Copyright: Copyright © 1995-2002 Diamond Cut Productions, Inc.


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FIVE/LIVE

Demo User's Manual

                                                             Eighth Edition

(FIVE/LIVE Release 5.0D / 5.0 DL)

Proprietary Notice:

Diamond Cut Productions, Inc. owns both this software program and its documentation.  Both the program and documentation are copyrighted with all rights reserved by Diamond Cut Productions, Inc.   See the License Agreement and Limited Warranty for complete information.

Published by:

Enhancedaudio.com

3600 Board Rd.

York, PA  17402

Info@enhancedaudio.com

www.enhancedaudio.com (Sales)

Copyright© (1996–2002) by:

Diamond Cut Productions, Inc.

P.O. Box 305,

Hibernia, NJ 07842, United States of America

Notice:

Diamond Cut Productions, Inc. does not recommend the use of any of its products in emergency communications environments where the failure or malfunction of the product can reasonably be expected to cause compromise of the communications system, or to significantly affect its safety or effectiveness.  Products are not authorized for use in such applications unless Diamond Cut Productions, Inc. receives written assurances, to its satisfaction, that: (a) the risk of injury, or damage has been minimized; (b) the user assumes all such risks; and (c) potential liability of Diamond Cut Productions Inc. is adequately protected under the circumstances.

Special Thanks: Konstantin Themelidis, Gregory E. Miller, CommEdge.com, Ron Bowser of Wyoming, Catalin Grigoras, PhD, Forensic expert, Dan McDonald, Steve Cain, Forensic Tape Analysis Inc., Dan Brown, Jake Ivey, T.J. McCormick, Graham Powell, Peter Mosher, Ontario Centre of Forensic Sciences, Dick Begley, Multimedia Music Australia, Jason Begley, Multimedia Music Australia


Welcome

Thank you for trying another fine product from Enhanced Audio.  We strive to provide you with products that push the envelope in features, but leave enough money in your wallet to afford the envelope.  We will do our best to not only provide you with the best software on the planet, but also provide you with the best support materials we can.  This help file is your window to the product.  Please use it.  Though we will support this product to the best of our ability via telephone, email, and web site support, chances are, everything you need to navigate FIVE and LIVE is right here.  We’ve done our best to make this manual and the other supporting documents as user friendly and intuitive as possible…give it a try…reading is fundamental.

How to Get Started

The full version of the software comes with a 350 page printed manual.  This demonstration version includes a very complete help file containing a Step by Step getting started guide which will have you using the product like a pro in no time.  To access the Step by Step guide, type the word "Step" into the Index box just to the left of this text.  Click on the Step by Step guide link and you'll be led thru a typical restoration project.  This is an easy way to get familiar with our product.  The entire Step by Step Guide with three restoration project examples is also available as a single PDF file from www.enhancedaudio.com

Important Note: We want you to be able to enjoy success with our product quickly.   The Step by Step Guide mentioned above will help you get oriented in just a few minutes.  Should you have additional questions, we also offer great tech support on this demo version.  We'll be glad to answer your questions via email at mailto:support@tracertek.com or via phone at 717-764-9240.

This demo version is limited to files that are 60 seconds or shorter and saving files has been disabled. Other than that, this is a completely functional version of the product.


About the Product

DC FIVE and DC LIVE/Forensics is a comprehensive set of tools designed for audio restoration and enhancement.  This extensive toolkit will allow the user to remove extraneous noise and also enhance the sound from any audio source without degrading the content contained on the original.  Recognizing that there is a tradeoff between the degree of noise removed from a source and the fidelity, transient and frequency response maintained, we have sought to provide the highest level of user control while maintaining ease of use over the variables which effect the audio restoration process.

Professional audio engineers, ham radio operators and audiophile hobbyists use our products for re-mastering, editing, noise removal and signal analysis. It is also used by many local, state, and federal government agencies for Forensics law enforcement applications, particularly in surveillance situations.  Private sector Forensics engineers use it for post processing Forensic recordings of phone calls and surveillance tapes.  Radio and television networks have found the LIVE (real-time signal processing) feature of particular value in their real-time applications.


About the Version

FIVE/LIVE has been in the works for over 2 years.  We’ve added a ton of new cool features.  If you’re an experienced FIVE/LIVE user, use the list below as your table of contents.  It includes every new feature to the product and includes the page number so that you can quickly access all of the newest info without rehashing the stuff you’ve already read.  Here is a list of the features making their debut in FIVE/LIVE:

This version if the DEMO version and is limited to files of 60 seconds or shorter. In addition you cannot save any of the result files. Otherwise this is a fully functioning version of DC FIVE or LIVE5.


History of Product

Diamond Cut Audio Restoration Software was written by two engineers in their spare time to facilitate the very specific needs that arose in their restoration of the Edison Lateral Collection of Test Pressing Recordings, which is located at the Edison National Historic Site in West Orange, New Jersey.  Rick Carlson and Craig Maier developed this program over a nine-year period. When they made it available to the general public many years ago, it was with the idea in mind that if it solved some audio restoration problems for them in dealing with the Edison archive, it might also be of use to others confronted with similar problems.  Since then, it has grown into the most used audio restoration product on the planet, and beyond.


About the Manual

With FIVE/LIVE, we’ve redesigned our user’s manual to try and make it easier to navigate.  With over 400 pages of raw information first passed on to us by our scientists, we tried to make a manual that was basic enough for a first time, non-experienced user, but filled with the kind of detail that any propeller head would love.  We’ve also condensed a good deal of the material…all that means to you is that if you have an interest in the Punch and Crunch Filter, you simply look it up in the Table of Contents.  You’ll not only find all the information describing the tool, but also any tutorial information on that specific tool all in the same location.  Our Tutorial section now contains only multiple tool procedures.  As mentioned earlier, this manual supports both DC LIVE/Forensics and FIVE, so be aware that features native only to the LIVE will be marked accordingly.


Getting Started

Note:  Everything you need to know to install the software, hook up your hardware and begin to use the program is in the next two chapters.  This is must reading.  You’ll find it easy and fun!

So, enough of our yacking…let’s install this software and have some fun.

In this section, we’ll tell you everything you need to know in order to install, configure, check out and start using the software.  Read through this section carefully and perform the steps indicated and you’ll have a perfectly working real time audio restoration workstation!


Step One - Make Sure Your System Meets Our System Requirements

Most machines today are more than capable of handling audio processing.  Hard drives have also grown to a point, where you really have no limits on what you can do…except perhaps time.  As we’ve not yet developed time manipulation outside of the computer, let’s just deal with what you’ll need in the way of a PC.

•	200 MHz Pentium or better.  (Note: Real time or faster performance can be achieved on all algorithms when using platforms based on 233 MHz Intel Pentium processors, or better.  Previewing multiple filters will require a faster processor.)

•	Diamond Cut LIVE's "live, real time feed-through" mode requires a Pentium 300 or faster.

•	16 bit Stereophonic Sound Card with line level inputs, or a "Digital input" Card.  Real time feed through requires a full duplex sound card which can record and play at the same time. *

•	32 Mbytes of RAM for Win98, 128Mbytes for XP.  **

•	Windows 98SE, Windows ME, Windows 2000, or Windows XP, NT 4.0 Sp3+***

•	Audio Source Material

•	An Audio Delivery System (to match whatever source recordings you have)

•	A Hard Drive with enough space to accommodate your Wave files.  A formula is provided to calculate the space requirement but you should have about 2.5GB free to make a full audio CD. (The program only requires about 3 Mbytes for itself)

•	Mouse, Keyboard, and Color Monitor

*  You now possess space age technology for audio restoration.  If you haven’t yet replaced your bundled sound card (which is probably the lowest quality audio component you own), contact Tracer Technologies or visit www.tracertek.com

** If your computer has more than the minimum required to run you OS adequately (64MB on 98SE, 128MB on XP), further increases in the quantity of RAM will not appreciably speed up the FIVE/LIVE algorithms since they are almost totally bound by the processor.  Upgrading to a faster processor WILL allow the filters to run much faster.

*** This product uses DirectX, and requires DirectX  5.0 or higher. Microsoft only officially supports DirectX 3.0a on NT 4.0. There are many unsupported DirectX 5 for NT installations available on the Internet. Diamond Cut Productions does not recommend using NT 4.0 with this product.


Step Two - Installation of Product

This is pretty straightforward stuff.  If you are reading this in the helpfile, you obviously have already performed the following step successfully!

Put the install CD into your drive.  If Auto-run is enabled, installation will begin immediately.  If not:

Press the "Start" Button, choose "Run" from the pop-up menu

Press the "Browse" Button and point to your CD ROM Drive

Find the dcautorun.exe icon and double click on it

The Diamond Cut install program will start. Follow the instructions on the screen.

The software will be installed in a folder called Diamond Cut Productions. A sample Wave file is supplied called DEMO1.WAV.

Complete context sensitive help and Step-by-Step Tutorial Guides are part of the standard Windows Help system available from the Help menu.

Now simply double click the program Icon to run FIVE for the first time.


Step Three - Configure Your FIVE/LIVE Program

FIVE/LIVE does not require any difficult configuration, but it does require that your sound card be installed and working properly. If you have more than one sound card in your system, make sure the one you wish to use has been selected in the Sound Card Selection screen (use the Edit/Preferences/Soundcard Menu).

Next, check the Temp File Path under the Edit/Preferences/General menu. FIVE/LIVE automatically assigns temporary file names for files that are being processed. You should set the temporary drive path for the disk drive that you wish to use for audio editing. This is usually the drive with the most free disk space. Keep in mind that high quality (44.1KHz) stereo recording consumes 10.5MB of disk space per minute.

Generally, this is where we get calls from customers who experience one form of problem or another.  This is usually because most people don’t record onto their computer’s hard drive very often and they may not have their system set up for this type of application.  Before you call us…read the next few paragraphs:

1.	Make sure you have the output of your audio source plugged directly to the Line input of your sound card.  (Do Not Use the Mic Input of your sound card).

2.	If you have a turntable, make sure that you are using either a stereo preamp or the Tape Out on your Stereo system in between the sound card and the turntable.  Your turntable cannot generate enough signal on its own to make a good line level recording.  You’ll need the additional boost applied by either the Tape Out or the Preamp.

3.	If you try to play or record and you get the message "Cannot play the specified format…etc."  This is a message generated by Windows…not our product.  It is telling you that FIVE/LIVE is trying to play a Wave file through a device that is not able to do this.  This could be a result of Windows having a Modem listed as the primary device for playing Wave files…or some other non Wave device.  Check in your control panel settings to make sure that your sound card is listed as the primary playback device.  Also, make sure the FIVE/LIVE is set correctly for your sound card under Edit/Preferences/Soundcard.

4.	If you can hear audio playing through the speakers but the record VU meters are not jumping… Check your sound card’s mixer.  Many times, the Recording mixer is turned down or muted.  Just find the mixer (in many cases it’s a little yellow speaker icon located in your Windows Tray beside your clock), go to File/Properties and look for Recording.  Click on Recording and turn up your Line inputs.  Important note – also make sure your Line Input is selected in this mixer – not your mic input.

We’ll cover more in the Troubleshooting section of the manual, but these are the highlights.


Step Four - Connect Your Computer To Your Audio System

Your sound card has both an input and an output.  The output is used when you play an audio file and the input is used when you record audio into the computer.   Refer to your sound card manual or the little icons on your soundcard to identify which jack is for output (playing back audio) and which is for input (recording audio).

We first want to connect speakers to the output of your sound card so we can hear audio.  You probably already have speakers connected.  If you do, then fine.  Just leave them alone.  If not, you’ll need to connect the output of your sound card to either an amplifier or amplified speakers with line level inputs.

Note:  Your sound card records and plays audio at a certain known level.  This standard level is referred to as Line Level.  Just about all devices play or record with this exact audio level so they’re all compatible with each other.  CD players, Tape Recorders, Amplifiers, Receivers, VCRs, DAT machines, Mini discs and most other devices all play or record this standard Line Level signal.  This means you can plug your tape player output or other line level device directly into the input of your sound card since they both use the same type of signal.

Next, we need to plug some audio into the Input of your sound card so we can record this audio into your computer.  If you are using a line level device, just plug it into the sound card line level input using whatever adapter cable mates your player device (i.e., tape recorder) to the input jack on your sound card.

Big Note on Turntables - There are two common playback devices which are NOT line level and which CANNOT be plugged directly into your sound card.  The first is a Turntable.  A phono cartridge puts out a signal that is much smaller than line level.  Your sound card cannot use this signal as it is – it must be amplified up to a standard Line Level before it gets to your sound card.    Therefore, you must use what is called a preamplifier.  If you have a stereo system with an input jack on the back labeled "phono", you already have a preamp.  It’s in your stereo system.  Just leave your turntable connected to the phono-in of your stereo and connect a Line Level output of your stereo to the sound card input.  Many times, this line level output is marked as "Tape Out".  Now, any audio (including your turntable) you can hear from your stereo system will be available to your computer for recording.

If you don’t have a stereo system to connect your turntable to, or it’s in another room, you’ll need a stand-alone preamplifier.  These are not expensive and you can get one at www.tracertek.com.  Again, this takes the tiny signal from your turntable and amplifies it a bit so that it’s now a standard line level signal which can be plugged directly into your sound card.

Another device which does not put out a standard Line Level signal is a microphone.  If you want to record from a mic, just plug the mic into the jack on your sound card that is labeled Microphone.  If you need to use this mic jack, Again, refer to your manual for your sound card for specifics.

In most cases, you’ll simply either be taking the output of your source to the input of your sound card.  This is pretty easy to visualize since the computer is really acting like a familiar tape recorder with an input and an output.  Many audiophiles have very sophisticated and complicated audio systems which allow many different kinds of hookups.  Therefore, there are many alternative methods for connecting your computer to a sound system in order to be able to use FIVE/LIVE.  Here are several: (NOTE:  average users will not have to use these or other complicated hookups – just supply some audio to the input of the sound card and listen to it on the output)


Method #1: Using a home stereo tape-monitoring loop

1.	Connect a stereophonic magnetic phono pickup system to an audio pre-amplifier with magnetic phono equalization inputs.

2.	Connect your line level sound card input to one of the pre-amplifier's tape recording outputs.

3.	Connect your line level sound card output to one of the pre-amplifiers tape monitoring inputs.


Method #2: Using a DAT machine with digital ins and outs

1.	Connect a stereophonic magnetic phono pickup system to an audio pre-amplifier with magnetic phono equalization inputs.

2.	Connect the DAT machine analog output to a tape monitoring input on the pre-amplifier.

3.	Connect the DAT machine analog input to a tape output on the pre-amplifier.

4.	Connect the Digital Output of the DAT machine to the Digital Input of a "Digital-Only" sound card in your computer.

5.	Connect the Digital Input of the DAT machine to the Digital Output of the "Digital-Only" sound card.


Method #3: Using a mixing board and Analog Sound card

1.	Connect a stereophonic magnetic phono pickup system to a magnetic audio pre-amplifier (these are available without all of the bells and whistles associated with a full-blown home audio pre-amplifier).

2.	Connect the Outputs of the magnetic pre-amplifier to two of the line level inputs on your mixing board (one input for each channel).

3.	Connect the line level outputs of the Analog Sound card to another pair of line level inputs on your mixing board.

4.	Connect your tape recorder (DAT or Reel to Reel or whatever) line level outputs to another pair of line level inputs on your mixing board.

5.	Connect any other input devices you may require into the remaining inputs of your mixing board.

6.	Connect the Main mixer output to your power amplification system.

7.	Connect the Monitor Outputs from your mixer to the line level input of your sound card.

8.	Connect the tape recorder line level input to the Stereophonic Headphones output jack on your mixing board.

Warning:

Method #3 is the most versatile method for setting up a small sound restoration lab.  However, because it is so versatile, feedback loops are easily created which can produce very annoying and potentially dangerous signal levels (to your ears, power amplifier and loudspeaker).   So you must be careful not to allow the output of a device to feedback into the same device when operating the mixing board.  Always think twice before raising a slider control on your mixing board utilizing this method.


Step Five - Turn Your Screen Saver and Background Tasks Off

In some cases when recording, if your screen saver or other background task automatically comes on, it may interrupt your recording or add glitches that you didn’t want.  It is always better to turn this off before recording.    Though this is less of a problem with faster computers, most users will feel more secure if they turn off automatic backups, screen savers, etc.  This allows the computer to fully concentrate on recording clean audio.


Step Six - Choose your Operating Mode

iamond Cut can be operated in either of two basic operational modes.  Choose your operating mode based on your personal preferences.  Beginners will likely find that the Fast-Edit mode is easier to get oriented with since it works more like traditional audio editing programs.  More advanced users will likely switch to the Classic mode since this offers a fully optimized restoration environment.


Fast-Edit One Screen Editing Adds the Element of Speed

The new Diamond Cut Fast-Edit (single file editing mode) mode operates much like a word processor where all editing is done on one file. The original file is not modified until a Save is performed. Fast-Edit mode maintains a separate history file representing the editing sequence and offers almost unlimited Undo capability.  The advantage of the Single file Editing mode is its "greased lightening" speed, leaving you with more time for your domestic chores.  Where editing examples are used in this manual in order to highlight the use of an editing or filtering sequence, the Classic (Source and Destination) mode is utilized.

In this mode, you can use the "Undo" feature found in the Edit menu if you are not satisfied with a particular process.  Also, you can highlight a particular step in the Fast-Edit history monitor in order to quickly jump back to a previous editing state.  The editing processes will be temporarily undone back to the selected point in the Edit history monitor. If you want to permanently go back to a previous editing state you can simply double click on the last edit you want to delete in the Fast-Edit Window.  All editing done after that point will be removed and you can continue your editing session. The Delete function can also be found by clicking with the right mouse button.


Classic Mode

FIVE/LIVE Classic mode always operates in a non-destructive manner. The Source and Destination mode (the Diamond Cut Classic technique) involves the use of a "Source and Destination" set of files. When a file is processed with a FIVE/LIVE filter or effect, the software reads the Source file, modifies it with the selected filter or effect, which then writes it to the Destination file. The main workspace of FIVE/LIVE always has a Source and a Destination file in the Classic Mode. This mode of operation has a few important benefits:

1.	The original source file is not modified, leaving it available for instant comparisons with the processed version.

2.	The original material can always be recovered if the results of processing are unsatisfactory.

3.	Selected sections of the file can be reprocessed using different filter parameters or different filters entirely (see sync mode).

