Selection The Equalizer Stage follows the Broadband Filter Stage in the processing chain. To select the type of processing to be applied by the Equalizer Stage, use the filter select combo box (or “drop-down menu”) shown in Figure 4-13. To make a selection, first click on the small arrow to the right of the combo box and then, using the menu that appears, click on the desired selection.
4.4: GENERAL FILTER CONTROL WINDOWS This section provides detailed description of the control window for each type of processing that the PCAP II Plus implements in the General Filter, Broadband Filter, and Equalizer processing stages. For any Filter or Equalizer block, the control window for the selected filter is accessed by clicking on the Config button. 4.4.
Description of controls (Figure 4-15) is as follows: Conditional Adaptiation: For advanced users only. Novice users should keep Filter will adapt set to Always. The Threshold setting has no effect in this case. Conditional Adaptation allows the adaptive filter to automatically Adapt/Freeze based upon Master Control Panel bargraph levels. This can be very useful in situations where there are pauses, or breaks, in the speech being processed.
very slow adaptation, while a Mu of 3808 x 2-14 provides fastest adaptation.* As a rule set this rate to approximately 100-200 initially, to establish convergence, then back off to a mid value to maintain cancellation. Auto Normalize: Selects Normalized (LED illuminated GREEN) or Fixed (LED not illuminated) adaptation rate. Normalized is recommended. When Auto Normalize is enabled, the specified Adapt Rate is continuously power scaled based upon the input signal level.
affecting any other adaptive filters in the process. Store Button: Used to store adaptive filter settings and coefficients. This button is disabled in Stereo-Linked mode. See section 4.4.1.1: for more information. Recall Button: Used to load previously stored adaptive fitler settings and coefficients. This button is disabled in Stereo-Linked mode. See section 4.4.1.2: for more information.
4.4.1.1: Storing Adaptive Filter Settings and Coefficients This section relates to both the Reference Canceller and the 1-Channel Adaptive Filter. Each of these filters allows the user to store the filter settings of the adaptive filter as well as the coefficients stored within the filter. When clicking the Store button a window will appear (Figure 4-16) displaying the current working folder and showing all the previously stored adaptive filter settings files.
current working folder and showing all the previously stored adaptive filter settings files. If there have been no previous settings stored then no other files will be listed. These files have the extension “*.cof”. Coefficients stored from a 1 Channel filter can be loaded by a Reference Canceller and vice versa, although this is not recommended. Figure 4-17 Adaptive Filter Recall Settings Window Before the OK button will be enabled a valid settings file name must be entered.
4.4.2: 1-Channel Adaptive Filter Application: 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.
Click on the Clear button if you desire the filter to completely readapt based upon the new Conditional Adaptation settings. The Adapting LED indicator shows the current adaptation status of the filter; if the LED is illuminated GREEN, the filter is adapting, and when the LED is not illuminated the filter is not adapting. In Stereo, Linked Input Mode, only the Left channel adaptation status is displayed.or GRAY when the filter is frozen.
Auto Normalize: Selects Normalized (LED illuminated GREEN) or Fixed (LED not illuminated) adaptation rate. Normalized is recommended. When Auto Normalize is enabled, the specified Adapt Rate is continuously power scaled based upon the input signal level. This generally results in faster convergence for a given Mu. When Auto Normalize is not enabled, the specified Adapt Rate is utilized at all times without power scaling Prediction Span: Sets the number of samples in the prediction span delay line.
4.4.3: Lowpass Filter Application: 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.
Description of controls is as follows: Cutoff Frequency: Specifies frequency in Hertz above which all signals are attenuated. Frequencies below this cutoff are unaffected. Minimum Cutoff Frequency is 100 Hz, while the maximum Cutoff Frequency depends upon the System Bandwidth setting. Cutoff Frequency can be adjusted in 1 Hz steps. Stopband Attenuation: Specifies amount in dB by which frequencies above the Cutoff Frequency are ultimately attenuated.
4.4.4: Highpass Filter Application: 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.
Transition Slope: Specifies slope at which frequencies below the 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 System Bandwidth settings. Filter Active Button: Used to switch the filter in and out of the process without affecting the operation of the other filters.
4.4.5: Bandpass Filter Application: 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.
Description of controls is as follows: 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 Cutoff Frequency is 0 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.
OK Button: Use either this button or the “X” button at the upper right corner of the window to exit the dialog and return to the Master Control panel. A graphical description of the Bandpass filter and its controls follows in Figure 4-24.
4.4.6: Bandstop Filter Application: 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.
Figure 4-25 Bandstop Filter Control Window 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 Upper Cutoff Frequency is 10 Hz above the Lower Cutoff Frequency, while the maximum Upper Cutoff Frequency depends upon the System Bandwidth setting. Upper Cutoff Frequency can be adjusted in 1 Hz steps.
NOTE: The Filter Active button is equivalent to, and linked with, the Active button in the associated Filter block on the Master Control Panel. OK Button: Use either this button or the “X” button at the upper right corner of the window to exit the dialog and return to the Master Control panel. A graphical description of the Bandstop filter and its controls follows in Figure 4-26.
4.4.7: Comb Filter Application: 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").
Description of controls is as follows: Comb Frequency: Specifies fundamental frequency in Hertz of comb filter. generated at multiples, or harmonics, of this frequency. Notches are NOTE: Comb Frequency changes whenever the System Bandwidth setting is altered; if you change the System Bandwidth setting, you will need to readjust the Comb Frequency for any Comb Filters selected. Notch Limit: Specifies frequency in Hertz above which no notches are generated.
This procedure minimizes the filtering to only that needed for the hum. Since a comb filter is a reverberator, a 1-Channel Adaptive Filter is often placed after it to reduce the reverberation and clean up any residual noises escaping the comb filter. A graphical description of the Comb filter and its controls follows in Figure 4-28.
4.4.8: Notch Filter Application: 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 source varies, then the 1-Channel Adaptive filter is recommended.
depends upon the System Bandwidth setting. Notch Frequency is adjustable in 1 Hz steps. Notch Depth: Depth of the notch that is generated. Notch Depth is adjustable from 10 dB to 60 dB in 1 dB steps. Notch Width: Width of the generated notch in Hertz. NOTE: Notch Width varies with the System Bandwidth setting. Filter Active Button: Used to switch the filter in and out of the process without affecting the operation of the other filters.
4.4.9: Multiple Notch Filter Application: The Multiple Notch filter is used to remove, or "notch out", up to 16 single-frequency noises, such as tones or whistles, from the input audio with minimal effect to the remaining audio. This is accomplished using a frequency-sampling-synthesized 1024-tap FIR filter which is calculated in the PC by the PCAP II Plus Master Control program.
Figure 4-31 Multiple Notch Filter Control Window Description of controls is as follows: Notch Freq (1-16): Specifies frequency in Hertz which is to be removed from the input audio by each of the 16 notches. Minimum Notch Freq is 1 Hz, while maximum Notch Freq depends upon the System Bandwidth setting. Set Notch Freq to Out (scroll box in full left position) if the notch is not desired. Notch Freq is adjustable in 1 Hz steps. Notch Width (1-16): Specifies width in Hertz for each of the 16 notches.
Depth Control: Used to specify the dB attenuation to be applied for all notches currently implemented. However, unlike the Notch Freq and Width controls, the Depth control is fully interactive and has instantaneous effect at all times; clicking the Build button is not necessary for the Depth control to have effect. Range of adjustment is 0 to -90.3dB.
4.4.10: Slot Filter Application: 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. 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.
Description of controls is as follows: 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 System Bandwidth setting. Slot Frequency is adjustable in 1 Hz steps. Stopband Attenuation: Specifies amount in dB by which frequencies other than the Slot Frequency are attenuated. Stopband attenuation is adjustable from 10dB to 60dB in 1 dB steps. Slot Width: Width of the generated slot in Hertz.
4.4.11: Multiple Slot Filter Application: 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, e.g. telephone “DTMF” tones. The Multiple Slot filter is used to isolate, or "slot", up to 16 single-frequency signals, such as tones or whistles, from the input audio, eliminating all other audio.
Figure 4-35. Multiple Slot Filter Control Window Description of controls is as follows: Slot Freq (1-16): Specifies frequency in Hertz which is to be isolated from the input audio by each of the 16 slots. Minimum Slot Freq is 1 Hz, while maximum Slot Freq depends upon the System Bandwidth setting. Set Slot Freq to Out (scroll box in full left position) if the slot is not desired. Slot Freq is adjustable in 1 Hz steps. Slot Width (1-16): Specifies width in Hertz for each of the 16 slots.
button after any change in System Bandwidth or Input Mode in order to rebuild the filter. Stopband Control: Used to specify the dB attenuation to be applied in between the slots currently implemented. However, unlike the Slot Freq and Width controls, the Stopband control is fully interactive and has instantaneous effect at all times; clicking the Build button is not necessary for the Stopband control to have effect. Range of adjustment is 0 to -90.3dB.