4.	Every filter that is run results in a new file which can have yet another filter run on it.  All these intermediate files are always available so the users can instantly go back one or more steps in the restoration process.


The Source and Destination Workspace

When you open a wave file in FIVE/LIVE’s Classic Mode, two workspaces will appear.  The top one, called the Source Workspace will display an envelope consisting of the program peaks of the wave file just opened.  If you are using Peak files, the entire waveform should be visible in the window. If you have peak files turned off, then the amount of the wave file that will be displayed is determined by your "display limit" settings that can be found in the Preferences section of the Edit Menu.  The display will consist of a black signal on a yellow background. The Destination Workspace just below the Source workspace will contain no waveform information initially, and will contain a gray background color.  Both of these two workspaces display amplitude on the Y-Axis (vertical) and time on the X-Axis (horizontal).  When you initially open a file, the entire file is displayed, and is periodically represented by a sample of the peak of the waveform envelope.   When you Zoom-in on a portion of the waveform, at some value of magnification, you will begin to see continuous waveforms, rather than impulse representations of your wave file signal.  For more information regarding Zooming-In on a wave file or Zooming-Out on a wave file, please refer to the sections entitled "Zooming-In & Zooming-Out on portions of a wave file."  Please note that the active workspace is always shown in yellow.

If you are working with stereo wave files, the workspace will display a pair of waveforms.  The top waveform in either of the workspaces represents the left channel while the bottom waveform represents the right channel.  If the signal is monophonic, only one waveform will be seen in the workspace(s).

At the top of the two workspaces is a Title bar, which contains the name of the opened Source wave file. Above each of the wave file workspaces you will see the Sample Rate that was used to create the file on the right-hand side.  Next to the Sample Rate, you will find the mode in which the file was recorded, or processed, being either indicated as "Stereo" or "Mono".

After a wave file has been processed by one of the functions under the Filter command, the output of that file will be sampled just like the Source file and displayed in the Destination workspace just below the Source workspace.  It will become highlighted in yellow just following the completion of a processing session.

At the top of each of the two workspaces, you will see three time displays.  Each display indicated in Minutes: Seconds: Milliseconds.  The time display on the right-hand side of the workspaces indicates the starting time of the portion of the wave file being displayed in the particular workspace.  The time display on the left-hand side of the two workspaces indicates the ending time of the displayed portion of the highlighted wave file.  When a file is initially opened, the display on the left will indicate 00 : 00 : 00. The left-hand display will indicate the total time duration of the opened wave file.  If you use the Zoom function, the left-hand display will now display the start time of the highlighted Zoom-In portion of the wave file.  The right-hand time display will indicate the end-time of the highlighted Zoomed-In portion of the wave file.  The total time duration of the Zoomed-In highlighted portion of the wave file will be displayed on the status bar located below the workspaces.

At the right hand side of each of the two workspaces, you will see two vertically oriented slider controls next to one another.  These are useful for viewing details in a selected portion of a waveform which has been Zoomed-In on.  For example, there may be a small transient that you want to see in more detail that is riding on top of a much larger waveform.  The control on the farthest right is the display gain control.  Using your mouse, you can move this control up and down in order to change the display gain. Moving it down will increase the gain of the display, making the waveform larger on the display. However, it may get so large as to move the portion of the waveform in which you are interested off of the top or the bottom of the display screen.  The control just to the left of the gain control is the offset control; this is used to move the entire portion of the waveform in which you are interested back into view.  You should experiment with these controls a few times to get a feel for how they behave, and then you will begin to understand their usefulness.

At the bottom of each workspace, you will see the Time Axis Scroll-bar.   This control is also operated by the left mouse button, and is used to move the play pointer to various locations within the display workspace. Sometimes, there can be a few seconds of delay when using this slider, so be patient as it performs the calculations to keep up with your commands.  When you are Zoomed-In on a portion of a file, the slider control can be used to move the display start-point within the highlighted field using either the slider with your mouse, or by using the arrow controls which are located at each end of the slider.  The Time Axis Scroll-bar position is always relative to the entire file length, no matter how zoomed-in on a particular waveform you may be. Clicking on the right arrow button will move the waveform to the left of the workspace 1/10th of the overall display length and clicking on the left-hand arrow button will do the same thing, only moving the file in the opposite direction.  If you click on the Scroll-bar (not the slider control itself,) the waveform will move one full frame to the left each time you click.

Note 1: The Time Axis Scroll-bar is inactive when you are fully zoomed-out.

Note 2: If you are not using Peak Files, then the software only reads the first few Megabytes of a wave file for the initial display.  No wave file processing operations are adversely affected by this action. Portions of your wave file not shown on the display can still by played, filtered and operated on just as if they were displayed.  To set the size of the waveform that will be displayed, use the Preferences dialog box (found under the edit menu) and increase the "Display Length Limit" to the size of the file you wish to be displayed.  Keep in mind that the larger the display size, the longer it will take to initially open a wave file. We recommend the use of Peak Files for faster display.

Note 3:  The Destination Workspace can be converted into an Audio Spectrograph time synced to the Source Workspace.  Please refer to the appropriate section of the manual for details on the operation of the Audio Spectrograph

How to Choose Your Mode:  Choosing between Classic and Fast Editing mode is fairly simple.  If you start with one and decide you want to try the other one, you can simply go to the Edit Menu and click on Preferences and the General Tab.  There you will see a check box that enables Fast Edit Mode.    You then need to exit the program and rerun it for the changes to take place.


Step Seven - Testing Your System

By now you should have installed the software and connected your speakers to the sound card output and some audio playback device (such as a tape player) to the sound card input.  It’s time to make sure your efforts have borne fruit.

The first thing we’re going to do is run the FIVE/LIVE program.  Do this by double clicking the icon on your desktop.  Put away the tips screen and click on File/Open Source.  You’ll now see a familiar windows file selector box.  Navigate to c:/program files/diamond cut productions/Live5/Wavefiles.  Open the file called Demo1.  This is a standard .wav file and was included with the program.  We’ll actually clean up this file in the next section, but for now we just want to play it.

Once the file is selected, it will open in the program and you’ll see the graphic drawing of the audio waveform on your display.  It will look like this:

Figure 1- A file is opened

Notice all the nice icons at the top of the display.  If you move your mouse over any one of them and leave it there for a second, a little box will open telling you what that icon does.  To play our file, we need to find the little icon that looks like a right pointing triangle.  This is the standard symbol for "play" that you’ll find on any tape deck.  Click on this icon and you should hear audio and see the cursor move.  Note:  if you aren’t getting audio at this point, check out our troubleshooting steps later in the manual.

Let the audio play to the end of the file and it will stop automatically and reset itself so you can play it again.  Let’s play it again using a keyboard shortcut.  Shortcuts are simply keys on your keyboard that perform a function without having to use the mouse.  Power users love shortcuts and the Play shortcut is the easiest of all.  Just hit the spacebar now and the audio starts playing.  Hit the spacebar again to stop the audio from playing.  Easy, isn’t it?  Many beginners like to spend hours doing this, but let’s move on to test the recording capability of your system.

For our recording test, we’re going to assume you have a tape recorder connected to the input of your sound card.  A CD player or even a turntable with a preamp uses exactly the same process, so just make sure you’re hooked up correctly.

To record, start your tape recorder playing a pre-recorded tape (or start playing an LP, etc).  You should hear the audio coming from your speakers.  Now click the red Record button on the toolbar in FIVE/LIVE.  A nice record screen comes up which looks like this:

Figure 2- The Record Window

This screen will allow you to set the recording parameters and make sure the recording is happening correctly.  Let’s start by making sure we are set to record a stereo file at 44.1 KHZ, 16 bit.  Just use the drop down boxes to select these parameters.  Note the Recording Device box.  Your sound card should be listed here.  If not, drop the box down and choose the sound card that has the audio being fed to it.

Now click the Pause button in this window.  This puts the program into what is called Record/Pause mode.  The VU meters on the left should now start dancing as the program is actually seeing the audio from your tape recorder.

If those meters are dancing, you are now done with testing your installation.  Click Stop, and then click Close.  Answer "yes" when the program asks if you want to discard this recording.  You now have a perfectly working audio restoration workstation and you can go on to the next section where we’ll teach you to use this powerful tool.

If you found that your meters didn’t jump when you hit the Pause button, the problem is very likely caused by your sound card being set to the Mic input and not the Line Level input.  Full troubleshooting help is included later in this manual, but this common problem is easily and permanently solved by double clicking the little yellow speaker icon in your tray – just to the left of the clock on the bottom of your screen.  Now choose, Options, then Properties.  Click the record button and then click OK.  Now click the check box under Line Input and your meters will start to jump in FIVE/LIVE indicating that all is now well.  You just turned on the Live Level input – which is where our audio is, after all.

Now that your system has been installed and checked, it’s time to get to the software and perform an actual restoration project.

Step By Step Guides

In this section, we’re going to take you by the hand and lead you through some complete examples of typical restoration projects.  You’ll remove noise and "sweeten" the audio.  When you are done, you’ll be licensed to drive the product yourself without training wheels.

This entire training process will take you about 10 minutes and will give you all the basics you need to know to use the program.  Please follow along step by step thru the first two examples – Easy and Advanced Record Restoration.  If you are a Forensics audio user, please go thru all three Step By Step Guides.

In these Guides, we’ll tell you exactly what to do and even explain why it is you are doing these things.  It’s important that you follow along exactly as one step builds upon the next.

Note:  The actions you are expected to take are highlighted in Bold.   Let’s get started!


Easy Record Restoration Step By Step Guide

 First, we need to run the program.  If it’s not already up, just double click the icon on your desktop to start it.  FIVE/LIVE starts with a tip screen, which offers quick suggestions as to how to use the program.  You can disable this by clicking on the check box in this screen.

In this guide, we’re going to use the Classic Source/Destination mode to run you thru the software.   You can choose this mode by clicking on Edit/Preferences in the program.  Make sure the check box that is labeled Enable Fast Edit is unchecked.  If you just unchecked it, close and restart the program to get going in Classic mode.

Next, open the Demo1 Wave file the way we did in the testing section above.

Listen to this file by either clicking the Play button or hitting the spacebar – it’s full of clicks, hiss, low frequency noise, etc.  It’s a mess.   Stop the playback when you’re done listening.

You work with FIVE/LIVE by choosing filters to apply to audio.  Some of the filters remove noise and others enhance the audio though they are all referred to in this guide as filters.  You choose a filter by identifying the type of noise you want to remove and then selecting the filter that removes that type of noise.  Make sense so far, right?

In just about every case when clicks and pops are present, we want to remove them first.  Trust us, this is the right first step whenever you get clicks and pops on records or other recordings.

To remove the clicks, we’ll choose the Impulse filter.  Clicks are short noise impulses so it makes sense to use this filter.  To choose the filter, click on the Filter menu item and choose EZ-Impulse Noise.  You’ll see the filter open like this:

Figure 3- The Impulse Filter Appears

Note that the filter takes up only a portion of the screen.  You can still see the waveform and all the menus and icons.  You can move this filter around on your screen and position it wherever you want.  Just grab the title bar at the top of the filter window and drag it around.  Try it now.

Now let’s focus on the filter itself.  Just about EVERY filter in FIVE/LIVE will work the way this one does.  Once you learn this one filter, you’ll be able to use them all!  Because of this, we’re going to cover this filter in great detail.  Be sure and follow along.

Figure 4- The EZ Impulse Noise Filter

In the filter window, you will see several features.  First, you’ll notice sliders that control various filter parameters.  These sliders can be adjusted while you listen to the audio so you will instantly hear the result of any changes you make.  Our sliders are labeled Scratch and Crackle in this filter.

Next, you’ll have Radio buttons that control other aspects of the filters.  Again, you can change them while listening, so you’ll hear the results instantly.  The Radio buttons in the Impulse filter are Speed and Accuracy.

In every filter, you will have a Preview button.  This is the most important button here.  This button will start the audio playing while it is being processed by the filter.  You will hear the results of the filter instantly.  This makes for easy adjustment of the filters.

Also in every filter, you will have a Bypass checkbox.  This takes the filter in and out of the audio stream instantly.  When you listen to a filter being applied by the Preview button, you may want to be able to compare the processed audio to the original audio.  Clicking this checkbox will bypass the filter and you’ll be hearing the original audio.  Unchecking it will instantly put the filter back into the signal path.  This way, you can "fine tune" even subtle effects with FIVE/LIVE.

Lastly, every filter will have controls for Presets.  A preset is a saved group of settings for this filter.  Go ahead and drop down the Preset box now – it’s the white box at the bottom of the filter.  Click on some of the presets and watch the sliders as they move to good starting points for common tasks.  You can tell a lot by looking at the name of the presets.  Now select the preset labeled Default.  This one is already set up with good settings for our Demo1 file.  Every filter will also have a Save and Delete button that allows you to save your own presets under any name you want – and delete them too!

You’re probably ready to try this filter by now, but there is one more thing that is common to each filter that you should know.  Hit F1 now on your keyboard.  Notice that our online Help comes up with information on this specific filter.  This context sensitive help is available for each and every filter.

Put the Help screen away and let’s clean some audio.  If you’ve already been listening to this filter, then shame on you for jumping ahead.  Simon has not said click the Preview button yet.  As punishment, please go back to page one of this manual and start reading again.  We’ll wait for you here.

Welcome back.  Now let’s get started by clicking the button labeled Preview.  You will hear the audio as it is being filtered.  You will still hear the low frequency rumble and the hiss, but the clicks should be gone.  Let it preview all the way to the end of the file.  Notice that once it reaches the end, it will automatically start over at the beginning.  This is called Looping and is automatic when you are previewing with a filter (thought it can be turned off in the Edit/Preferences screen).

Let’s just confirm that the clicks are gone.  To do this, check the Bypass box in the filter while it is still previewing.  Now the filter is bypassed and the clicks will once again be audible.  Listen for a while and then uncheck the Bypass box.  Now the filter is again doing it job and the clicks are no longer heard.

It’s time to learn how to adjust a filter.  While you are previewing, move the two sliders to the bottom.  This makes the filter less aggressive and it will filter less.  Notice that the clicks return as the sliders are moved down.  Moving them back up results in more and more of the clicks getting removed.  All the filters work this way – you just adjust them while you listen.  As you might expect, if you move them too far up you’ll make the filter too aggressive and you’ll get distorted, stuttering or otherwise bad audio – just move them up enough to get the desired result.  Set them both at 50.

Now click on Cancel in the Progress window seen here:

Figure 5- Monitor your progress

This stops the preview from playing.  By Previewing, we have adjusted the filter and confirmed that it is doing its job.  The next step is to click on the Run Filter button.  This takes the filter just as you have set it and applies it to the demo1 file and creates a new file in the lower window.  This new file has been run through the filter and has the clicks removed. At this point, click on Close in the EZ Impulse filter window since we’re now done with it.

Look at the two waveforms.  The top one is called the Source.  This is where we normally work on a file, preview filters, etc.  The bottom is called the Destination and is the result of our filtering efforts.  You can play either one by clicking in the respective window.  You have not changed your original file at all – rather we’ve created a new cleaned up version.

Now it’s time to remove that rumble sound, but how can we do that if we work on the Source window and we really want to remove the rumble from the semi-clean file in the Destination window?  The answer is a little command under the File menu called Make Destination the Source.  This moves the file in the bottom window up to the top where we can work on it.  Click on File/Make Destination the Source now.

A File Selector box will come up and suggest a new name for this file.  Just click on Save to accept it.  Note:  FIVE/LIVE will automatically assign sequential names to new files.  While you’re new to the program, always just accept its recommendation as to file names.

Now we are ready to remove that low frequency rumble.  To do that, we’ll choose the High Pass filter.  Click on Filter/High Pass now.  A high pass filter will remove all frequencies below a certain point and allow all higher frequencies to pass.    Drop the preset box and select the preset called "Demo audio Wave file de-rumble".  It’ll look like this:

Figure 6- The Highpass filter

We’ll start to go a bit quicker now since you already know what most of the buttons and controls do on this filter.  To find out specific info on this filter, don’t forget you can call for help by pressing the F1 key.  We are going to free you now to preview and play with this filter on your own, but when you’re done, return it to these settings by once again clicking on the demo Wave file preset.  Run the filter when you are ready.

You now have removed two of the annoying noise types in this file.  First, we rid ourselves of the clicks and pops, and then we removed the rumble.  Now it’s time to get rid of that loud hiss sound.  First, remember to move our Destination file up to the Source window.

Now click on Filter and select the Continuous Noise Filter.  This filter is perfect for hiss and other types of continuous noise.  Also, click on View and make sure you have Time Display checked at this time.  The Time Display box shows you various timing calculations with the program and will come in handy as you will see.  Your screen should look something like this:

Figure 7- The Continuous Noise filter is awake

The Continuous Noise filter is one of the coolest in the whole program so follow along carefully here.  This filter will remove just about any continuous noise in a program, but it needs you to give it a sample of this noise.  Once it is able to examine the noise, it will be able to seek it out and get rid of it.  Play the file now using the Play button at the top and listen carefully to the first couple of seconds.  You will hear an area right at the beginning, which contains only the noise and no good audio.  That’s a great spot from which to grab a sample of the noise.

To do this, we need to click and drag with our mouse to select this area.  The area highlighted in yellow in the illustration above is the area we want to select.   Just click about ½ inch from the left edge and, while holding the mouse button, drag all the way left.  Let the mouse button go and you will see an area highlighted in bright yellow.  This is the area you have selected.  Move the mouse pointer over either edge of the selected area and the pointer will turn into a Left/Right indicator.  Click and drag as necessary to select an area from the beginning of the file that is about .5 seconds long.  Use the "Span=" display in the timer window to confirm that your selection is around ½ second long.

Now hit the spacebar.  You’ll play only the selected area.  It’ll be quick.  This allows you to audition a selected area to make sure you are really working on the correct area.  You should only hear the hissing noise.  Again, your selected area should look like the one shown above.