4.4.12: Spectral Inverse Filter Application: The Spectral Inverse Filter (SIF) is an equalization filter which automatically readjusts the spectrum to reduce noise and muffling effects. It is especially useful when the voice has been exposed to reverberations and bandlimited noises. For an automatic implementation of the SIF, use the ASIF (see Section 4.5.
The equalization effect of SIF is very beneficial with reverberant audio and recordings exposed to substantial recorder wow and flutter. The noise sources must remain stationary for SIF to be effective. SIF cannot readjust itself to changing noises, such as music. In such cases, the 1-Channel adaptive filter is recommended. A second SIF equalization mode is Attack Noise. This mode is especially useful in reducing band limited noises such as horns and mechanically induced noises.
Analyzer Block: Filter Operation Block: Used to control the spectrum analyzer which acquires the original audio power spectrum; this spectrum is displayed and continuously updated in the Filter Display area as a yellow trace. Analyzer controls include: • Clear button which is used to zero the averager memory and cause the averaged spectrum to be recalculated anew. • Run button which allows the user to start (GREEN LED indication) or stop (LED unlit) update of the averaged spectrum.
Figure 4-39 SIF Control Window When Attack Noise Selected Filter Amount Block Specifies Filter Amount and Output Gain. Equalize Voice or Attack Noise Filter Amount specifies the maximum amount of volume reduction that can be applied by the inverse filter within the specified frequency limits; this may be set to the approximate difference in amplitude between the largest and smallest input spectral components within the frequency limits. This value varies between 0 and 100%.
Equalize Voice or Attack Noise Output Gain specifies the digital boost to be applied to the entire spectral inverse filter curve. Normally, Output Gain is applied in the Equalize Voice mode; the gain is usually 0 dB in the Attack Noise mode. This boost is necessary to make up for the volume reduction performed by the inverse filter. Output Gain should be initially set to approximately 0 dB.
Figure 4-40 SIF Custom Curve Window Build Button: Builds the spectral inverse filter based on the original input audio spectrum and the SIF control settings. When clicked, the mouse cursor will change to an "hourglass" shape, indicating that the PC is busy calculating the spectral inverse filter coefficients and sending them to the external processor.
selected file from disk. These stored settings are bandwidth independent, although they will will only be relevant for the bandwidth in which they were originally created Figure 4-41 SIF Store Settings Window Recall Button: This button allows the user to recall a previously stored spectral inverse filter curve from a file, Clicking this button brings up the Recall Spectral Inverse Filter window (Figure 4-42) referencing the current working folder.
Figure 4-42 SIF Recall Settings Window Examples of Spectral Inverse Filters: In the examples below, SIF will equalize a voice spectrum using various Equalize Voice Filter Amounts and Output Shapes. When the Filter Amount is small, only the peaks in the spectrum are flattened. As the Filter Amount is increased, lower energy segments are equalized. The top trace in each of the figures below gives the filter curve and the bottom trace gives the original input spectrum.
Figure 4-44 SIF with Filter Amount set to 85% Figure 4-45 SIF with Filter Amount set to 100%, Output Gain set to 3.
The Output spectral Shape may be selected as Flat, illustrated in Figure 4-43, Figure 4-44, and Figure 4-45 above. It may also be set to Voice or Pink. See Figure 4-47. The Attack Noise mode does not attenuate Out-of-Limits signal, but equalizes and attenuates inLimits signal frequencies. Figure 4-46 illustrates. Figure 4-46 Attack Noise operation, Filter Amount set to 100% Applications Suggestions The following suggestions may be beneficial in setting up and operating the Spectral Inverse filter.
The Analyzer Gain should be increased to display weaker energy components. Do not overload (OVL) the analyzer, as the spectral information would become corrupted. Try to capture a representative spectrum using the Run button. Once a stable, representative spectrum is obtained on the display, freeze the analyzer (GREEN Run LED unlit). This same spectrum may be used for several different variations of the filter (changing Limits or Filter Amount, as an example).
Figure 4-47 SIF with Output Shape set to Voice Experiment: Select a section of audio and display its spectrum. Freeze the analyzer and vary different control settings, storing built filters. Compare the results and determine the best solution. Always compare these different filter solutions using the same input audio.
4.4.13: Hi-Res Graphic Filter Application: 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 example, a microphone acoustically modified as a result of concealment), a compensation filter that reshapes the audio to a normal spectral shape might be desirable.
Description of controls/indicators is as follows: Filter Display: Graphically displays the current shape of the filter. Also used in conjunction with the mouse to draw a new filter shape or to edit an existing one (see New, Edit, and Normalize button descriptions). A grid is provided to assist the user in visually judging frequency and attenuation at any point in the display. Freq and Atten Used to precisely readout the frequency in Hertz and attenuation in dB Readouts: at any point in the filter curve.
Figure 4-49 Hi-Res Filter Store Window Figure 4-50 Hi-Res Graphic Filter Recall Window 4.4.13.1: Hi-Res Graphic Mini-Tutorial The New, Edit, and Normalize buttons are used to graphically manipulate the shape of the filter curve. Their functions are complex, and thus are best illustrated in the following mini-tutorial: 1. Master Control Panel, set System Bandwidth to 5.4 kHz, Input Mode to Mono, and Number of General Filter Stages to 1.
2. Click on the Config button to bring up the Hi-Res Graphic Filter control window. When used for the first time, the control window will be the that of the previous Figure 4-48. 3. Click on the New button to draw a new filter. The screen will now appear as follows: Figure 4-51 New Hi-Res Graphic Filter Display Had you accidentally clicked the New button, you could click on Abort to restore the previous filter. 4.
Figure 4-52 Hi-Res Graphic Draw in Progress 5. Complete drawing the filter curve as shown below (Figure 4-53) by drawing points all the way to the right edge of the filter display area. Figure 4-53 Completed Hi-Res Graphic Draw When you have drawn the last point (must be at or beyond the right edge of the filter display area), the mouse cursor will change to an "hourglass" shape for a few seconds while the filter is being calculated.
Figure 4-54 Hi-Res Graphic Edit Window In this window, you can make the entire filter curve drop by a specified amount prior to editing the curve. This can be used to create headroom which can be used to increase the gain (decrease the attenuation) in one portion of the curve relative to the rest of the curve. For now, select a drop of 0dB (No Drop) and click on Proceed. 7.
Figure 4-56 Hi-Res Edit In Progress 9. Complete drawing the new portion of the filter curve as shown below (Figure 4-57) by drawing points all the way to the right edge of the edit region: Figure 4-57 Completed Hi-Res Graphic Edit When you have drawn the last point (must be at or beyond the right edge of the edit region), the mouse cursor will change to an "hourglass" shape for a few seconds while the filter is being recalculated.
Figure 4-58 Normalized Hi-Res Graphic Filter This completes the Hi-Res Graphic Filter mini-tutorial.
4.4.14: 20-Band Graphic Filter The 20-band Graphic Filter 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 slider controls found on analog graphic equalizers found on many consumer stereo systems, and thus should be very familiar to even the novice user.
Link Frequency Sliders Checkbox: When checked, this feature links the 20 slider controls together such that when any one of them is moved, all others move together in sync. Uncheck the box to move the sliders without affecting the others. Center Frequency: Note that the Center Frequency of each band is labelled underneath each slider, and that bands are more closely spaced at low frequencies Gain Indication: Above each slider control, the gain for that frequency band is given.
Filter Active Button: Used to switch the filter in and out of the process without affecting the operation of the other filters. Button LED is illuminated RED when the filter is in the process, and is not illuminated (GRAY) when the filteris out of the process. Note: The Filter Active button is equivalent to, and linked with, the Active button in the associated Filter block on the Master Control Panel.
4.4.15: Tri Parametric Filter Application: The Tri Parametric Filter consists of three adjustable IIR filter “substages”, connected in series, which can be used for both peaking and nulling portions of the input signal’s frequency spectrum. These substages can be configured independently, and are typically used as precision notch filters, which perform nulling of the input signal at a specified Center Freq over a specified width.
Description of operation: The Center Freq text box / slider controls allow the user to either type in the desired Center Frequency of each stage directly, or to make mouse adjustments. Center Frequency can be specified in 1 Hz steps. The Width text box / slider controls work similarly to the Center Freq controls. The displayed Width, however, is proportional to the Center Frequency, and will thus be automatically changed whenever the Center Freq controls are adjusted.
Figure 4-61 Graphical Representation of a Parametric Filter 103
4.4.16: Inverse Comb Application: The Inverse Comb filter is used to indentify 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 slots. This filter is the exact opposite of the Comb Filter. Adjust the Inverse Comb Frequency to the desired spacing between slots (also known as "fundamental frequency").
Harmonic Limit: Specifies frequency in Hertz above which no slots are generated and the signal is attenuated. Minimum Harmonic Limit is 100 Hz, while maximum Harmonic Limit depends upon the System Bandwidth setting. Harmonic Limit is adjustable in 50 Hz steps. Attenuation: Depth of the attenuation region around the slots that is generated. Attenuation is adjustable from 10 dB to 60 dB in 1 dB steps.