It’s time for the fun part.  You will like this.  Hit the Sample Noise button in the Filter window.  The filter will analyze the noise sample and will display the characteristics of the noise in red.  The blue line is the filter that has been designed to get rid of this noise.  Notice how they track with each other.   Yours should look like this:

Figure 8- Continuous Noise Filter does its job

If you have jumped ahead again and clicked on Preview, there will be no lunch for you today.  First, we’ve got to explain those little blue dots on the filter line.  Notice how the blue line is above the red noise sample?  The higher the blue filter line, the more filtering occurs at that frequency.   The blue dots are called Control Points and allow you to adjust the amount of filtering manually.  We’re going to do that in a second, but the Help file contains all the complete info on this filter.

Before we click on Preview, we need to again select the whole file so we’ll hear the whole thing and not just our ½ second noise sample that is now selected.  To select the whole file, just double click anywhere in the waveform display.  Notice how the whole thing gets highlighted in yellow.  Now, go ahead and click on Preview.

Listen to the audio.  The clicks are gone.  The rumble is gone and the hiss is gone!  Well, not quite.  You can still hear a bit of hiss can’t you?  While Previewing, click the bypass button a few times to take the filter in an out of the signal path.  Yes, the hiss is reduced, but there is still some of it there.  Let’s adjust this filter to get rid of all the hiss.

Make sure your bypass button is NOT checked and make sure you are previewing the audio.  Now grab the 2nd control point from the right side and move it up a bit as shown below.

Figure 9- Tackling the remaining hiss

It’s like magic, isn’t it?  The hiss completely goes away because you told the filter to be a bit more aggressive on the high frequencies.  Remember, moving the blue line up make it filter more.  Now stop the preview, click on Run filter, and close the Continuous Noise filter.

You now have a Destination file which is completely restored.  To finish, click on File/Make Destination the Source, accept the file name and you’re done.  This file is the completed version and all others can be deleted.  You can now exit the program or close the files using the commands under the File Menu.

What you’ve learned in this section

In this quick Step by Step guide, you’ve learned how to launch the program, how to select filters, how to preview them, how to bypass them, how to load and save presets, what are Source and Destination files, how to apply multiple filters to a single file, how various filters work and much more.  Not bad for a few minutes!  Next, we’ll give you a more advanced Step by Step in which you get to use the most powerful feature of FIVE/LIVE, the MultiFilter.


Advanced Record Restoration Step By Step Guide

You learned a lot in the last Step by Step Guide, so let’s continue by restoring this same file using a completely different approach.  This will show you the flexibility and power of FIVE/LIVE and will introduce you to our powerful MultiFilter.  Also, we will do this advanced restoration Step by Step Guide using our new Fast Edit Single Screen mode.  We’ll assume you’ve done the Basic Guide above and are therefore comfortable with the basic functions of the program so we will take this at a little faster pace.  As before, the actions are expected to perform will be in Bold type.  Let’s get started!

First, we want to make sure that you are in Fast Edit mode.  To do this, click on Edit/Preferences and make sure the box labeled Enable Fast Edit is Checked.   If it’s not already checked, put a check mark in the box by clicking it.  Now close the program and then restart it.   The operational mode change will only take affect when you start the program.

Remember, Fast Edit mode is a single screen mode of operation with no Source or Destination file window.  This mode of operation has the advantage of allowing almost instant cuts, pastes and copies on even very large files.  It also may be more familiar to users of traditional audio editing programs.  You can change from Fast Edit to Classic Mode anytime you want.

Next, we want to enable the Fast Edit History box by clicking on View and then checking Fast Edit History.  This will open your History box.  This box will show you everything that you do with this file in the order you do it.  It also allows you to return to an earlier state of the file quickly and easily.

Now open the Demo1 Wave file in the same way we did before.  Select an area near the center of the file using a mouse click and drag.  Your screen should now look similar this:

Figure 10- A highlighted section in Fast Edit mode

Let’s cut out the highlighted area.  Just click on Edit and then Cut.  Boom, it’s gone.  Fast Edit mode is called that because it’s FAST.   You will now have two entries in your History window – the top one is for the original file as opened, and the second one for the version you now see in front of you which is somewhat shorter.

Click on the first entry in the Fast Edit History window now.  The original file instantly comes back!  Did I say this thing was fast?  Everything you do will show up in the history window and you can bounce around your various versions of a file at will.  Click on the first and second entries a few times to confirm this and then right click on the second entry and choose Delete.  You just deleted the cut up version and we’re back to the original file.

Now click on Filter and choose MultiFilter.  Double click anywhere in the waveform display to select the entire file for processing.  Notice how FIVE/LIVE allows you to select an area or the entire file while the filter windows are open.  This is a very handy feature.

The MultiFilter is one of the most powerful features of FIVE/LIVE.  It allows you to string together several filters and preview or apply them all at once.  This saves a lot of time and makes it possible for users to create some incredibly sophisticated filter setups.

Notice that the MultiFilter has an input on the left side and an output on the right side.  Between the input and output is a signal path.  You can drag any filter you want into this signal path.  At this time, remove any filters that are in the signal path by clicking on them and dragging them out of the signal path.

Remember when we did the Basic Step by Step Guide above, we used three filters; the Impulsive noise filter for clicks, the High Pass filter for rumble and the Continuous Noise filter for the hiss.  Drag these three filters into the signal path with the Impulse filter on the left side followed by the High Pass filter and then the Continuous Noise filter on the rightmost side.  Now double click on the Impulse filter icon in the signal path of the MultiFilter.  Your screen should look something like this:

Figure 11- Working with the MultiFilter

To open the individual settings for any filter in the MultiFilter signal path, just double click it and make your settings.  For the Impulse filter, drop the preset box and choose "Demo Audio Wave file Declick".  You could preview and adjust while listening here, of course, but we’ve already got some good settings available as a preset.  Close the Impulse filter window.

Now Double click on the High pass filter and select the "Demo Audio Wave file De-Rumble" preset.  Close the High Pass filter window when done.

Lastly, open the Continuous Noise filter and choose the "Demo Audio Wave file DeNoise" preset.  Close the window.  We used these presets only because they are available.  You can always choose any preset or you can set the filters yourself by just previewing as we did in the Basic Step by Step guide.

Note:  Each individual filter is able to be previewed even in the MultiFilter.

Now we have three filters strung together.  Click the Preview button in the MultiFilter window.  You will hear the audio being filtered by all three of these filters.  Click the Bypass button a few times to marvel at what is happening here.  Feel free to giggle and then cancel the preview.

Now we’re going to add a fourth filter.  Drag the Paragrahic EQ into the signal path and put it last in line (on the rightmost side).  Double click the Paragraphic EQ to open its adjustment window.  Now click on the MultiFilter Preview button.  You may have to move the Paragraphic out of the way to access this button.  You are now hearing the audio filtered thru 4 filters.  Make sure the Bypass button is not checked.  While the preview is happening, find the blue control point in the Paragraphic EQ which is at about 110hz.  You can see this on the horizontal scale on the bottom of the EQ adjustment area.  Move this up to increase the bass frequencies in our old recording.  The more you move it up, the more low frequencies you get.  It should look like this:

Figure 12- Using the Paragraphic EQ

You should notice that you are applying all four filters at once and you are adjusting one of them.  The results of your adjustments are instantly heard while all the filters are being applied.  Cancel the preview, run the filter, and Save the file.  Fast Edit mode requires that you save files when you are done.

What you’ve learned in this Advanced Step by Step Guide

You now know how to choose and use Fast Edit Mode, how to bring up and use the MultiFilter, and how to Preview and adjust while using the MultiFilter.


Forensic Audio Step by Step Guide (LIVE Only)

If you are a new user of LIVE, please go through the Basic and Advanced record restoration guides.  We know you won’t be restoring records with LIVE, but these guides provide an excellent introduction to the overall usage of the program.  We’ll assume you have this basic knowledge for this Forensic Guide.  As before, the specific steps for you to perform are in Bold.

To get started, run LIVE and put it in Classic (Source/Destination) mode.  Now open the file called Forensicdemo.wav from the /program files/Diamond Cut Productions, LIVE/Wavefile/ directory.

Play this file.  Like a lot of Forensic files, the signal to noise ratio is very bad.  The noise is much louder than the voice and this makes it difficult to understand what is being said.  The speech is masked by a loud buzz and has a high frequency and a low frequency noise throughout the piece.  As always we start by listening so that we can identify the noise components that make up our problem.

It’s easy to hear in this file that we will need multiple filters to clean this thing up.  Therefore, let’s use the MultiFilter.  Click on Filter/MultiFilter and bring it up and clear out any filters that are in the signal path.  The file contains a loud Buzz that almost overwhelms the good audio.   To remove hum and buzz, we will use the Harmonic Reject Filter.  It’s called this since hum and buzz are rich in harmonics and this filter can remove them.  We’ll need one of these filters for sure but don’t put it in the MultiFilter yet.

Next, listen again.  There are other noises in this piece that you probably can’t easily identify by listening.  However, we don’t only have to listen to the audio, we can also see it.  Click on View and then choose the Spectrum Analyzer.  Set the Spectrum Analyzer to a resolution of 43.07hz and 1024 FFT size.  Now click on Play and you’ll see the audio as well as hear it.  After listening for a few seconds, hit the spacebar to stop the playback.  The Spectrum Analyzer continues to show you the frequency spectrum of the audio.  Notice that there are some "spikes" of audio in here.  One, for example, is at around 950hz.  You can click on any spot in the display and the Spectrum Analyzer will show you the frequency at that spot.  It should look a lot like this:

Figure 13- Using the Spectrum Analyzer

If you look closely, you can see that there are actually four spikes and some random noise above 12000hz.  We can remove specific frequency ranges with a Notch filter and we can remove that high frequency noise with our Continuous Noise filter.  So now we know all the basics to attack this problem – we’ll need a Harmonic Reject filter, 4 Notch filters and a Continuous Noise filter.    You can see how we used the tools to come up with a general plan of attack.  Now, close the Spectrum Analyzer.

We have already created a MultiFilter preset designed to clean this file.  Select the MultiFilter preset called "Forensic Demo Clean Up Filter".  Notice that our basic filter is not quite the same as we had predicted in the earlier analysis.  We know you want to hear this, so we will explain each component of this MultiFilter preset after we listen to it in action.

Note:  This is a computationally challenging filter.  Even a pretty fast computer might stutter or pause while trying to do all these calculations in real time.  If this happens just stop the preview.   Hit the MultiFilter Preview button and listen.  Click bypass in and out a few times to get the full effect.  Cancel the Preview when you’re ready.

Now click on Run Filter and listen to the resultant file in the Destination window.  Almost as impressive, check out the before and after Waveform display.  It should look like this:

Figure 14- The power of the MultiFilter is obvious

Notice how the total jumble of noise in the source window has changed to a nice speech pattern in the destination window.   You could even make this a bit more "HIFI" if you wanted, but the goal of most Forensic restoration and enhancement projects is to make speech understandable.

Now we will explain the filters we used in this preset.  Double click each filter as we explain it and examine the settings.  Close each filter when you finish looking at the settings.

Brick Wall Filter – this is set to reject all low frequency sound and all sound above 11khz.  Remember the noise in the Spectrum Analyzer at 12khz and above?  We took it out with this filter.

Harmonic Reject Filters – both are set at 60hz, which is a common power line induced noise frequency (50hz in many countries outside the US).  Why do we use two of them?  Because multiple filters of this type will deepen the attenuation of hum and buzz noise.  It’s like one worker digging a hole 2 feet deep and then another also digging a hole 2 feet deep, but the second worker starts in the hole created by the first.  You end up with a 4-foot hole.

Spectral Filter – Remember we thought we might need 4 notch filters?   We could have done that, but our Spectral Filter allows us to set up as many as 32,000 notch filters in one!   Additionally, this filter allows us to either boost or cut any frequency range we want including ones like these that are not related to each other harmonically.  This is just a more efficient way to handle several unrelated noises.

Continuous Noise Filter – This filter is set up to aggressively filter just about all frequencies outside the speech range.  Remember, this doesn’t take out all sound, but targets noise while leaving behind good audio.  It’s a good filter to add to the mix.

Median Filter – This was chosen to add intelligibility to the voice.  It is good for enhancing muffled speech.

Paragraphic EQ – This is set to raise the volume of the speech frequencies.

So you see, we really did follow our initial path by using Harmonic filters, Notch filters and Continuous Noise filters.  We ended up adding some additional tools that further enhanced our audio.  You will likely do the same thing as you perform your own restorations.   This process of first planning a course of action and then making it better is a common one that you will use often.

Close all filters, Make Destination the Source, and exit the program.


Live Mode Step by Step Guide

This guide will introduce you to another major feature of the program.  As before, follow along with the included demo file and perform the actions printed in Bold type.  We assume you have mastered the other concepts we introduced in the Basic and Advanced record restoration Step by Step guides.  Please make sure you have done these earlier guides before attempting this one.

Live mode is a powerful feature of FIVE/LIVE that allows you to remove noise and enhance audio as it happens.  You do not have to record the audio to the hard disk before applying filters.  This is useful for cleaning up radio reception for Hams, SWLs and Forensic users.  If you have your computer connected to your home stereo, you can use Live mode to actually restore your records while you play them.  It can also be used "on site" by Forensic users who may need to listen to (and record) surveillance audio as it happens.

To use Live mode, make sure you have your speakers connected to the sound card output and an audio signal connected to the line level input.  This audio signal can be from a radio, turntable (with preamp) or any other signal.  For this Step-by-Step guide, we’re going to assume you are using a radio to provide a signal to the program.  In this case, you would connect the headphone or other audio out of your radio to the line input of your sound card.

Turn on the radio and you should hear audio from the speakers connected to your computer.  This step depends somewhat on the brand and model of sound card you have, but the great majority of sound cards will simply take whatever audio appears on the input and send it directly to the output of the sound card.  If you don’t hear anything, just keep going in this Guide, as we’ll check everything out a bit later.

Now turn off your Radio and run the LIVE program if you haven’t already and then open the file called "Radiodemo.wav".  It’s in the same directory as the other demo files.    Before we do this live, we’re going to experiment a bit with a recorded bit of audio.

Once the file is loaded, listen to it.  You can hear that there is A LOT of noise in this file.  It is understandable, but would not be fun to listen to due to all the noise.  Once the file has finished playing, open the MultiFilter (under the Filter Menu) and choose the preset called "Live Mode Demo Cleanup – SW Radio".  We have provided in this preset some tools that are useful for this demonstration file.  This will also be a good starting point for your own efforts as you use the program in Live mode.

Double click the Continuous Noise Filter to open its adjustment window.  You can look at the settings for all the tools in the MultiFilter preset and you should be able to understand what they are doing based on your learning so far.  Remember, you can always open a specific filter and then hit F1 to find out about it.  Your screen should look something like this:

Figure 15- Multiple Screens Working Together

We want to bring your attention specifically to the Continuous Noise filter here because we are dealing with a concept that you might not have considered – How do I use this filter in Live mode when I can’t take a sample of the noise?   Remember, the audio is happening in real time and is typically not recorded to the disk at all in Live mode.  It’ll therefore be impossible under these conditions to actually take a noise sample.  Our preset setting here shows how you can deal with this.  Notice that the filter is set to provide a fairly large amount of attenuation to the low frequencies and the higher frequencies (remember, the higher the blue controls, the more noise is removed at that frequency).  With radio signals, we will seldom care about very low or very high frequencies anyway, so this setting gets rid of a lot of noise without unduly bothering our good signal.  The controls between 300hz and 3000hz (which are speech frequencies) were then simply adjusted for good results.  Go ahead and hit the MultiFilter Preview button now.  Click Bypass in and out to hear the improvement.

Notice the slider labeled Attenuation.  This control determines the overall amount of filtering that is going on for the entire filter.  If you move this up, it will filter more overall.  While Previewing, move this around to find the point you like best.  You’ll find that this very weak signal is now quite understandable – if you speak German.  The other filters are also adjustable, of course.  You’ll quickly find that there is a tradeoff here.  The more aggressive you make the Continuous Noise filter, the more digital artifacts you will induce.  Just adjust it to please your ears.

Now stop the Preview and let’s go into Live mode.  Turn on your radio so that you can hear the signal in your computer speakers.  Note:  if you don’t hear anything continue on anyway as some sound cards don’t monitor incoming audio by default.  Double click the leftmost icon in the MultiFilter so that you can adjust your audio settings.  Do that and then click on the button labeled "Live Preview" and the program will immediately start providing filtered audio to your speakers.  You should certainly be hearing audio now.

Your screen should look like this in Live mode:

Figure 16- Live Mode is Engaged

Important -- do you hear two versions of the audio – one a bit later in time that the other like an echo effect?  If so, you need to turn down the input of your sound card.  If you have a standard sound card, just double click the speaker icon in your tray and choose Options then Properties.  Now click on Recording and then click "OK".  Now move the slider labeled Line Input or Line down to the bottom.  The echo should go away!  If you have a more sophisticated sound card, refer to its documentation to find out how to turn off monitoring or turn down the Line Input control.

If you have the DC Live Forensics version of the program, you cannot only process audio in real time, but you can record it as well.  Just click the Log to Disk button.  Additionally, you can also select VOX recording (the program starts and stops recording automatically with the signal that comes in) and have the program mark the segments with the date and time of their occurrence.

Here’s a tip.  You can now add or change the filters as you please.  A good one for radio work is the Dynamic Processor using the ALC mode.  This can help a lot with fades in a signal as it will even out the audio in real time.

What you have learned:

In this Step-by-Step Guide, you’ve learned how to use the MultiFilter in Live mode to process audio as it happens.   You’ve learned how to adjust your sound card for Live mode – remember, however, that some low quality sound cards cannot record and play at the same time and therefore can’t be used in Live mode.  You also learned how some filters could be used to make radio reception much better.


Section 2 - Workspace

In this section we’ll discuss everything you’ll see when you open FIVE/LIVE for the first time.  These items are arranged in their Menu order…meaning that we start at the left, open the first menu and proceed from there.  We’ll also discuss the various ways to access these menu items.  In FIVE/LIVE, many items in the Menu bar can be accessed via Hot Keys and icons on the Tool Bar.