4.4.17: Limiter/Compressor/Expander Application: The LCE (Limiter/Compressor/Expander) is a three-section dynamic signal level processor, recommended for advanced users only. Dynamic signal level processing enables the user to manipulate the overall dynamic range of a signal, generally to correct for near-party/farparty and/or “quiet talker” scenarios. The three types of level processing available (three sections) are limiting, compression, and expansion.
Compression Region: In this region the output signal level changes at a fraction of the rate of the increase of the input signal level. The dynamic range of the output signal is thus is reduced with respect to the input signal. 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 allows signals of wide dynamic ranges to be squeezed into more limited dynamic ranges of recording media and transmission channels.
The Compression Threshold can be adjusted through a range of –90dB to 0dB by using the scroll buttons beside its text entry box, by entering the value in its text entry box, or by clicking on the Compression Threshold line using the mouse pointer and adjusting it to the desired value.
4.5: BROADBAND FILTER CONTROL WINDOWS 4.5.1: Noise Reducer Application: The Noise Reducer is a frequency-domain spectral-subtraction filter that implements automatic noise reduction over 512 separate frequency bands. 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).
Figure 4-64: Noise Reducer Control Window Description of controls/indicators is as follows: Master Attenuation Control: Used to specify the amount of noise reduction that the spectral subtraction attempts to apply to the input signal. Adjustment range is 0 (no attenuation) to 100% (maximal attenuation) in 1% increments.
4.5.2: NoiseEQ™ Application: Like the Noise Reducer tool, the NoiseEQ™ is a frequency-domain spectral-subtraction filter that implements automatic noise reduction over 512 separate frequency bands. 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).
Figure 4-65: NoiseEQ™ Control Window Description of controls/indicators is as follows: Frequency Specific Noise Reduction: Used to specify the amount of noise reduction that the spectral subtraction attempts to apply to the input signal within each of 20 separate groups of frequency bands. Within each band, adjustment range is 0 (no attenuation) to 100% (maximal attenuation) in 1% increments.
Filter Active Button: Used to switch the filter in and out of the process without affecting the operation of the other filters. Button LED is illuminated RED when the filter is in the process, and is not illuminated (GRAY) when the filter is out of the process. Note: The Filter Active button is equivalent to, and linked with, the Active button in the associated Filter block on the Master Control Panel.
4.5.3: Adaptive Spectral Inverse Filter (ASIF) Application: The Adaptive Spectral Inverse Filter (ASIF) is an equalization filter that automatically readjusts 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. This filter is much like the Spectral Inverse Filter (SIF, See Section 4.4.
Figure 4-66: ASIF control window Description of controls/indicators is as follows: Display Trace and Display Controls: The display trace can be used to view either the filter input and output audio or the ASIF filter response. The radio buttons in the Display Trace block allow the user to select which type of trace is shown. When viewing audio on the display, the user can choose to view the Input Audio only, the Output Audio only, or both input and output audio.
dynamic range from 0 to -70 dB. When viewing the filter response on the display, the user can choose the dynamic range of the trace. This selection is made using the drop-down menu in the Filter Trace block. Because the filter can both attenuate and boost frequencies, the trace grid does not have hard range limits. Instead, the trace limits are automatically adjusted to “good” settings based on the user-specified range and the filter response values.
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.
• Custom – user draws custom curve to be applied 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.
output level. Filter Amount This setting controls the degree to which the ASIF 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 result. 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.
Voice Limits: frequency range, or “ASIF region,” over which the ASIF is applied. Two red markers, controlled by the sliders below the display trace, indicate where the lower and upper voice limits are located. The markers may also be adjusted by clicking and dragging within the display trace. Viewing audio on the display trace while manipulating the markers is an easy way to identify where your ASIF region limits should fall.
To begin adapting from the previously adapted filter state (i.e. if the current input audio is similar to the store-time input), simply click the Adapt button to enable filter adaptation. To use the saved settings but re-start filter adaptation from an initialized state (i.e. if the current input audio is different from the store-time input), click Clear to clear the filter, then click Adapt to enable filter adaptation.
Examples: 1. This example shows a typical use of the ASIF. The target audio is speech, but the speech is severely muffled. The muffling can be visualized on the audio trace, where the yellow input trace shows a sharp drop-off for frequencies above approximately 600 Hz. The ASIF boosts these muffled frequencies. The blue output trace shows that the muffled frequencies have been boosted so that the overall signal spectrum looks more like the selected pink output shape.
Figure 4-69: Filter trace view for ASIF Example 1 2. The Filter Amount controls the degree of filtering that is applied to the signal. In Example 1, the Filter Amount was set to 100%, or full filtering. The figures below show the same filter at other filter amount settings. Notice that, as the filter amount is reduced, the filter shape converges to the selection output shape (with gain applied).
Figure 4-70: Filter response from Example 1 with Filter Amount 85% Figure 4-71: Filter response from Example 1 with Filter Amount 0% 124
4.6: EQUALIZER CONTROL WINDOWS This section provides detailed description of the control window for each equalizer mode. For any digital Equalizer block, the control window for the selected equalizer is accessed by clicking on the Control button. The output equalizer is used to reshape the noise-reduced digitally filtered signal. In the process of enhancement, high frequencies might be boosted (this is common with both the 1-Channel adaptive and Spectral Inverse filters).
Figure 4-72 20-Band Graphic Equalizer Control Window Description of controls/indicators is as follows: 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 0dB and -40 dB in 1dB steps. Link Frequency Sliders Checkbox: When checked, this feature links the 20 slider controls together such that when any one of them is moved, all others move together in sync.
All Down 1dB Button: This button shifts all sliders down by 1dB from their current position; no slider, however, will be allowed to go below -40dB. 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. Store Button: This button allows the user to store a slider configuration to a userspecified disk file that will not be lost when the computer is turned off.
4.6.2: Spectral Graphic Equalizer Application: For many output equalization requirements, the 20-Band Graphic Equalizer described in Section 4.6.1: should be adequate. Its resolution is limited, having only 20 coarsely-spaced slider controls; some applications require finer control of the output spectrum shape. For this reason, the 115-band Spectral Graphic Equalizer is provided (a 460-band version is also available as the Hi-Res Graphic Filter described in Section 4.4.13: ).
4.6.3: Dual Parametric Equalizer The Dual Parametric Equalizer is identical to the Tri-Parametric Filter described in Section 4.4.14: , except that only two substages are available instead of three. Please refer to that section for operating instructions.
4.6.4: Limiter/Compressor/Expander The Equalizer Stage Limiter/Compressor/Expander is identical to that found in the General Filter Stages. Please refer to Section 4.4.17: for more details on this tool.
4.7: STORING AND RECALLING INDIVIDUAL FILTER SETTINGS Several of the filters and equalizers provide the ability to store their individual settings to a disk file. These settings can be recalled for later use with other configurations. These individual filter settings files are in addition to the setup files that store the entire configuration. Filters or equalizers that provide these features will have a Store or Recall button on their control windows.
Figure 4-74 Store Individual Settings Window for the SIF 4.7.2: Recalling Individual Settings Files Upon clicking the Recall button from the respective filter control window the Recall Filter window will appear. This window will be intialized to open in the current working folder (See Section 6.3: for more information about the current working folder). The appropriate file extension will be placed in the File Name entry field.
4.8: VISUALIZATION TOOLS 4.8.1: Spectrum Analyzer and Coefficient Display Buttons Figure 4-76 Spectrum Analyzer and Coefficient Display Buttons Clicking the Spectrum Analyzer button until the LED is illuminated BLUE brings up the Spectrum Analyzer window; clicking until the LED is not illuminated closes the window. For fast access to the Spectrum Analyzer window, press on the keyboard. See Section 4.8.2: for complete instructions on operating the controls in the Spectrum Analyzer window.
4.8.2: Spectrum Analyzer Window Application: To properly utilize the processing tools, it is often necessary to measure the frequency characteristics of the input signal. This assists in determining the type of filtering needed. Also, after processing the signal, it may be desirable to compare the frequency characteristics of each digital filter output to those of the input signal, thus determining the effectiveness of each digital filter.
Figure 4-77 Spectrum Analyzer Window Description of controls/indicators is as follows: Averager Block: Controls averaging of successive FFT spectral traces. Number of Averages selects the number of averages to be applied to spectral traces in the Exponential and Peak Hold averaging modes (explained below). The more averages applied, the smoother the displayed spectrum waveforms will be; however, the waveforms will also update more slowly as Number of Averages is increased.
(spectrum only builds up using specified Number of Averages, can never come down, thus capturing any strong peaks that might occur). Marker Block: Used to turn the vertical red marker in the Spectrum Display area on and off via the Marker button; GREEN LED indicates the marker is on, LED OFF indicates the marker is off. The marker allows frequency (Freq) and magnitude (Mag) readout of any point in the spectra.