Figure 17- FIVE/LIVE Before Wave is Opened

Important Note:  If you see a Tool or a Menu Item listed here but don’t see it in the program, keep in mind that many items are grayed out or not available until a Wave file has either been recorded or opened within the program.

Figure 18- Menus awaken after Wave file is opened


Open Source

This command opens the desired audio file on which you will perform the FIVE/LIVE processes.  It displays a periodic sampling of the file’s peak amplitude envelope vs. time in the source graphical workspace.  Please note that the software allows you to load .wav, MP3, .aif and .aiff files and automatically converts them to .wav files upon opening.  If you click once on a prospective file, the Wave File Details will be displayed (File Size, Channels, Sample Rate, Bits per sample, Time Length of File).  This function can also be activated by using the file folder icon in the File Tool bar in the upper left hand corner of the workspace or by using the Ctrl + O hot key.


Convert MP3 or AIFF To Wave format (Tutorial)

FIVE/LIVE provide a means for converting MP3 files into wave files for editing.  You can perform your signal processing after the conversion to .wav and then re-convert back to MP3’s with an appropriate encoder if necessary.  The MP3 decoder supports Blade, Lame, Fraunhofer, and many other formats.  Also, it can handle formats with or without an ID3 tag and with either fixed or variable bit rates.   The process of converting from MP3 (*.mp3) or AIFF to wave (*.wav) is extremely simple:

1.	Click on the File menu with the left mouse button

2.	Click on the "Open Source"

3.	Under the "Files of Type" selector box, find .mp3 or Aiff and click on it

4.	Click on "Open"

5.	The file conversion will begin

6.	After a period of time, a waveform will appear in the Source Workspace.  This is a 16 bit, converted wave file representation of the MP3 or AIFF file.  The original MP3 or AIFF file determines the sample rate and number of channels.

7.	It will have the same name as the original file except it will have the .wav extension

8.	If there is already an existing .wav with the same name, a number will be added to the end of the name to distinguish it.

Note 1:  No actual editing takes place on the MP3 file…only the subsequent wave file.

Note 2: Drag and Drop is supported for this feature.


Drag and Drop File Support Now Available

FIVE/LIVE fully supports drag and drop file opening.  You can drag a file onto the open program window or you can drag a sound file icon right on top of the FIVE/LIVE Icon on your desktop.


Save Source

Saves the file to its current directory.  This menu item is only active in Fast Edit mode…in Classic mode, files are automatically saved as you work on them.


Save Source As

Save your Source file as it is, or assign it a new name, sample rate, dithering options or new location.


Wave To MP3 Conversion Now Available Via Popular Outboard Encoders

FIVE/LIVE has the ability to use an external Encoder to help you turn any wave file into an MP3.  This process simply requires that you download one of these popular encoders, go to your Edit/Preferences/Mp3 Encoder and define a path to that particular .EXE file.  From that point on, if you choose to SAVE AS or Batch File Processing and want the end result to be MP3 files, FIVE/LIVE will use that encoder to make the conversion.


Close Source

This command closes a previously opened Source and Destination Wave file.


Open/Create Playlist

This feature allows you to create a playlist of Wave files for reproduction at a later time.  It allows a sequence of Wave files to be transferred to an audio medium without having to manually cue up each Wave file.  This is useful, because you will not have to sit in front of your computer and manually cue each file for transfer to your final destination recording medium.  Instead, you can go out and mow your lawn while this process is taking place.  What follows are the procedures for creating, exporting, and reproducing a playlist sequence.

The Playlist Editor allows you to quickly and easily arrange a playlist of "songs" for sequential playback (to a tape player, Mini disc recorder, etc.) or for CD recording.

Figure 19- The Playlist Editor


Creating A Playlist (Tutorial)

1.	Under the File Menu, click on the "Open Playlist" feature with your left mouse button.

2.	Using the mouse, place the cursor at the beginning of the file name data entry field.

3.	Depress the "Delete" key on your keyboard to clear out any characters in the field.

4.	Type into the field the file name you desire to call the playlist you are about to create.

5.	The Playlist editor window should now appear.

6.	With the left mouse button, single click on "Add Files."

7.	Search for the Wave file that you desire to be the first in the sequence of your playlist.  When it has been located, single click on the file, and it will become highlighted.

8.	Click "Open."  The highlighted file will appear in your playlist.

9.	Go to step # 6.  Repeat this process until your list has been completed.

Note: File order can later be re-arranged by using the "arrow-up" and the "arrow-down" buttons on the playlist editor.


Playlist Export Feature (Tutorial)

The Playlist can be exported so that other programs such as disk label makers and CD-ROM writing programs can use its data.  The feature will export the following parameters:

1.	Title Number in sequence

2.	Title of Selection

3.	Length of Time of Selection

4.	Total Length of Time of the Selections

This feature is activated by merely clicking on the Export button found on the Playlist dialogue box.  It is saved in the *.cue format.  When you click the Export List button, you can specify a file name it will be saved in the standard *.txt format and be opened by many other programs.


Cue sheet support for external CD making programs

FIVE/LIVE now provides the ability to export a playlist in standard .cue format which is compatible with many of today’s popular CD writing programs.  In order to export this list, simply follow the steps below:

For the purposes of this tutorial, we’ll base this export on Golden hawk’s CDR-Win software.  Other popular programs will work in a similar manner.

1.	Click on File/Open/Create Playlist

2.	Give your Playlist a name (Frank is a good one)

3.	Now click on Add Files and choose one or more files to add.  You can select multiple files by using the standard Shift or Ctrl-Click sequence

4.	Click on Export and choose .cue as your Export File type.

5.	Choose the cue file, assign a name and remember the location of your .cue file.

6.	Run CDR-Win and click on the Record Disc Button.

7.	Click on Open Cue sheet and select the file you’ve just saved.

8.	Click on Write CD.


Reproducing a Playlist Sequence (Tutorial)

1.	Under the File Menu, click on the "Open Playlist" feature with your left mouse button.

2.	With the left mouse button, double click on the playlist that you desire to reproduce.

3.	Highlight the top Wave file in the list by clicking on it with the left mouse button.  This will be the first song played in the list.  If you do not desire to start the sequence with the top song, you can click on any one of the Wave files below it, and the playback sequence will then begin with that Wave file.  All Wave files above that particular one will not be reproduced.

4.	Click on "Play List."

5.	The program will commence the playlist starting with the highlighted Wave file, and working its way down the list.  The file that is presently being reproduced will appear highlighted.

6.	When the process is complete, and you do not need to reproduce the playlist again, click on "close."


Open Destination

This command allows you to define the name and desired storage location of the processed version of the Wave file that you are about to create through the use of the various signal-processing tools of the FIVE/LIVE.  The use of this command is optional since the software creates temp files automatically.


Save Destination As

Since FIVE/LIVE does not require the Destination workspace file name to be defined before your audio processing session, this command is used to define a Filename and directory location for your Destination file following the completion of an audio processing session, should you desire to save it.  This function can also be activated by using the file folder icon in the File Tool bar of the workspace or by using the Ctrl + S hot key. Note:  It is almost never necessary to manually save a file in Classic Mode.  The program automatically does this for you.


Close Destination

This command allows you to close a file that has just been processed from the Source file.  The working Destination file will have been stored on a temporary basis in a temp.wav file.  When you attempt to close the Destination file, you will be prompted to indicate whether or not you want to save it.  If you do, then you will be prompted to define a path and a name for your processed file to be saved in.


Make Destination the Source

This command takes the file that has just been processed, and makes it the Source file in a new workspace window.  This is a useful feature, since most sound jobs require several passes utilizing several different signal-processing techniques to affect a complete audio restoration.  When using this command, the program will prompt you to name the file.  It is recommended that you accept the suggested name.  The original Source file may be deleted when making the Destination the Source by using the appropriate checkbox in the dialog box.  This command is grayed out until you have a Destination file.

Important Note:

Be aware that all Temp Files are deleted at the end of your FIVE/LIVE session but no files will be deleted without you being asked if it is okay to do so.


Delete Files

This feature allows FIVE/LIVE to delete a file from a hard drive.  Since .wav files tend to be huge, this command will be used often.  The software will prompt you to be sure that you want to delete the selected file before doing so. Remember that every minute of stereo audio sampled at 44.1 KHz consumes 10.584 Mbytes of disc space, which is useful to know when it comes time to clear up some disc space in order to get ready for your next sound restoration job.


Deleting a Wave file (Tutorial)

1.	Warning!  This operation cannot be undone!

2.	Click on File and a pop down window will appear.

3.	Click on Delete File and the Delete File Dialog Box will appear.

4.	Choose the Drive and the Directory from which you desire to delete a Wave file.

5.	Click on the Filename that you desire to delete in the Filename field. *

6.	Another dialog box will appear, inquiring whether you are sure that you want to delete the chosen file.

7.	If you click on "yes", the file will be deleted.

8.	If you change your mind, and click on "no", the FIVE/LIVE program will revert back to its initial window, and the file will not be deleted.

* Note 1: Multiple files that are sequential in the file listing can be deleted in one operation by clicking on the first item on the list, and then dragging the mouse pointer down to the last file you desire to delete.  The files that are about to be deleted will be highlighted.

*Note 2: Multiple files that are not sequential in the file listing can also be deleted in one operation by holding down the CTRL key at the same time that you click on the appropriate item you wish to delete with the left mouse button. The files that are about to be deleted will be highlighted


Print Screen With Notation Now Added

This commands prints the screen as defined by your choices identified in "Print Setup."  It prints the present FIVE/LIVE screen as "WYSIWYG" (What You See Is What You Get).  Note that the screen is a very complex graphic with high-resolution images for Forensics applications.  As a consequence, some laser printers will exhibit a problem using the Print command because of their rasterization demands – this just means your printer does not have enough internal memory and would be unable to print any graphic of this complexity.  This is not generally the case with inkjet printers.   This function can also be activated by using the Printer icon in the File Tool bar of the workspace or by using the Ctrl + P hot key.


 Printing Help-File Topics

Sometimes it is useful to be able to read Help file topics from paper rather than from your computer screen.  This is accomplished in the following manner:

1.	From the Help file, select the topic you are interested in printing.

2.	Click on the File Menu with the left mouse button.

3.	Click on Print Topic.


Printing A Screenshot (Tutorial)

Though FIVE/LIVE now includes a Print function, you may still find the need to print a specific multi-layered screen.

1.	When you’ve selected the screen you want to print, simply hit Alt /Print Scrn simultaneously.  This will copy the image into your Windows clipboard.

2.	Enter any paint-type program or even a word processor that accepts graphics and hit Ctrl V or use the Paste tool from the Editing menu.


Print Preview

Simply shows a snapshot preview of what you’re about to print.


Print Setup

This command opens the Print Setup dialog box in which you can define the following parameters:

•	Choose the Default Printer or choose some other printer

•	Choose the orientation of your printout sheet:

1.	Portrait  (This orients the paper vertically)

2.	Landscape (This orients the paper horizontally)

•	Choose the paper parameters that you desire:

1.	Size  (Default value is 8 1/2 inches x 11 inches)

2.	Paper Source (Choose between the paper cassette or manual feed)

•	Properties:  This allows you to select amongst the various printer dependent parameters provided by your specific device.


Page Setup

This feature allows you to select among the following printing choices:

1.	Include File Information in printout (File name and Date of last modification)

2.	Print the Source Waveform

3.	Print the Destination Waveform or Spectrograph

4.	Create a Page Title up to 80 characters long (the default title is "Diamond Cut Audio Restoration Tools".)   To enter your title, just type over the default title.  The maximum length that will appear on your printout will depend on the font that you choose, smaller fonts allowing longer character strings.  To define the font size, go to the Preferences menu.  Under Preferences, choose the "Display" tab.  Under the "Display" tab, you will find a field in which you can enter an integer value for "Display Font Size".

5.	Set Margins Spacing


Exit

Note: Exiting will also clean up all temp files.  It will ask you if you want to save any unsaved files with before you can exit.


Undo

This feature allows you to return to a previous version of a destructively edited file after using such features as Mute, Fade-In, Fade Out, Cut, and Copy / Paste / Insert.  After an "undo" is performed, it is removed from the "undo" listing.  You can also access the function using the Ctrl + U Hot Key.

All single file operations can be undone by using the Undo menu item. The default number of undo levels is 10. The number of undo levels is selectable in the Preferences dialog box.  When you close FIVE/LIVE (Exit the program) all undo information will be lost.


Undo Procedure Using Classic editing mode (Tutorial)

1.	Click on the "Edit Menu."

2.	Click on "Undo."

3.	Click on the menu level of undo that you desire.  The top "undo" in the stack represents the state of the Wave file prior to the latest operation which you performed, and the next down in the stack is the one before that, and so on.  Variable levels of undo are provided, and are settable under the Preferences section of the "Edit Menu."


Undo Procedure Using Fast-Edit Mode (Tutorial)

Use the Fast-Edit history window to temporarily go back to the last step. This method does not permanently undo the last action. Use the right mouse button to delete the last operation.  Double click or use the right mouse button menu in the Fast-Edit history window to delete the last action or series of actions.  You will always be prompted by the computer to make sure this is what you want to do.  This operation does not display the undo levels in the Edit/Undo Window as the Classic mode does.


 Copy

This is the key with an icon consisting of two paper documents contained within its perimeter.  It is used to Copy a highlighted selection of a Wave file from either the Source or Destination Wave file, and place it onto the program's clipboard.  This function can also be accessed Copy Icon in the upper left tool bar or by using the Ctrl + C Hotkey.   You can also access Copy and several other functions by selecting an area and using the right mouse button menu.


Copy and Paste Procedure (Tutorial)

1.	Determine the section of the Wave file that you desire to copy for the "copy and paste" operation.  You may use the Zoom-In feature if you desire to copy a very small portion of the Wave file.

2.	At the beginning of the sector of the Wave file which you desire to copy, click down on the left mouse button and keep holding it down as you drag the timing bar (with the mouse) towards the right of the workspace.

3.	Stop dragging the mouse at the end of the desired "copy" sector of the Wave file.

4.	Release the left mouse button.  You will notice that the sector between the two timing bars will remain highlighted in yellow.  This will be the "copy" sector.

5.	Click the right mouse button anywhere in the workspace area and a pop-up window will appear, providing you with three choices.

6.	Click the right mouse button on "copy" and the highlighted sector will be transferred to a temporary storage location on your hard drive.

7.	After the transfer is complete, highlight the area in your Wave file where you desire to "paste over" the previously copied segment using the previously mentioned mouse drag procedure outlined in steps # 2 through 4.

8.	Click the right mouse button again anywhere in the workspace area.

9.	This time, click on "Paste Over."

Important Note:

The timing rules used by "Copy and Paste Over" are as follows:

1.	Copy and Paste Over operations always begin at the leftmost timing marker of the Highlighted area of the workspace.

2.	If the "Copy" sector is shorter than the "Paste Over" sector, the length of insertion is determined by the length of the "Copy" sector of the Wave file.

3.	If the Copy sector is longer than the "Paste Over" sector, the length of insertion is determined by the length of the "Paste Over" sector of the Wave file.


Manual De-Clicking with "Copy and Paste Over (Tutorial)

1.	Listen to your Wave file and determine the location of the click, pop, or thud that you desire to eliminate.

2.	Zoom-In on the section of the Wave file containing the click using the feature having the same name.

3.	Continue Zooming-In alternately listening to the Wave file until you see the troublesome artifact in the FIVE/LIVE workspace.  It will take some training to be able to identify transients visually, so be patient during your learning curve.

4.	Using the left mouse button, highlight a sector of the Wave file just prior or just after the transient event, being careful not to overlap the highlighted sector onto the actual transient.  The highlighted sector must be at least as long (or longer) as the transient event.

5.	Click on "Edit."

6.	Click on "Copy."

7.	Using the left mouse button, highlight the transient event itself.

8.	Click on "Edit."

9.	Click on "Paste Over."

10.	Zoom back out and listen to the Wave file.

Important Note:

The replacement algorithm used in the Impulse Noise Filter is much more sophisticated compared to one used in this manual de-clicking procedure.  Whenever possible, you should use the Impulse Noise Filter to de-click a record.  Only in the unusual or extreme case wherein the Impulse Noise Filter has been unable to automatically remove a particular artifact, should you use this manual process.


Cut

Just as the title implies… Cut allows you to highlight an area and then cut it out of the file.  Cut places the removed area in your clipboard for pasting or eventual elimination.   This feature is useful when it is necessary to reduce the musical portion of a segment for a competitive event in which the total length of the program is governed, and you do not want to eliminate either the beginning or the end of the song to achieve that end.  You can also access this function by using the Ctrl + X hotkey.


Splicing out a portion of a Wave file (Tutorial)

1.	Highlight and play the portion of the Wave file you believe that you would like to splice out.

2.	Play the sector (using the Play button on the toolbar) to make sure that you have identified the correct timing for the segment you wish to remove.  Re-highlight the correct area if necessary.

4.	Click on the Edit Menu.

5.	Click on "Cut."

Important Note:

Fast Edit Mode can perform multiple cuts on even very long files almost instantly.  .


Manual De-Clicking with Cut (Tutorial)

1.	Zoom-In on the area of interest in the Wave file.

2.	Highlight the click or pop impulse using the mouse.  This should be an incredibly small segment.

3.	Click on the Edit Menu.

4.	Click on "Cut".

Important Note:

Removing clicks by cutting, although very easy to use, is not recommended because it actually shortens the total program length from the original.


Paste

As the name implies and similar to every word processor you’ve ever used, Paste allows you to take whatever element you’ve cut or copied and put it back into your current file in one form or another.  We’ve created many different ways to paste your material.  We’ve thought of everything but Paste Eat, which we’ve reserved for the Elementary School edition of the product.


Paste Interpolate

Paste Interpolate allows you to manually correct a recording impulse noise defect such as a tick, pop, click or thud.  To use this function, merely highlight the area in the Source file in which you are observing a noise event.  Next, click on the Edit Menu, and scroll down to "Paste".  Lastly, click on the Interpolate feature, and the event will be replaced with a new waveform.  This new waveform is calculated by a high-order modeling algorithm utilizing up to a maximum of 10,240 samples of data.  You can also access this function by using the "I" Hotkey.