A vertical red marker is used to read out the exact magnitude(s) (Mag) of any frequency (Freq) in the spectrum display. To manage the marker, simply click the mouse cursor on the desired point in the Spectrum Display area, or utilize the controls in the Marker Block, described above. Hide Controls Checkbox: Unchecking this box allows the Spectrum Display to utilize the full window space; recheck the box if access to the controls is required.
4.9: COEFFICIENT DISPLAY WINDOW Application: Particularly when setting up the Ref Canceller filter, it is often useful to display the impulse response (filter coefficients) of the filter. Additionally, it is sometimes desirable to know the precise time-domain response of any of the General Filter stages. For these reasons, the Coefficient Display window has been provided. The Filter stage to be displayed is specified in the Filter combo box within the Display block by clicking on the desired Filter.
Figure 4-78 Coefficient Display Window Description of controls is as follows: Display Block: Used to select the General Filter stage whose coefficients are to be displayed via the Filter combo box; available options depend upon the System Bandwidth and Input Mode settings. Also, the Number of Coefficients to be displayed can be specified; options are power-of-two steps from 32 up to the Filter Size as specified in the control window for the filter.
entering the desired coefficient number in the text box moves the marker instantly to the desired point. Zoom Combo: Used to specify the vertical zoom factor to be used when displaying coefficients. Coefficients may be zoomed from 1X (no scaling) to a maximum of 200X. If the scaled coefficient exceeds the maximum vertical display limits, it will be clipped prior to display. NOTE: This clipping does not affect signal processing.
4.10: MASTER CONTROL PULLDOWN WINDOWS 4.10.1: Saving Setups to Disk Files Application: To save time configuring detailed control settings, complete setups may easily be saved to disk setup files for future reuse. These files are particularly handy when making presentations which require multiple setups, or when it is desired to precisely duplicate the enhancement procedure at some point in the future.
Figure 4-79 Save Setup File Window 4. You will need to specify a filename for the setup. Click on the File Name text box, then type the desired filename (up to 256 characters). All setup filenames must have the .DAC extension; thus, the .DAC extension is automatically included in the text box. If you delete the .DAC extension, an error message will be generated. 5.
4.10.2: Recalling Setups from Disk Files Application: To save time configuring detailed control settings, complete setups may easily be saved to disk setup files for future reuse. These files are particularly handy when making presentations which require multiple setups, or when it is desired to precisely duplicate the enhancement procedure at some point in the future. Also, this feature allows easy transfer of enhancement setups between PCAP II Plus systems simply by exchanging the “.DAC” setup files.
wrong setup file is selected. To browse a Description, scroll to and click on the desired file listed in the Setup Files list box. 4. Once the desired setup file name has been found, either double-click it or click on OK to open it. A message will alert the user that 60 seconds may be required to completely open the setup. An "hourglass" mouse cursor will now appear, indicating that the PCAP II Plus is busy configuring itself with the recalled settings from disk.
4.10.3: Using Calibration Mode Application: The PCAP II Plus calibration mode allows verification of the PCAP II Plus functionality. When enabled the PCAP II Plus will output on both digital and analog outputs the signal type selected in the Signal Generator Output area. One of a number of tests can be selected and run via the Run Test button. To download a complete calibration and test procedure in “.pdf” format, please visit the DAC website at www.dacaudio.com.
4.10.4: Storing Setups to External Processor Stand-Alone Memories Application: The PCAP II Plus external processor has the capability to store up to ten enhancement setups internally in nonvolatile Stand-Alone memories, allowing the external processor to be operated in those setups without needing to be connected to a PC.
3. Click on the selected memory. The following window (Figure 4-84) will now appear: Figure 4-84 Auxiliary Switch Function Window This window allows the function of the AUX switch on the external processor front panel to be programmed. Click on the program option desired, then click on OK. 4. A message window will now appear to alert you that the previous contents of the selected memory will be lost, and that 30 seconds could be required to transfer all settings to the memory.
4.10.5: Generating Setup Reports The Master Control program has a Report-Generator feature which allows hardcopy printouts of all PCAP II Plus control settings to be generated. This is useful when the enhancement procedure needs to be documented. To access this feature, click on Report-Generator in the Master Control menu bar.
Figure 4-86 Help Pulldown Menu Click on Contents to access the help Contents window. The Contents window will display all the subjects for which help is available for the PCAP II Plus Master Control program. You may browse through the displayed subjects and select help for a particular subject by double-clicking the desired subject. The help utility is, in fact, a separate Windows program from the Master Control software.
5: . OPERATING PCAP II STAND-ALONE Before the PCAP II Plus external processor can be operated Stand-Alone, at least one of the StandAlone setup memories must be programmed using the procedure in Section 4.10.1: . Operate the PCAP II external processor as a Stand-Alone audio processor as follows: 1. Connect 12VDC power, and switch the POWER switch ON. 2.
2. To bypass all digital filters, allowing the original unfiltered audio to pass through to the LEFT and RIGHT ANALOG OUTPUTS and S/PDIF digital OUTPUT RCAs, switch the PROCESS/BYPASS switch to BYPASS. To filter the audio, switch to PROCESS. 3. The AUX switch is programmed to have one of the following functions when switched to AUX: a. No effect at all (default) b. Input Limiter IN c. Input HPF IN d. Output AGC IN e. Output HPF IN f.
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6: . PCAP II PLUS CUSTOMIZATION OPTIONS 6.1: CONFIGURING PCAP II FOR DIFFERENT COM PORTS WARNING: Should you choose to share a single COM port connection on your computer with a PDA, e.g. a Palm Pilot, connecting each to the COM port only when needed, be sure to shut down any “Hot Sync” manager application that might be running before attempting to run the PCAP II Plus Master Control software. Otherwise communication with the PCAP II Plus will fail due to conflict with the Hot Sync manager.
4. Click on "File", then "Exit" (or click the “X” button at the upper right corner of the Master Control screen) to exit the Master Control program. This will save the new COM port setting.
6.2: CONFIGURING PCAP II PLUS FOR DIFFERENT BAUD RATES This section is intended for advanced users, only. The PCAP II Plus system is capable of operating at RS232 symbol rates of 9600, 14400, 19200, 38400, and 115200 baud. Each PCAP II Plus is factory-configured to operate at 115200 baud, which works well for most modern PCs. However, some applications (such as modem or network links) may require the use of a different baud rate. Change the PCAP II Plus baud rate as follows: 1.
c. On the Master Control menu bar, click on the "Settings" menu bar option. d. When the pulldown menu appears, click on “Connect Settings”, then "Baud Rate". The display will now appear as shown in Figure 6-2. e. Click on the baud rate setting which corresponds to the new external processor DIP switch settings. After clicking the desired baud rate again click on the "Settings" menu bar option and then click on "Connect to PCAP II Plus".
6.3: WORKING FOLDER Within the File menu on the Master Control Panel is the option for Edit Working Folder. This option provides a way of designating where individual filter settings as well as “.dac” files will be stored. Figure 6-3 Working Folder Menu Option Once you have clicked on the Edit Working Folder menu option the Default Working Folder Window will appear. The working folder selection area will default to begin set to the current working folder.
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7: DAC1024T AND DAC4096T ADAPT RATE SETTINGS The DAC1024T and DAC4096T both had 12 position Adapt Rate Switches on their front panels. The lower the number the switch setting was turned to the faster the filter coefficients adapted. For those users who are used to using those settings the table listed below shows how the switch settings correspond to the (ADAPT RATE) x 2-14 value shown on the filter control screen.
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8: PCAP II SPECIFICATIONS Analog • Two rear panel RCAs: left for mono operation, left and right for stereo operation, and for primary and reference for 2CH adaptive filter operation. • Zin = 50kΩ, level adjustable -8 to +15 dBm. • Two rear panel RCAs: left and right for stereo operation, single output for mono and 2CH adaptive filter operation. • Zout = 100Ω, Vout = +8 dBm. • Panel stereo jack and volume control. Suitable for 8 ohm stereo headsets.
selected (if ANALOG input, output rate forced to 44.1kHz in AutoSYNC position) • Accommodates any valid S/PDIF digital input signal over a sample rate range of 25-108kHz • 6144-tap flexible adaptive/fixed filter module, dynamically allocable in 1024-tap sections to as many as 6 filter stages. • Two 256-tap FIR filters for output equalizers. • Four 256-tap FIR filters for input and output highpass filters. • Four 128-tap FIR filters for interpolation / decimation.
Construction Packaging • 1.5" H x 10.0" W x 9.75" D, 3 lbs. Rugged aluminum enclosure with black powder-coat finish. Power • 10 - 16 VDC @ 2.0A, maximum. • Universal AC adaptor supplied. • Front: phones jack, monitor select, and volume control; standalone mode select, filter clear, adapt/freeze, process/bypass, aux/off switches; two input level controls and tricolor level LEDs.
Output Equalizers • 20-band graphic, spectral graphic (115 lines resolution), and parametric equalizers Level Control • Microprocessor-controlled input limiter and output AGC Spectral Analysis • Dual-channel FFT with exponential averaging • 460 line resolution • 70 dB display and 110 dB scalable dynamic range 164