Manual De-Clicking with Paste Interpolate (Tutorial)

1.	From the Source Workspace, listen to your Wave file using the Play feature, and determine the approximate location of the click, pop, or thud that you desire to eliminate.

2.	Zoom-In on the section of the Wave file containing the click using the feature having the same name.

4.	Continue Zooming-In, alternately listening to the Wave file until you see the troublesome artifact in the FIVE/LIVE workspace.  It will take some training to be able to identify transients visually, so be patient during your learning curve.

5.	Highlight the transient event with the mouse drag procedure.

6.	Depress the "I" key on your keyboard.  The transient will be replaced with new signal approximating the audio waveform that should have been there.


Paste Over

Paste Over allows you to insert the portion of the Wave file located in the clipboard over the top of a different location in your Wave file or to other Wave files. (This operation will delete the portion of the Wave file that previously had been in the particular location, installing the temporary file in that position instead.)  The "Copy and Paste Over" feature in FIVE/LIVE can also be used to manually "de-click" or "de-pop" a sound source (see tutorials under "Copy").  This function can also be accessed using the Ctrl + V Hotkey or by right clicking the mouse after you’ve copied or cut a portion of the Wave file.  For various tutorials of Copy and Paste Over, you can refer to the Copy section of this manual.


Paste Insert

Unlike Paste Over, Paste Insert does not wipe out the sector of the Wave file where you desire to place the contents of the "Copy" temporary file.  Instead, it inserts the area your pasting into the targeted area and just moves the existing material over.


Paste Mix

Paste Mix allows you to add or "mix" one file (or a portion thereof) to a second file.  This feature is useful for creating "voice-overs," or inserting special effects on top of a previously created sound track. This feature works in conjunction with the Copy function. In many cases it will require that two files be opened, one in the Source Workspace, and a second in the Destination Workspace.  But this is not mandatory in that you can "paste mix" a portion of a file back onto itself if desired. The file that you open in the Source Workspace can be the file onto which you will be mixing.  The File which you will be establishing as the "voice over" or special effect, might be the one opened in your Destination Workspace.  In other words, you can mix the Destination file into the Source File, in this example. The process can also be performed in reverse, wherein you can mix a portion of the Source file into the Destination File.  These processes are undo-able, so that you can experiment until you are satisfied with the result.  To use this feature, you will be highlighting the portion of the Destination File that you want to mix into the Source file.  You will then use the Copy command to place it on a clipboard.  Then you will highlight the Source file location in which you want the voice-over mixed in.  When you run Paste Mix, you will be able to adjust the Source and Destination gain settings over a range of from +12 dB to – 100 dB.


Paste Crossfade

Paste Crossfade is the cousin of the Paste Mix feature.  It operates in a similar manner, with the difference that there is a time varying function applied to the gain settings, so that a "cross-fade" effect can be produced.  This feature is useful when you want to fade one song (or file) into another, with no "dead-air" in between.  When you run Paste Crossfade, you will be able to adjust the File-1 (clipboard) Start and Stop Gain settings as well as those for the target file (File-2).  You have available four gain controls in total.  You will also be able to control the dynamics of the cross-fade by selecting Linear In, Log In, or Log Out. This feature is undo-able when executed under the Edit Menu. The "Crossfade" feature is also available under the Filter Menu.


Using the Paste Cross-fader (Tutorial)

This method of cross fading is undo-able.

1.	Open the File (song) into the Source Workspace that you desire to be the first in your cross-fade timing sequence.

2.	Open the File (song) into the Destination Workspace that will be the second song in your cross-fade timing sequence.

3.	Highlight, using the mouse drag procedure, the file to which you will be cross-fading in the Destination Workspace.  You must highlight the file from the point of segue all the way to the end of the song, assuming that you desire to maintain the entire song in the sequence.

4.	Click on "Copy" under the Edit Menu.  This procedure may take a bit of time as this relatively large file is copied onto the clipboard.

5.	Next, highlight the end of the file located in the Source Workspace.  The area that you highlight will determine the cross-fade timing interval.  The longer you make this interval, the slower will be the cross-fade sequence.

6.	Click on "Paste" under the Edit Menu.

7.	Next, click on "Paste Cross-fade" under the paste menu.

8.	Set the gain controls in the dialog box as follows: (default values will work)

•	File 1 Levels:

•	Start = -100 dB

•	Stop = 0.00 dB

•	File 2 Levels:

•	Start = 0.00 dB

•	Stop = -100 dB

9.	Choose the cross-fade timing that you desire.  Linear usually produces a pleasing effect.

10.	Click on "OK"

11.	After the processing has been completed, you may click on the play button on the Toolbar to hear your results.


Paste As A New File

This feature provides a convenient means to chop a large file into smaller pieces, and assign new Wave file names to these subset files.  It is useful for creating a number of Wave files that could be listed and quantized for CD-R indexing from a single large file such as that which you might have from transferring a Vinyl LP, or a concert tape recording to FIVE/LIVE.  A popup window will appear in which you can redefine a file name for each "chopped" file subset.


Paste Silence

Simply insert a predetermined amount of silence in your Wave file and select from the Beginning of the file, End of the File or the Beginning of a selected area.


Select All

Selects the entire Wave file.  Can also be activated by using the Ctrl + A Hotkey.


 Pencil Editing

This device is provided for single sample editing of your waveform.  Though not too useful for general waveform drawing, it can be very useful for eliminating tiny clicks and pops that will show up once you’ve zoomed in closer on the waveform.  It will not be active until you’ve zoomed into this level.   You can access this feature by using the Pencil icon located in the toolbar or using the Ctrl + E Hotkey.  Remember, you must be zoomed it before this will activate.  Watch your pencil icon…when you’ve reached the correct resolution, it will activate itself.


Manually Declicking a Waveform With the Pencil Icon (Tutorial)

1.	Zoom in on your problem area using the Zoom icon…a click will look like a sharp mountain peak.

2.	Single click on the pencil icon with the left mouse button.  The button will remain down.

3.	Move the pencil along the waveform area and depress the left mouse button to make the pencil write a new signal.

4.	Many times, just drawing a flat line in place of a click will remove the click and leave no discernable audio residue.


Mute

This feature uses direct hard disk editing to allow you to mute a selected portion of your Wave file.  With Mute, the audio file is reduced to nothing in the selected area, but the area remains. The Mute feature is also useful for getting rid of noise at the beginning or the end of a recording.  You can access this feature via the Ctrl + M Hotkey or by depressing the right mouse button and using that menu.

Important Note:

Do not mute the beginning or the ending of a Wave file before operating the Impulse Noise filter.  Doing so will cause it to function at an extremely slow rate of speed during the muted section, because it will have a very difficult time calculating a signal to noise ratio on a signal containing all zero’s.  Perform the Wave file muting function after all other filter operations have been completed.


Muting Procedure (Tutorial)

1.	Determine the position in the Source or Destination workspace in which you desire to apply the Mute Function and highlight that area.

2.	 If you change your mind regarding the sector that you desire to mute, merely double click the left mouse button anywhere in the workspace area, and the entire file will again become highlighted in yellow.  Then, repeat steps 1 through 4.

3.	Click on "Mute" (Edit Menu), and a dialog box will appear which says "Mute will set the selected section of the file to silence.  Do you want to continue?"  With the left mouse button, click on either "Yes" or "No."


Manual De-Clicking with Mute (Tutorial)

1.	Listen to your Wave file using the Play feature, and determine the location of the click, pop, or thud that you desire to eliminate.

2.	Zoom-In on the section of the Wave file containing the click using the feature having the same name.

3.	Continue Zooming-In alternately listening to the Wave file until you see the troublesome artifact in the FIVE/LIVE workspace.  It will take some training to be able to identify transients visually, so be patient during your learning curve.

4.	Using the left mouse button, highlight the transient event, being careful not to highlight the complete event, and not just a portion of it.

5.	Click on "Edit."

6.	Click on "Mute."

7.	Zoom back out and listen to the Wave file.

Though you are actually muting a segment of the audio file, if you use this method tastefully, you may not hear it in the final playback.


Fade In

Fade In does what the name implies when applied to the beginning of a Wave file.  You can choose between linear or logarithmic envelopes, and you can also choose the time period for the fade in by selecting the portion of the Wave file over which you desire the fade in to occur.  Lastly, you can choose the "start level" for fade in as well as the "stop level."  ("Level" is the start and stop loudness for the Fade-In).  Fade In operates on the selected file (which can be the Source file)


Fade-In Procedure (Tutorial)

1.	Listen to the beginning portion of your Wave file and determine the position near the beginning of your Wave file during which you desire to produce a "Fade-In" effect.

2.	At the beginning of the sector of the Wave file that you desire to apply the "Fade-In" effect, click down on the left mouse button and keep holding it down as you drag the timing bar (using the mouse) towards the right of the workspace.

3.	Stop dragging the mouse when you arrive at a location just prior to the actual beginning of the signal portion of the Wave file.

4.	Release the left mouse button.  You will notice that the sector between the two timing bars will remain highlighted in yellow.  This is the sector during which you have chosen to apply a  "Fade-In" effect.

5.	You can click the right mouse button to hear if you have chosen the correct portion of the Wave file to apply the "Fade-In."

6.	Click on the Edit Menu function.

7.	Under the Edit menu, Click on Fade In - - -

8.	Choose the type of Fade that you prefer; either Linear or Logarithmic.

9.	Set the "Start Level" slider to zero gain (all the way down.)  The default setting for this control is zero gain.

10.	Set the "Stop Level" slider to 0 dB (unity gain.)  The default setting for this control is unity gain.

11.	Click on OK.  The "Fade-In" function will be performed on the chosen portion of the Wave file.

Note: After a Fade-In has been performed; there may be a sector of your Wave file containing some noise at the very beginning just prior to the start of the Fade-In.  This can be eliminated with the Mute function.


Fade Out

Fade-out also does what the name implies. It contains all of the features outlined in the "Fade In" description except that it normally works near the end of the file.  Fade-Out runs under the Edit menu, and unlike the various Filter functions, operates on the selected file (which can be the source file.)


Fade Out Procedure (Tutorial)

1.	Determine the position near the end of your Wave file where you desire to apply the "Fade Out" effect.

2.	At the beginning of the sector of the Wave file where you desire to apply the "Fade-Out" effect, click down on the left mouse button and keep holding it down as you drag the timing bar (with the mouse) towards the right of the workspace.

3.	Stop dragging the mouse when you arrive at a location in the file where you want total silence to occur.

4.	Release the left mouse button.  You will notice that the sector between the two timing bars will remain highlighted in yellow.  This is the sector during which you have chosen to apply the "Fade-Out" effect.

5.	You can click the right mouse button to hear if you have chosen the correct portion of the Wave file to apply the "Fade-Out."

6.	Click on the Edit Menu function.

7.	Under the Edit menu, Click on Fade Out - - -

8.	Choose the type of "Fade Out" which you prefer, either Linear or Logarithmic.

9.	Set the "Start Level" slider to 0 dB (unity gain.)  Unity gain is the default setting for this control.

10.	Set the "Stop Level" slider to the zero gain position (all the way down). Zero gain is the default setting for this control.

11.	Click on OK.  The "Fade-Out" function will be performed on the chosen portion of the Wave file.

Note:  After a "Fade-Out" has been performed, there may be a sector of noise after the "fade-out" and the end of your Wave file.  This can be eliminated with the Mute function.


Single file operations

Because of the nature of several operations, the Cut, Copy, Paste, and Fade menu items operate on the file that is currently selected. This means that a Cut will delete a section of the Source file if it is the currently selected file in the workspace. Likewise, a Fade operation will modify the highlighted section of the selected file (Source or Destination).


 Gain Change

FIVE/LIVE provides a Gain Change feature that is useful for correcting loudness deficiencies on recordings, or to provide the additional headroom required before running the graphic equalizer filter.  Gain Change can be very creatively applied using the contour graphical interface.  It can also be utilized globally on a file, or selectively to bring out a weak vocal, etc.

The following is a summary of the control parameters and the range of adjustment provided for the Gain Change algorithm:

A.	Type (Fade In / Fade Out / Gain Change)

B.	Slope (Linear / Logarithmic / Curve)

C.	Gain Ranges:

1.	+20/-100 dB

2.	+/-20 dB

3.	+/-10 dB

4.	+/-3 dB

D.	Start Level (dB)

E.	End Level (dB)

F.	Shape (Gain vs. Time):

1.	Straight Line (2 Green Cursors) (start and end gain values)

2.	Curved Line  (4 Green Cursors) (curvilinear inflection point controls)

The Graph shows you how you have programmed the gain to change as a function of the selected Wave file time axis.  You can use the mouse to drag the two green control points to establish the time relationship that you desire.  Often, a flat line is appropriate; however, sometimes the loudness of old 78's decreased toward the end of the recording by a few dB.  This can be corrected by a gain correction starting at 0 dB and ending with perhaps 3 dB (depending on the severity of the problem). The reason this occurred is that the early cutting lathes did not provide automatic gain (or frequency response) compensation controls.  When the curve shape is selected, two additional green cursors appear. The two additional green cursors can be moved both vertically and horizontally allowing you to create numerous curvilinear gain vs. time relationships.

Important Warning:

Digital systems, like analog systems, can be overdriven to the point of "clipping" the signal.  This will produce non-desirous distortion (except on rock n' roll).  Before applying a gain increase to a Wave file, study the amplitude of the signal and be sure that you are not adding so much gain as to exceed the dynamic range of the system which is most often set to 2^16 LSB's (or whatever the resolution of the recording that you have digitized).  If you do, signals will appear to flatten out horizontally at their peaks on the Source or Destination Workplace displays.  If you do indeed notice clipping after a gain change, you can Undo this function.

Playback Controls

If you are new to audio processing on a PC, fear not.  The playback controls will be a few familiar friends that you’ll recognize instantly on your tool bar.  These tools closely match their "real world" counterparts on every cassette or tape deck that you’ve ever seen.  They simply help you quickly and easily navigate through your Wave file in any direction and speed you choose.  Because the laws of hardware do not bind us, we’ve been able to improve these controls and make them more accurate and useful than their hardware brethren.


 Rewind

You click here to rewind the cursor in the file.


Pause

This is the key with two vertical lines contained within its perimeter.  When activated, the playback of the Wave file will pause at that location.  Play can be resumed by activating the play button, depressing the spacebar on your keyboard or by simply hitting the Pause button again.


Fast Forward

You click here to move forward in the file.


 Stop

This is the key with the black square with a green dot contained within its perimeter.  It is used to stop either a record or a playback session.


 Record

This is the button with the red square.  It is used to place the FIVE/LIVE program into Record mode.  You can also access this function by hitting the Ctrl + R Hotkey.  Pressing either the hot keys, menu item or clicking on the Record button will activate the Recording Window.  From there, you see the following items:

Figure 20- The Recording Window

•	Recording Level Meters- Just like a standard stereo, you press the Pause button on the Record transport, start your source audio playing, and these meters will report the amount of signal flowing to your sound card.  No adjustment can be made here…simply go to your sound card’s mixer to increase or decrease the signal.

•	Recording Device- If you’ve set up your sound card(s) correctly, this list will display them and allow you to choose your desired recording device.  If nothing is displayed here, it’s time to revisit both Windows and your sound card installation instructions.

•	VOX Recording- This check box activates VOX recording.  If you want to trigger your record automatically on some sort of input signal, this is where it happens.  You simply set a Trip Level that is measured in dB.  Record will remain paused until a signal above the level you’ve indicated is received on input.  Record will stop when the system senses no input at that level for a period determined by the "Minimum Duration" settings.  For more info, see below.

•	Trip Level- Threshold Range is from +1 dB to –100 dB.  This "trips" the system into recording when the desired level is reached.

•	Minimum Duration- Time Range is from 0 to 360 seconds.  After the system begins to record, it will stop automatically when the desired signal stops occurring for the amount of time you’ve dictated in this field.

•	Channels- Allows you to select Mono (1 channel) or Stereo (2 channel) recording

With the ever-increasing penetration of DVD into the market, we’ve added the DVD quality 192khz sampling rate to our sampling rate supported list.

•	Sample Rate- Allows you to choose your desired sampling rate.  Here are the choices:

a.	192.000 or 96.000 KHz. Normally associated with DVD recording and playback, these rates provide bandwidth well beyond the audio spectrum.

b.	48.000 KHz This rate will produce slightly higher values of bandwidth than 44.1 KHz, but is primarily used for pro-audio applications.

c.	44.1 KHz.  This rate will produce a full 20 KHz recording bandwidth, however it consumes disk space at a fast rate - - - 5.29 Mbytes per minute per input channel.  This is the same sampling rate used on commercial Compact Discs.

d.	37.500 KHz* This rate was also used in the very earliest history of digital audio recording.

e.	32.000 KHz* This rate was used in the very earliest days of digital audio recording.

f.	22.05 KHz.  This rate will produce a recording bandwidth of 10 KHz, which is adequate for the restoration of old acoustical recordings.  This setting consumes disk space at the rate of 2.645 Mbytes per minute per input channel.

g.	16.000 KHz *This rate is used primarily in the Forensics field.

h.	11.25 KHz  (This rate will produce a recording bandwidth of only 5 KHz, which is useful for the restoration of old spoken word recordings, telephone conversations, and first generation movie soundtracks.)  It consumes disk space at the lowest rate of only 1.3225 Mbytes per minute per input channel.

i.	Custom  Using this setting, you can fill in any sampling rate you choose under the next  "Sample Rate (Hz) field.  This could be useful in some situations; provided your sound card can support anything but the standard sample rates.

Note: Sample Rates Marked with * are not standard rates and you will not normally use them or find them in the sample rate selection drop down box.   You can select them using the Custom function shown below.

•	Sample Rate (Hz)- If you choose "Custom" sampling rate, this is where you manually type in the sampling rate you want.  (Your sound card must support this rate, however)

•	Resolution- This is where you’ll choose the bit width of your recording. You can choose from the following:

a.	8 bit (Games)

b.	16 bit (most common - CD Quality Audio)

c.	20 bit (professional audio)

d.	24 bit (professional audio)

•	Record Position- This window, like your tape counter, simply tells you where you are in your recording.

•	Transport controls- Just like a common tape recorder, you can Rewind, Pause, Record, Stop, and Fast Forward in your recording.

•	Current System Status- This Gray box normally defaults to the "Stopped" position, but also turns Red on Record and Blue on VOX Waiting.

•	Free Disk Space- Tells you how much space is available for recording on your hard disk.

•	Available Recording Time- Does a quick calculation on the fly of your hard disk space, sampling rate, bit width and gives you an estimate of how much recording time remains.


VOX Recording Allows You To Automatically Start Record Via An Incoming Signal

As indicated above in the Record window, VOX (Voice activated Transmit), or signal activated recording, is possible with the FIVE/LIVE.  This feature allows the recorder to start and stop itself based on the presence or lack of presence on an audio input signal.  The recording activation process is triggered by signal level sensing as determined by your setting of the Trip Level control.  The record de-activation process is triggered by a combination of the signal level sensing system in conjunction with the Minimum Duration control setting.  For example, you will want to set the Minimum Duration to a value long enough so that the recorder does not stop between songs on an LP.  In that example, you would probably want to set the control for about 10 seconds. To use the VOX feature, bring up the Recording window and check VOX recording and set the Trip Level and Minimum Duration appropriately for your specific application.

In the LIVE Forensic Version, you can also perform automatic time stamping of VOX events when you have Log to Disk selected (see the Live Mode section for more information).  This function adds a time stamp for each recording event and is accurate to less than 1 second.


Recording Audio Signals onto your Hard Drive (Tutorial)

•	Before your first recording, please review the quick tutorials for Getting Started.

•	This process needs only to be performed once, if you only have one sound card.  Your chosen setting will be saved.  However, if you have multiple sound cards, this process will have to be repeated anytime you desire to change the input or output configuration to FIVE/LIVE.

•	Now, click on the red Record button found on the toolbar.  You will see the above Record window.  Determine your settings and select the desired recording device from the drop down window.

•	Click on the Pause (Record Pause) button in the recording window.  If your source is analog, adjust the output level of the sound source or the input sensitivity of your sound card until the green VU meter bar graphs are modulating vertically to the maximum degree possible short of activating the red overload indicators.  Please note that this adjustment may consist of a hardware control (gain or volume) of the output signal feeding into your sound card OR you may use the input mixer of your sound card to perform this function.  Sample the loudest portion of the sound source to assure that no overloading will occur when the transfer is made to your hard drive.  Please note that digital sources are not gain adjustable.  Whatever the gain settings that were used initially on this type of source, translates to the gain that you will get when transferring to the hard drive via FIVE/LIVE.  If the original analog to digital transfer was either overloaded or under recorded, it will remain that way.

•	To commence recording, simply click on the Record button.

•	The "Record Position" is analogous to the tape counter of a conventional tape recorder.  It uses realtime measurement. (Minutes: Seconds)

•	To pause the recording, click on the Rec. Pause button on the toolbar.  You may continue to record from the pause position by repeating the method outlined in step #5.

•	Recording can be terminated by clicking on the Stop button.  (Stop is the square button containing a smaller black square within its perimeter).

•	To save your recording, click on the Save button in the "Record File" dialog box.  You will notice that name has already been assigned to your file.  You can either keep the assigned name for your file, or rename it at this time.

Important Note:

FIVE/LIVE is compatible with Wave files that were originally created by other programs.  It is not necessary to record a Wave file using the Recording routine in order to use its processing features.


 Play

This is the key with a black and green arrow, which is pointing towards the right.  This key is used to play your file.  The playback will begin at the leftmost portion of the yellow highlight area of your Wave file.  You can also access this function by pressing your keyboard spacebar.


Playing a Wave file (Tutorial)

1.	Launch FIVE/LIVE.

2.	If you haven't already done so, you must define your Output Device.

A.	To do so, click on Edit, and then on Device I/O Selection.

B.	Choose the output device that you desire, and then click on OK.

3.	Next, click on File, and then on Open and select the file you wish to play.

4.	You will notice that the waveform will appear in black on a yellow background.  This area of the window is the Source workspace.  If your Wave file is monophonic, you will see just one waveform displayed in the workspace.  If your Wave file is stereophonic, then you will see two waveforms displayed, with the top waveform representing the left channel and the bottom waveform representing the right channel.  The length of your Wave file is displayed in Minutes: Seconds: Milliseconds in the timeline at the bottom of the Source Workspace.  It’s a good idea to also run on the Time Display window by clicking on View and then checking Time Display.

5.	To play the file, click on the Play button on the toolbar or depress your keyboard spacebar.  You will notice the playback cursor begin to "march" across the workspace as the system plays the file.

6.	To terminate playback, click on the stop button on the toolbar or hit the space bar.

7.	If only a portion of the Wave file had been highlighted for playback, and now you desire to playback the entire file, double click on the workspace background, and the entire file will become highlighted, and ready for playback.


Pausing and Resuming Playback

1.	To pause playback, click on the Pause button on the toolbar

2.	To resume playback, you can either click on the Play button, Pause button again or depress the keyboard spacebar.

Playing and Pausing Portions of Wave files using the Right Mouse Button

1.	To start the playback of your Wave file at a specific location, use your mouse to move the mouse arrow pointer to the desired location in the Source or Destination workspace.

2.	Single click the right mouse button and you will notice a "Play from here" option in the menu.  Click it.

3.	The file will continue playing until its end unless you click the right-hand mouse button again anywhere in the Source or Destination workspace.  Playback will terminate the moment the right-hand mouse button is clicked.


 Play Looped

 This button allows you to play either the entire file or a highlighted area over and over again back to back.  Playback will continue until the button is depressed again or the space bar is pressed.  You can also access this feature by using the "L" Hotkey.

Note: This feature will not work if the "Play From Cursor" method of playback is selected in the Preferences menu.


Time Bracketed Play Range Added For Convenience
Time Bracketed Play Range

This feature allows you to play a range of time around a selected area.  It is useful when you need to listen for small anomalies on a sound recording in order to find their exact location.  You can choose between the following, which can be activated from the keyboard using the following keystrokes (shown in brackets):  To use this function, select an area by clicking and dragging.  You can play this selected area by using the spacebar or the Play button.  You can also play the selected area PLUS a specific amount of time in front and behind the selected area by simply hitting the 1 thru 4 keys on your keyboard.  For example, if you select an area 10 seconds in length, hitting the 4 key will play starting at a point 2 seconds before the selected area and continue for an area representing 2 seconds after the selected area.

Play Range + 0.5 seconds  (1)

Play Range + 1 second (2)

Play Range + 2 seconds (3)

Play Range + 4 seconds (4)


Timer Recording Allows You To Set Your Computer Like A VCR

The new Timer Recorder feature is useful for such applications as recording your favorite radio broadcast at a defined time, or to commence a surveillance recording based on a particular time of day.  It is a single event 24-hour timer, which clears itself after the event has passed, or can repeat its action the following day by checking the "Re-Occurring Event" box.  To set a "Start Time", click on the number to be changed in the "Start Time" field, and use the up and down arrows until the readout agrees with the desired start time.  The same process is used for setting up the timers "End Time."  The Re-Ocurring Event checkbox will cause the recorder to record the same time span every day.   Use the "File Name for Recording" in conjunction with the "Browse" feature to define the file location and name into which the recording shall be placed.  You must leave the software running for the Timer Recorder feature to work and you must close the Timer Recording window.  You can minimize it, but do not exit the program.  If you want to disable the Timer, click on the "Cancel Timer" checkbox.

Figure 21- The Timer Recording Window


Make Waves

This feature provides you with the capabilities of a programmable audio signal generator.  It can produce Sine, Square, and Triangle waveforms of adjustable frequency and amplitude.  The frequency of these waveforms can also be swept from one value of frequency to another.  The Make Waves generator is also capable of producing random waveforms of the white (and pink or brown when used in conjunction with the Paragraphic equalizer) variety.  Make Waves is useful for calibrating and verifying the performance of the audio equipment used in your sound restoration laboratory, especially when used in conjunction with the onboard Spectrum analyzer.  These tools will also be useful to help you better understand the functionality of some of the filters provided in the application.  The Sweep and Random generator is especially useful for characterizing the frequency response of electrical and acoustical systems.  The Triangle Wave generator is useful for characterizing the time-domain linearity of an audio system or component.  The following controls with their adjustment range are provided:

1.	Start Frequency: 1 to 22,000 Hz.

2.	Stop Frequency: 1 to 22,000 Hz.

3.	Length: (Duration of the tone burst) 1 Millisecond to 5 Minutes.  (Data entry is in seconds.)

4.	Amplitude: 0 dB to -96 dB

5.	Linear Sweep check box (on or off)

6.	Sine, Square, Triangle Wave or Random (white noise)* selector

7.	File Type (Stereo or Mono)

8.	Sampling Rate (11.025, 22.05, 44.1, 48, 96, and 192 KHz)

9.	Resolution (8, 16, 20, and 24 bits)

The following chart summarizes the Harmonic Distortion components produced by the various waveforms provided by the "Make Waves" generator:

Waveform	THD%	3rd Harm	5th Harm	7Th Harm	9th Harm
Sine	0	0	0	0	0
Square	44	33.3	20	13.8	10.8
Triangle	12	11	4.1	2.0	1.3

*Note:

Applying the white to pink noise converter found in the Paragraphic equalizer preset listing to a white, random noise Wave file can create pink noise.


Using the Making Waves Audio Signal Generator (Tutorial)

1.	Click on the "Edit" Menu with the left mouse button.

2.	Click on "Make Waves" with the left mouse button.

3.	Choose between Sine Waves, Square Waves or Random by clicking on the appropriate box with the left mouse button.

4.	Set the desired "Frequency" by clicking on the # or #'s which you desire to change with the left mouse button.  Use the keyboard to enter the new values with the "start frequency" slider control.  The allowable range is from 1 to 22,000 Hz.*

5.	Set the "Length" of the file, which you desire to create using the mouse, and direct keyboard entry.  Data entry must be in terms of seconds.  If you desire 2 minutes, enter 120.  If you desire 10 Milliseconds, enter 0.01.

6.	Set the "Amplitude" which you desire, anywhere from 0 dB to -96 dB.  0 dB is the largest signal value which you can produce, with the peak value of the waveform being + / - 32,767 LSB's.

7.	Click on OK and a Source file will be created for you containing the signal that you have just defined

Important Note 1:

If a sweep generator function is desired, click on "linear sweep" and then adjust the "stop frequency" control to the desired value.  The generator will then produce a linear sweep of frequencies ranging from the start frequency value to the stop frequency value over the interval of time defined by the "length" control.    The Sweep and Random Generator is useful for characterizing the frequency response of electrical and acoustical systems.

Important Note 2:

Pink or Brown noise can be created from the random white noise generator by using the converter found in the preset list under the Paragraphic equalizer filter.  First, create a sample of random (white) noise using the Make Waves Generator.  Next, process it through the Paragraphic equalizer using the above-mentioned factory preset.


Change Sample Rate / Resolution

The Change Sample Rate feature allows you to convert a Wave file from any common sample rate to any other common sample rate.  It also allows you to convert from any common resolution (or file depth) to any other common value. These features are provided with a user selectable interpolation quality since improved interpolation takes more CPU horsepower and therefore more time to make the conversions.  This way, the user can make this tradeoff.  The following controls are provided with the associated ranges of the Change Sample Rate / Change Resolution Feature:

New Sample Rate: 11.025, 22.050, 32.100, 44.100, 48.000, 96.000, 192.000 samples/sec.

Resolution: 8, 16, 20, 24 bits

Conversion Quality: Choose between -

•	CD (16 bit interpolation)

•	Pro (24 bit interpolation)

•	Master (32 bit interpolation)


Dithering Option Added to both Sample Rate Conversion and Save File As

Dither: Choose between –

•	None

•	Flat Spectrum

•	Triangular High Pass

•	Noise Shape 2

To perform a conversion, bring up the file to be converted in the source window.  Select the desired conversion parameters from the outlined choices.  Run the filter and the results will be displayed in the Source window after a period of processing time.  The file will be re-named by incrementing its numerical extension by a value of one, so long as there are no other files with that same incremented name in existence in the chosen file directory.


Preferences

Figure 22- Configure the System Your Way!

Preferences are broken down into five sub-categories that can be accessed by clicking on the desire Tab at the top of the Preferences window.

General:

1.	Drive & Directory for Temporary Wave files

2.	Drive and Directory for Presets

3.	Enable Fast Edit Check box- This turns Fast Edit Mode on or off – you must close down the program and restart it for this change to take effect.

4.	Save Edit SessionError! Bookmark not defined.- When this is checked, you will be queried each time you exit the program as to whether or not you want to save the File Edit History file.  If this feature is not checked, the Edit File History file will be automatically deleted when you exit the program. By saving the edit history, you maintain the ability to go back to any previous editing state without destroying the original file. Keep in mind that this will use more disk space since all of the intermediate edits are stored in temporary files on your hard drive.  Note: The Edit History (or Session File) is stored in the same directory as the Wave file.  It contains the file extension .ses and should not be deleted manually because it will leave the associated temporary files on your hard drive.

5.	Use Status Bar Progress indicator rather than popup dialog- When checked, the progress of a particular filter will be indicated at the bottom of the screen using a progress bar rather than the classic popup dialog box.

6.	Auto Save Default Filter Presets - This feature allows you to choose between the action of returning the filter setting called Default to factory values or to your last personal setting after closing a filter.

7.	Number of Undo Levels Saved- This parameter allows you to choose the number of undo levels of destructive editing which FIVE/LIVE will maintain as stored files on your hard drive.  The default value for this parameter is 10.

8.	Nudge Size- This parameter defines the resolution of the left and right arrow keys on your keyboard as they apply to the Wave file-highlighting feature of FIVE/LIVE.  This parameter is defined in terms of samples.  After highlighting a portion of a Wave file, you can fine tune or "nudge" the highlighted area using the left and right arrow keys, and the Shift key.  The resolution of each click on an arrow key is defined by the value of "nudge size."

9.	Play Options- Allows you to choose "Play Selected Area", which defaults playback to the beginning of a selected area of the beginning of the file each time you press Play, or "Play From Cursor", which, like a tape player, starts from wherever you last played the file.

10.	Marker Reaction Time- This parameter allows you to compensate for your and your system’s lagging reaction time when dropping markers on the fly when using the "M" keyboard accelerator.  Its units are calibrated in milliseconds.

11.	Loop Preview on or off- This feature, when checked, will cause a "previewed" section of Wave file repeat itself endlessly until the filter or effect is canceled or the preview button is clicked on a second time.

12.	VU Meter Scale- Choose from Linear or Logarithmic scales.

Sound Card:

1.	Preview Buffers (2 to 50) (raise this value if you encounter stuttering during preview)- This parameter applies to preview mode only.  It allows you to choose the size of the buffer space that is used by the preview feature.  You can select between 1 to 50 buffers.  1 buffer = 4096 samples. The larger the buffer which you choose, the longer the sample which you will hear before the system repeats itself (stutters) if your system is not fast enough to run a particular algorithm in real time. However, the larger the buffer, the longer will be the delay time before you hear the results of a preview session.

2.	Input Device Selector (Browse to desired sound card Input)

3.	Output Device Selector (Browse to desired sound card Output)

Important Note:  Most tech support calls for "no playback" or "no record" symptoms relate to sound card issues.  FIVE/LIVE uses all standard Windows calls to playback and record audio and does not talk to your specific sound card directly.  Therefore, any sound card that works under Windows will automatically work with FIVE/LIVE.  If you are having problem playing or recording audio, make sure your sound card is working with other applications and make sure you have the correct inputs and outputs selected under Edit/Preference/Soundcard.

Display:

1.	Check "Don’t use peak files (.pkf) for display" (no time display shown when checked)

2.	Check "Sync Mode Scroll Tracking" (synchronizes source and destination displays)

3.	Check "Clean Display" which removes Wave file path name from the display

4.	Check "Show Splash Screen on Startup"  (saves time following software launch)

5.	Display Colors- This set of parameters allows you to choose the Source and Destination Workspace background, highlighted, and waveform colors.  When you click on either parameter, a color palate will appear, and you can click on the color combinations that suit your vision the best.

6.	Display Time Format:

a.	Minutes : Seconds

b.	Samples

7.	Display X-Axis Time Scale

8.	Display Font Size (enter value in terms of "points", 2 to 16)

Spectrograph:

Note 1: Please refer to the Spectrograph section of this manual for details

Note 2: Double clicking in the Spectrograph window can also access the Spectrograph preferences.

MP3 Encoder

Figure 23 - MP3 Encoder Preferences Setup

Setup is provided for an externally provided MP3 Encoder. At this time, we accept popular 3rd party encoders that are available from the Internet.   To allow the program to Save or Batch Process files into MP3 format, you must download an encoder and choose it in this screen.  The Browse button allows you to show FIVE/LIVE where the encoder is located on your hard drive.


File Split and Recombine

Sometimes it may be desirable to split a stereo Wave file into its left and right components, establishing individual Wave files for separate channel processing.  For example, if one of the stereo channels needs some equalization but not the other, using this splitting and re-combining tool can facilitate that function.  This easy to use tool allows you to either Split a Stereo file into two Mono files or combine to Mono files into one Stereo file.  You can select whether or not to open these files after the process is computer.  Note that the L/R buttons on the toolbar allow instant selection of either or both channels for playing and processing.

Figure 24 Easily Split and Recombine files with this cool new feature.


Manage Presets

Import and Export Your Favorite Presets Via The Internet

Figure 25- The Preset Manager

The Presets Archive stores all presets, either factory produced or user produced in folders according to their filter.  A new feature of FIVE/LIVE is the ability to share presets you have created with other users via email or web pages.   You can export any or all of your presets and share them with other users by sending them a single file.  See www.enhancedaudio.com for details.

Presets are stored in a Presets folder in the Diamond Cut directory on your hard disk as a single small file for each preset.  Since FIVE/LIVE comes with over 600 presets, it would be cumbersome to find and share them by sending or receiving one file for each preset.  . The Manage Presets window makes this easy.  You simply Create an Archive by clicking on the button and giving the new archive a name.  Now you can copy any individual or group of presets into this archive.  Double clicking a preset folder will open it and reveal its individual presets.  This archive file will exist as a single filename.zip file on your hard disk and will contain all the presets you copied into it.  Copying presets does not delete them from your own preset list.  This single .Zip file can then be sent to another user and opened using the Open Archive function.  The user can then place any or all of the received presets into any of his or her personal preset folders.


Saving a Preset (Tutorial)

1.	Establish the desired filter settings and states for a particular filter application.

2.	Click on the "Save" button.

3.	Using your mouse, place the cursor at the beginning of the data entry field, and double click the left mouse button.

4.	Delete any characters in the data field with the "delete" key on your keyboard.

5.	Type in a descriptive name for your setting (up to 32 characters in length).

6.	Click on "OK".  Your setting will then have been saved.


Recalling a Preset (Tutorial)

1.	With the left mouse button, click on the down arrow located on the right hand side of the setting list (located at the bottom of the Filter Dialog Box).

2.	With the left mouse button, single click on the filter setting preset description you desire from the listing.


Deleting a Preset (Tutorial)

1.	With the left mouse button, click on the down arrow located on the right hand side of the setting list (located at the bottom of the Filter Dialog Box).

2.	With the left mouse button single click on the filter setting preset that you desire to delete.

3.	With the left mouse button click on the "Delete" button.  A question box will appear.  If you still desire to delete the particular filter preset, click on "yes."  If you do not, click on "no."


Batch File Editor

A Batch File editor is provided, so that you can assemble a group of Wave files with similar problems and apply filters and their associated parameters to this entire list of files.   This will allow you to create your setups and then go out and mow the lawn or perform other chores while your computer and FIVE/LIVE are performing their chores unattended by you.

Figure 26- Process Multiple Files Simultaneously!

The Batch File editor includes the following controls:

1.	A "Filter Menu" which contains all of the FIVE/LIVE filters.

2.	A "File Time Range Selector" which allows you to select the time interval of the Wave files on which you desire to apply the filter / filters.  You can select between

A.	Entire File

B.	Last XX Seconds

C.	First YY Seconds

D.	Range from XX to YY Seconds

3.	The Wave file listing which you will create

4.	A "Select Filter Preset" box

5.	"Run Batch" button

6.	"Save Batch" button (for saving your batch settings)

7.	"Load Batch" button (for recalling a batch)

8.	"Close" button

9.	"Add Files" button to bring up the file manager

10.	"Delete Files" button


Batch File Editor (Tutorial)

1.	Under the "Filter" menu, click on the "Batch File Editor."

2.	Choose a filter that you wish to run a batch of Wave files on by using the left mouse button. *

3.	Select the desired filter preset for your batch processing.  If the desired settings have not already been saved under a preset name, you can double click on the filter icon and then set the parameters you desire for the batch processing run.

4.	Click on "Add Files."

5.	Select the desired Wave files from the file manager.  The file listing will appear in the Wave file-listing box.

6.	To delete a particular Wave file from your batch-mode listing, click on it, which will highlight it, and then click on "Delete Files."

7.	When your list is complete, click on the "Run Batch" button.

8.	The Batch listing can be saved by clicking on the "Save Batch" button.

9.	To recall a particular batch, click on the "Load Batch" button and select the desired batch using the file manager.

* Note:  If multiple filters are to be run, you can use the MultiFilter to create your filter sequence.


 MultiFilter

FIVE/LIVE provides the capability of cascading up to 24 filters together and running them as if they were a single filter.  The elements of the string of cascaded filters can all be unique, or repetitions of the same filter or a combination thereof.  This allows you the flexibility to construct your own favorite sequence of filters, along with their presets, and be able to save the cascade along with all of the filter parameters under one single preset name.

The filter combination thereby created can by previewed or run like any other single filter.   In Live Mode you can also run all of the filters in the chain through a full duplex sound card, using the computer as a digital signal processor.  For details on this mode of operation, please refer to the section on "Live" mode below.

The best way to envision the MultiFilter is to merely view it as another FIVE/LIVE filter that can be customized and sequenced. To create a customized sequence of filters merely click on the MultiFilter icon or select Filter/MultiFilter, and the MultiFilter dialog box will pop up.  On the left, you will see the source input to the system.  On the right of the screen, you will see the output.  The input is that Wave file which is present in the source window, (or in the case of LIVE, it can be the sound card input signal) as is the case with any other filter.  The output of the filter can be previewed bypassed, or run just like any other filter.  It can be run in Live Preview mode as well (see below).

To create your filter sequence, merely drag and drop the desired filters from the filter selection into the signal pathway in your prescribed order.   If you want to change the order of the filters in the cascaded chain, merely click on the filter in the pathway, and drag it to a new location.  To get rid of a filter, simply drag it away from the signal pathway with the left mouse button releasing the button when the filter is away from the signal stream.  To view the parameters of any given filters in the MultiFilter, double click on the filter icon in the cascaded chain and the particular filter dialog box will pop-up.  Adjust the parameters to the values that you desire, and they shall be saved as part of the sequence of filters that you have just constructed.  As with all of the filters, the parameters may be previewed while adjusting the parameters.  Be aware that when long chains of filters have been created, there will be a delay in time before the controls react to your changes.  If you desire to delete a filter, drag the filter out of the chain and release the mouse button.

After you have developed a chain which you find to be particularly effective for a certain type of task, you can save its settings under a descriptive name just like any other FIVE/LIVE filter for ease of recall later.

It is important to note that improved audio performance will be achieved using the MultiFilter as opposed to a sequence of Wave file processing.  The reason is that there will not be the quantization error buildup since the Multi filter will maintain the highest possible resolution for the signal throughout the process from the first filter to the last one in the sequence.  Single filing processing must always convert back to Wave file as the intermediate step between filter operations, producing a buildup of quantization errors.

Figure 27- Run Multiple Filters Simultaneously!


Live Feed-through Mode Added to FIVE

Live (feed-through) mode allows you to bypass the hard disk recording process, making your computer into a real-time feed-through digital signal processor.  Signals are fed into your sound card, processed through the filter or filters chosen, and then fed back out of the output of the sound card for use "live."  Live mode is useful for professional applications such as broadcasting, or surveillance work where a signal needs to be processed in real-time without the intervening process of hard drive recording.  It is also very handy for use in a Ham Radio, Short wave Listening station or even as a permanent part of a home audio system.  This feature is particularly useful for real time audio restoration as may be required by radio stations desiring to play old vinyl or 78's directly on the air.

This feature requires a full-duplex sound card and a fast computer.  We recommend the fastest computer that you can afford, since the faster that your computer can perform the mathematical functions, the more filters that you can cascade in "Live" mode. System "stuttering" is an indication that you have exceeded your computer's ability to keep up with the data processing in real-time.   Also, latency (the time delay for processing) is reduced as the speed of your computer is increased.  For professional applications, we recommend a 450 MHz Pentium 2 or higher for "Live" mode when used in conjunction with the MultiFilter feature, although slower computers will run with increased latency and smaller numbers of cascaded filters.

Live mode is accessed by clicking the button in the MultiFilter window.  To adjust the input parameters for your sound card, double click on the input icon (the turntable) and a dialog box will pop-up. The following input parameters are adjustable:

•	Mono or Stereo

•	Sampling Rate (up to 192 KHz)

•	Resolution (8 bit to 24 bit)

•	Input level indicators are also provided

Adjust the parameters appropriately.  Keep in mind that stereo signals will consume about twice the computer resources compared to monophonic signals.  So, if you do not need stereo, do not use the Live mode in stereo, else you will be adding extra latency to the signal and reduce the number of filters which can be run without stuttering.

Drag and drop the filters of interest into the signal path.  Adjust their parameters by double clicking on each filter, producing its dialog box.

To run the "Live" mode, merely click on the "Live Preview" button, and the signal will be processed through the filters in the signal path.  To adjust the output level, double click on the output device (the loudspeaker icon) and a dialog box will appear to facilitate this function.  You can also make waves in real time using the Live mode.  This feature is also contained in the output device dialog box.

Important Note: When using "Live" mode, it is not advisable to connect the soundcard I/O in an effects loop on a mixing console.  Doing so can produce echo effects due to the latency of the computational process.  Instead, connect the soundcard directly in the signal pathway, which you desire to process.


Live log to disk mode (LIVE ONLY)

You can now "Log to Disk" (to your hard drive) the output content of the signal path being used in Live mode.  This can be useful if it is necessary to create an archive recording of surveillance or a broadcast session.  To perform this function, merely click on the "Log to Disk" toggle button on the Live screen.  Thereafter, whenever you click on the "Live" button, whatever signal is being processed will be logged to the disk under the Filename indicated below the "Log to Disk" toggle button.


Live Mode VOX Time/Date Stamp (LIVE Only)

By checking the box labeled Automatic VOX Time/date Stamp, Live will add a marker each time recording starts in VOX mode.  This is useful in remote location or surveillance recording.  By enabling this function, LIVE will record only when audio is actually present to be recorded and each recording event will be marked with the exact time and date of the individual recording.   This function is useful only in Live mode.


Impulse Noise (Clicks and Pops)

Two Impulse Noise filters are provided in the FIVE/LIVE suite.  The Expert filter provides a very high degree of adjustability for those who want to optimize the system to their own personal preferences.  The EZ Impulse Noise Filter provides automation to simplify the process, with the tradeoff of less user controllability.  Both of these filters are non-linear algorithms used to eliminate pops, ticks, clicks, and crackles from audio recordings.  It is also useful for the elimination of "static" interference from AM, FM, or Short Wave radio broadcasts.   All of these signals generally look like impulses (although sometimes referred to as "spikes"), and therefore the name Impulse filter.  The algorithm essentially monitors for fast events, and when their value exceeds a threshold value, the algorithm blanks out the portion of the file wherein the fast event occurred, and re-inserts a waveform that is an approximation of the signal which would have occurred during the event.  The phase of the inserted signal is aligned to match the point in time in the signal so that there is no discontinuity, and therefore almost no artifact injected into the Wave file.


  EZ Impulse Noise

EZ DeClick and DeCrackle

The EZ Impulse Noise Filter combines the performance of the more controlled performance of the "Expert-Impulse" Filter with some automation and the resultant ease of use.  It has only two controls that need to be set by the user.  One control determines the aggressiveness of the Ez-Impulse Filter’s response to large scratches or impulses. The other control affects its response to "crackle" which is another term for smaller and densely populated impulses.  Two bar graphs provide the user with visual feedback as to the effect that each of the two controls has on the signal.  A tally of "Total Clicks Fixed" is also provided for use as a relative reference and an operational aid.  The following is a listing of the Ez-Impulse Filters controls:

Figure 28- EZ Impulse- Powerful and Easy To Use!

•	Scratch

This control sets the threshold for the detection and interpolation of large clicks, pops and scratches.  Low numbers are least aggressive, while large numbers are more aggressive.  The normalized range for this control is 0 to 100.00

•	Crackle

This control sets the threshold for the detection and interpolation of small clicks, crackle and impulses.  Like the Scratch control, low numbers are least aggressive, while large numbers provide a more aggressive response.  The normalized range for this control is also 0 to 100.00

•	Bar Graphs

Both the Scratch and Crackle filter controls have a "Clicks per Second" bar graph having a range of display of 0 to 400.  These two graphs are useful for setting their respective controls.  Indications that are too low may mean that the system is sensitive enough to pick up noise, while indications that are too high may indicate that the Wave file is being damaged by the action of the filter.  Using this, in conjunction with the "Preview" mode makes the setting of the Ez Impulse Filter very simple and effective.

•	Bypass

This control allows you to hear the unprocessed signal for comparison purposes.

•	Speed/Accuracy

Choose between a fast or more accurate process.  It is recommended that you choose Speed if you wish to preview using a slower machine.


EZ Impulse Noise Procedure (Tutorial)

The EZ Impulse Filter operating procedure is as simple as one, two, three.

1.	Place both the "Scratch" and "Crackle" controls to their lowest positions.

2.	Using "Preview" mode, adjust the "Scratch" Control upwards for the best balance between effective scratch removal and minimized distortion.

3.	Next, advance the "Crackle" control upwards for the best balance between effective crackle removal and minimized distortion.

Also, an array of presets is provided with the EZ Impulse Filter that may produce reasonable results without the need for adjusting the filters controls.  However, the best results will usually be obtained by following the three-step procedure.

Important Note:

The best results are obtained when following the above procedure exactly as stated.


Expert Impulse Noise

As your expertise with the product grows, so too will your demands from the filters.  The Expert Impulse Noise approach allows you to control all of the parameters and therefore obtain the most flexible results.  The following is a summary of the control parameter functionality and range that can be adjusted on the Expert Impulse Noise Filter:

Figure 29- Expert Impulse Noise Filter

•	Threshold

This is the Voltage derivative signal level above which the program decides that an impulse noise event has occurred.  It has a range of adjustment from 1 to 12,000 (in DAC counts).  A good starting point for the threshold value is 1/3rd of the full-scale value of the envelope of your Wave file.  For example, if the Wave file is almost full scale in amplitude (+/- 32,000 counts) set the threshold at around 10,000.

For 78 RPM records, start with a Threshold value of 1000, and adjust up or down depending on the results obtained.  Lower values of Threshold will produce a higher degree of de-clicking.  If it is set too low, however, you will produce distortion on your recording. If you do not want to start at 1000, use the 1/3rd of full-scale rule for your initial setting.

Note: Threshold should be set to its lowest value (slider down) for Vinyl LP applications.  Adjustments should be made using the Tracking adjustment for these applications.

•	Size

This is the number of samples during which the "click" or "pop" event must remain to be defined as an impulse noise event.  It has a range of adjustment from 2 to 60 samples.  Short "clicks" require a smaller setting compared to longer "pops."  A good range of values to start with for Vinyl LP applications is in the 10 to 15 area.  A good range of values to use for non-Vinyl (like 78-RPM records) is between 3 to 7 samples.

•	Tracking

This is the value of rectified output Voltage from a High-pass filter that is used to modulate the threshold Voltage of the filter.  When there is a lot of high frequency information present on the recording, like the crashing of cymbals, it is desirable to move the threshold higher in value so that the transients contained in such sounds are not mistaken to be impulse noise events.  Tracking has a range of adjustment from 1 to 100 (in relative units.)  Tracking is most useful on "high fidelity" recordings that contain a lot of "real" high frequency information such as loud dynamic cymbal crashes, or exaggerated sibilant sounds, which may be interpreted by the Impulse Filter as impulse transients.

Most 78's de-click best with the tracking turned all the way down (to a setting of 1).  For LP's, start with a setting of around 25 to 30, and adjust the value upwards if distortion is heard on high frequency passages or sibilant sounds, until the distortion disappears.  However, if the tracking is set too high, adequate de-clicking may not be obtained.

Note: Tracking should be set to its lowest value (slider down) for non-Vinyl LP applications.  Adjustments should be made using the Threshold adjustment in these applications.  If the threshold control is set too high, the Impulse filter will not completely de-click your recording.  If it is set too low, the filter will create distortion on your recording, especially on the higher registers of the audio scale.

•	Preview Mode

When enabled, this allows you to hear quickly the results of your chosen settings.  If your computer is a slower model, the system may "stutter" when this is enabled, however, the feature can still be quite useful for finding the best settings, since the "stutter" will not appear on the final product.  The slider controls can be adjusted "live" when the preview mode is enabled.  Preview mode is invoked merely by clicking on the "Preview" button on the filter dialog box.

•	Vinyl LP Mode

When this is enabled, a different type of detector algorithm is utilized which is more optimized for the wider bandwidth and narrower clicks and pops encountered with Vinyl LP's.  Vinyl mode works most effectively on Wave files that are sampled at 44.1 KHz.  When this mode is turned off, the detector is more optimized for slower and wider impulse noise.  Unlike most of the other controls, you cannot switch Vinyl LP mode on or off "live" when preview mode is invoked.  Use the following table for determining the correct mode for this selector based on the type of material you are working with:

Sound Source	Vinyl LP Mode
Vinyl LP (Stereo or Mono)	"On"
45 RPM	"On"
FM Impulse Noise	"On"
FM Stereo Impulse Noise	"On"
Acoustical 78's	"Off"
Electrical 78's	"Off"
Cylinders	"Off"
Hill and Dales	"Off"
Movie Soundtrack "Pops"	"Off"
AM or Short-Wave Static	"Off"
Vinyl LP (Stereo or Mono)	"On"

Note:  When you run the Expert Impulse Filter, a dialog box will appear which indicates the Clicks/Second and the Total Clicks Processed.  The Clicks/Second statistic is relative to the timing of your Wave file, and not the time frame in which your computer is processing the data.  This feature is provided to help you determine if you are "trashing" (creating distortion) your Wave file due to Thresholds or Tracking values that are set too low.  When the algorithm's parameters are set too aggressively, the Clicks / Second number will become extremely high, which could be an indication of impending distortion of your Wave file (although it depends largely on the condition of your Source material).

•	Optimization Mode

FIVE/LIVE provides you with two alternative replacement algorithms.  One choice is "Speed" and provides the fastest processing of your file, but will not necessarily put back the most accurate representation of the signal that should have been in the location of an impulse event in the first place.  The second is "Accuracy" which utilizes a higher order modeling algorithm for the replacement waveform.  This choice will take your computer longer to process but produces better sounding and more comprehensive results.

Note 1: Do not mute the beginning or the ending of a Wave file before operating the impulse noise filter.  Doing so will dramatically slow down the filter, because it will have a difficult time calculating the signal to noise ratio of a signal containing all zeros.  Mute the extraneous noises from the beginning and the ending of your Wave file at the end of all of your audio restoration processes.

Note 2: There will be occasions wherein 78-RPM recordings will benefit from the de-clicking action of the vinyl mode impulse filter in conjunction with the tracking control.  If the clicks are small and short, it is worth giving it a try.

Note 3: Multiple passes through the impulse filter, especially in vinyl mode while using the tracking control, can produce ever-improving results.

•	Hind Quaternion Mode (HQ Mode)

FIVE and LIVE have included an enhancement to the Impulse filter called Hind Quaternion mode (a name related to this new method used in the click detector portion of the Impulse filter).  This mode can be used with 78's, vinyl, or any source containing impulsive type noise or clipping distortion.  Its operation couples several related variables in this new detector algorithm together in a logical ratio-metric relationship allowing you to vary them with one control.  The Size slider controls these new variables.  The other controls on the Impulse filter still perform their previously defined functions.  In HQ mode, you will find that you have a much larger degree of control of the detector algorithm, especially through the use of the Size control.  Small fast rise time clicks will be detected optimally with small values of Size while larger and slower events will be best detected with larger settings.  The advantage of this new mode is an improvement in the ability of the detector to reject musical transients while still maintaining good click sensitivity.

•	Universal Mode

Universal Mode provides an adaptive element to the detector portion of the Expert Impulse Filter algorithm.  It can be used with any type of impulsive noise, whether the source is from records (78’s, 45’s, 33.3’s), radio static, or tape head static discharge. Universal Mode is of particular advantage in some wireless surveillance applications where occasional static needs to be mitigated. The advantage of this mode is its ability to adapt to changing signal environments.  The Tracking control is its primary means of adjustment.  The Tracking control effects the systems overall centered value of sensitivity while the Threshold control sets the minimum value that the system can apply to the detector. The size control does the same thing as with the all of the other modes associated with the Expert Impulse Filter.  The disadvantage of the Universal Mode detector is its diminished transient response.  This will cause the system to have some difficulty in dealing with the leading edges of rapidly changing audio material, since it will take some time to re-adjust itself.  This could cause some transient leading edge distortion and/or leading edge missed impulse detection.

Important Note:  These different Impulse noise filter modes (LP, 78, HQ, Universal) all take a somewhat different approach toward identifying clicks and pops.  Since these types of impulses can come in an almost infinite number of sizes and types, a user may select the mode that provides good results over a very wide range of impulsive noise environments.


Expert-Impulse Noise Filter Operating Procedure (Tutorial)

1.	Highlight the portion of your Wave file on which you desire to apply the Impulse Noise Filter.  (You may choose to highlight the entire file or any portion thereof.)  Sometimes, when confronted with extremely stubborn clicks or pops, or radio "static" it may be useful to use the Zoom-In feature first before running the Impulse Noise Filter on a "grouping."

2.	Click on the "Filter Menu" with the left mouse button.

3.	Click on "Impulse Noise."

4.	Start with the "Threshold" control at a setting of approximately 1000 for 78's.

5.	Start with a "Size" setting of between 3 to 7 samples for non-Vinyl applications, and use a setting somewhere in the 10 to 15 sample range for Vinyl LP and 45 RPM record applications.

6.	If you are de-clicking a Vinyl LP record, click "Vinyl LP" on with the left mouse button.

Note:

This feature is also utilized for 45-RPM records.  If you are de-clicking a 78-RPM record or something similar, make sure "Vinyl LP" is turned-off.  (It is important to note that Vinyl LP mode works best on Wave files that have been sampled at 44.1 KHz.)

7.	If you are de-clicking a Vinyl LP record, set the threshold control to its lowest value, and perform all of your adjustments with the tracking control, starting with a setting of 25 to 30.  If you are de-clicking a 78-RPM record or something similar set the tracking control to its lowest value and perform all of your adjustments with the threshold control.

8.	Click on "Preview."

9.	Listen to the "Previewed" version of the processing parameters that you have just set.  If your computer is too slow, it will "hick-up" or "stutter." (Do not be concerned that your final sound restoration will sound like this, since it will not!)  Try to listen "through" the stutter to judge what the Filter is doing.  If the "stutter" is too annoying to make a judgment of the performance of the filter settings use Run filter mode on a selected portion of the Wave file directly into the "Destination" workspace.  Alternately, run the filter, and then listen to the Destination Workspace in order to make judgments regarding your settings.  Iterate until you are satisfied with the results.  Note:  Excessive stuttering may well be caused by having the filter set to be too aggressive.  Even a fast computer will stutter when this filter is finding hundreds of clicks per second.  If this is happening, readjust the filter to be less aggressive.

10.	When the filter is running, you will see a display of "Clicks / Second" and "Total Clicks Processed."Generally speaking, when the threshold is set too low, the program will begin to react to sound transients rather than just noise transients.  If the "Clicks / Second is greater than 30, there is a good chance you are catching sound transients, and creating distortion on the output of the filter.  Most records will show less than 10 clicks per second when the settings are correct (except in extreme circumstances).  Keep adjusting the threshold setting until the clicks are being removed and distortion is not being produced on the filter output.  (The distortion that can occur will be most prevalent on the sibilant sounds.)  Keep in mind that lower value settings of the threshold control will cause the algorithm to be more sensitive to removing clicks and pops.  However, if it is set too low, distortion will also be produced on the sibilant sounds.

11.	If the algorithm is capturing the larger impulses but not the smaller ones, try decreasing the "Size" adjustment, and re-evaluate the results.  (You may also have to decrease the threshold control.)

12.	When you determine the best setting of the controls for your particular Wave file, click Run filter.  When the filter has completed its operation, the results will appear in the "Destination" workspace.


 Continuous Noise

This filter is useful for reducing background "Hiss" and other constant noise from a recording or from a noisy FM radio transmission. It is referred to as a "Continuous" noise filter because unlike Impulse noise, Hiss is present at all times.  When adjusted properly, this filter can almost completely eliminate all residual noise from a recording.  However, it is easy to overuse this filter and leave the recording sounding dead and lifeless, and also introduce digital artifacts into the music.

To use this filter, you must first take a sample of a section of noise. This noise template will then be used to decide what is noise and what is music during the filtering process. It is important to sample a section of the Wave file that does not contain any music so that the filter does not remove signals that contain musical information.

Figure 30- Continuous Noise Filter

The filter graphically shows a frequency spectrum of the sampled noise in red. This spectrum represents the amount of noise at each frequency band in the recording.   The blue line represents the filter design which has been accomplished by the program.   You can use the mouse to move the blue threshold line to tailor the kind of noise reduction that the filter performs.

This filter should only be used on recordings that have little or no impulse noise, or on recordings that have already been processed through the Impulse Noise filter.

This is one of two different types of non-linear filters that can be used to reduce noise from a signal source.  Like the Dynamic Noise Filter, it is useful for reducing "Hiss" from a recording or from a noisy FM radio transmission.  However, unlike the Dynamic Noise Filter, it will also reduce lower frequency noise.  When adjusted carefully, it can almost completely eliminate all residual noise from a recording.  However, when compared to the Dynamic Noise Filter, this filter is a bit trickier to adjust so as to avoid the introduction of digital noise artifacts into the Destination Wave file.  It also can have some detrimental effects on the "presence" and the musical transient content of a Wave file when not properly adjusted.

This filter takes a sampling of your file and converts it into the frequency domain utilizing a Fast Fourier transform.  Next it marches along to the next time interval and performs another Fourier transform.  It keeps repeating this process until the entire Wave file is converted into samples which are no longer representing the time domain, but strictly represented in the frequency domain, with the appropriate Voltage, phase and frequency co-efficients for each window contained in memory.  The entire audio spectrum is divided into up to 4,096 bands by a 8,192 point fast Fourier transform (FFT) algorithm (the resolution of which is user selectable). When a signal in a particular band exceeds a threshold that you can define graphically, then that particular band is allowed to pass its signal from the input of the algorithm to the output of the algorithm.  Lastly, the entire file is then re-converted back into the time domain via an inverse fast Fourier Transform.  So effectively the only time during which bandwidth is provided at any of the 4,096 frequency buckets is when there is a useful signal present in any particular bucket.  Otherwise, the various frequency bands are substantially attenuated.


Increased User Defined Control Points Added To Continuous Noise Filter

By simply right clicking or double clicking anywhere in the Frequency Spectrum of the Continuous Noise Filter, you can now insert your own Control Point.  This can help increase the flexibility and power of the filter.  On right click, the menu that appears allows you to Add a point, Delete a point, or reset all points to –100 dB.

The following is a summary of the control parameters functionality and range of adjustment provided for the Continuous Noise Filter:

•	Attack Time

This is the time required for any of the filters to "open up" on the leading edge of a signal that exceeds the threshold line on the spectral graph.

This represents the time constant normalized value at 1 KHz. The time constant for filter frequencies operating above 1 KHz will be shorter than the setting, and the time constant for filter frequencies operating below 1 KHz will be longer.

(The Attack time constant value is weighted with a -1 slope across the audio spectrum.)  Small values of attack provide excellent transient response, while long values provide a minimization of digital artifacts produced by the system.  A good value to start with for the Attack Time parameter is around 125 Milliseconds. The total range of adjustment for Attack time is 20 to 300 milliseconds.  Smaller settings will improve transient response but allow more digital artifact to pass through.  Larger values will decrease transient response, but will minimize the production of artifacts during the noise reduction process.

•	Release Time

This is the time allowed for any of the filters to "close down" or "decay" following a signal that falls below the threshold line on the spectral graph.  All remaining characteristics of the Release time Constant are the same in nature as the Attack time Constant. A good value to start with for the Release Time parameter is around 50 to 100 Milliseconds.  If there is too much filter "breathing", lengthen this time until you are satisfied. The total range of adjustment for Release time is 10 to 500 Milliseconds.

•	Attenuation

This control sets the degree of attenuation for signals that are present and below the threshold line.  The greater that one sets the Attenuation control, the greater will be the degree of noise reduction.  However, the greater the degree of noise reduction achieved, the greater will be the loss of the sense of "Ambiance" on the resultant recording.  So you must make a careful judgment as the correct tradeoff between noise reduction and ambiance for the material you are dealing with.  A good value for the Attenuation parameter is around 10 dB to start with.  If there is too much loss in signal ambiance, decrease this value.  If you desire more noise reduction, increase this value.  If you increase the attenuation too much, you will begin to introduce some digital aliasing artifacts into the Destination Wave file.  The total range of adjustment for Attenuation is 0 to 100 dB.

•	Threshold (Blue Graphical Threshold Line)

This feature controls the threshold value above which a signal at a particular frequency must exceed before it is passed through to the output of the algorithm without attenuation. It works in conjunction with the "sample noise" button.  Although the continuous noise filter has 1000 discrete frequency bands, it would be inconvenient to have to set each of them.  FIVE/LIVE provides up to 256 inflection points (shown as blue dots connected by blue lines on the graph of amplitude vs. frequency) that can be moved along both the frequency and the amplitude axis.  The software will place these inflection points automatically at approximately 10 dB above the noise floor after you perform the "sample noise" function.  Try these settings first to find out if the results are acceptable.  Thereafter, if there seems to be some noise that needs more attenuation at a particular frequency, adjust the threshold upwards utilizing the left mouse button at the frequency of interest until you are satisfied with the results.  The graphical threshold line is adjustable "live" when preview mode is enabled.  If you want more than the 10 inflection points provided by default, double click using the left mouse button while pointing at the desired position on the graph.  To remove inflection points, use the right mouse button in a similar manner.

•	Threshold Control Grouping

a)	Up & Down "Shift Threshold" Control - This feature allows you to globally shift the entire threshold line up or down independent of frequency.  The feature consists of an up and down arrow box. It is operated via the left mouse button. The amplitude resolution of the control is 4 dB / click. After clicking on either of the arrows, you will see the entire threshold line shift in the direction of the chosen arrow.

b)	"Reset" Control - This feature will restore all of the threshold line inflection points to their original default settings.

•	Keep Residue Function

When enabled, the "Keep Residue" function will allow you to "preview" (hear) or process to the Destination Workspace the algebraic difference between the Source File and the Filters Output.  In essence, you will be listening to the noise which would have been removed from the Source File had this function not been enabled.  It is sometimes useful via this function to be able to hear just how much of the real audio signal along with the noise components which you are removing from the source signal.  However, it is only fair to warn the user not to make final adjustments using this feature, as that technique can be quite deceptive.  It is always best to optimize your parametric settings for the best results while listening to the actual processed filter output signal (i.e. "Keep Residue" function off).

•	Smoothing

The smoothing checkbox applies additional processing that reduces the digital artifacts or "Musical Noise" that may be heard when applying this filter. This is most useful on very noisy recording with high levels of surface noise such as old 78s. Using smoothing, you should be able to increase the level of noise reduction without artifacts by 3dB or more.

The tradeoff, and there are always tradeoffs, is that the frequency selectivity is reduced in this mode. An example of where the smoothing function would not be useful would be when you are trying to remove tones or steady state buzz where you want to maintain better frequency selectivity.

This function does not apply affect the spectral subtraction mode.

•	FFT Size (Resolution)

Choose between 512, 1,024, 2,048, 4,096 and 8,192

The resolution of the continuous noise filter can be adjusted.  This parameter determines the number of frequency bins used by the FFT algorithm.  The actual number of frequency bins produced is the FFT Size divided by two (since the FFT produces data for both the real and imaginary planes.)  Large values of resolution produce the largest degree of noise reduction, while the best transient response will be realized with smaller values.  You will have to experiment with the various binary weighted values to determine the best resolution for the material that you are dealing with.  Listen for the best levels of noise reduction attainable while minimizing any digital artifacts.

Important Note:

This parameter cannot be adjusted while the filter is previewing or running.

•	Overlap (Window Overlap)

This allows you to choose the percentage of overlap between FFT windows.  The range for this adjustment is 10% to 50%.  Often, lower values of % overlap will produce better results in terms of digital artifacts produced, but will take longer for processing.  Conversely, larger values of overlap will produce more digital artifacts, but take less time to process. Start with 50% and lower it to achieve faster transients tracking.


Spectral Subtraction Mode In Continuous Noise

This changes the mode of the Continuous Noise Filter into a spectral subtraction system making it useful for some Forensics noise reduction applications.  As with the normal mode of the filter, one still takes a fingerprint of the noise when operating in Spectral Subtraction mode.  The amplitude signal of the resultant FFT is then subtracted from the entire Wave file signal thereby providing rejection of the highlighted fingerprinted signal.  The offending signal could be an air conditioner, city noise, automobile noise, etc.  Please note that the threshold control is not active in this mode.  To adjust the depth of effect, use the attenuation control.

Important Note:

If the parameters for the Continuous Noise filter are set incorrectly, it has the propensity to produce extremely strange sounds (digital artifacts)...some of them quite comical.  If you hear these "birdies", back down on the attenuation setting and / or some of the graph inflection points and they should disappear.  If not, increase the Attack time and decrease the Overlap percentage.


Continuous Noise Filter Operating Procedure (Tutorial)

This filter is the most mathematically complex of all of the FIVE/LIVE algorithms.  It will, therefore, take the longest amount of processing time to complete its calculations.  This algorithm will benefit the most from the use of a high clock rate computer.  This filter is also the most difficult filter to use correctly.  Aliasing artifacts can be produced when the settings are not correct for the particular Wave file you are attempting to "de-noise."  The first time you use it, it will be worthwhile to spend about an hour playing around with it in order to become familiar with its behavior.

1.	Highlight a quiet portion of the Source Wave file.  Often, this sector will be found at the beginning or at the end of the file, as with the lead-in or the exit groove of a record or the lead in of a tape recording.  The idea here is to capture a section of noise only, but no signal.  This will become the noise floor baseline for the subsequent operation of the Continuous Noise Filter.

2.	With the left mouse button, click on "Filter."

3.	Next, click on "Continuous Noise."

4.	When the Continuous Noise Dialog Box appears, click on "Sample Noise."

5.	Some calculations will be made in the ensuing moments.  When they are complete, a graph will appear showing the Amplitude (in dB) versus the Frequency of the Wave file noise floor.

6.	The measured sample noise spectrum is shown in red.  The noise threshold value versus frequency is shown in blue.  You can set the blue graph threshold value, although FIVE/LIVE will automatically choose some settings for the threshold limit line that is a good place to start.

7.	If you choose to change the graphical threshold contour, follow the procedure outlined in steps 7 through 10.  Using your mouse, place the pointer on the left-most blue threshold marker on the graph (one of ten blue dots).

8.	Depress the left mouse button and move the dot either up or down so that it remains somewhere above the red line graph at the bottom end of the spectrum. The higher this line is from the red line, the greater will be the degree of noise reduction at frequencies near this particular dot.  If the dot is placed below the red graphical line, no noise reduction will be applied to these frequencies.  This is sometimes the preferable setting for the blue threshold line near the bottom end of the audio spectrum (below a few hundred Hertz).

9.	Next, move the next blue threshold marker just to the right of the first one, and using the mouse, set it somewhere above that particular frequency on the spectrum graph.

10.	Repeat this process until all ten threshold markers are located somewhere above the "noise floor" graphical representation of your Wave file.  Now the blue line should be located above the red line at all frequency locations.  Note that the best contour can only be achieved by not only moving the markers along the vertical axis, but along the horizontal (frequency) axis as well.

11.	Set the "Attack" time initially to 25 milliseconds.

12.	Set the "Release" time initially to 50 or 100 milliseconds.  (The "Release" time constant should always be set longer than the "Attack" time constant for a realistic sounding operation of the filter.)

13.	Set the "Attenuation" control initially to 10 dB.  (Higher numbers results in higher levels of noise reduction.)  Too much noise reduction will produce digital artifacts and detract from the "ambiance" of the recording.

14.	Highlight the portion of your Wave file on which you desire to apply the Continuous Noise Filter.  (You may choose to highlight the entire file or any portion thereof.)

15.	Run the Filter.

16.	Play the section that you have just processed, and determine which parameters need modification.  If there is a "lagging" response to audio signals, decrease the "Attack" time.  If there is a "swell" of noise following an audio crescendo, decrease the  "Release" time.  If portion of the audio spectrum is sounding dull, lower the "threshold" line at the frequency of interest.  If a portion of the audio spectrum is sounding noisier compared to the rest then raise the "threshold" at the frequency range of interest.

17.	When you are satisfied with the results, re-run the Wave file (in its entirety) to achieve the final processed results in the "Destination" workspace.

Important Note:

 The threshold line inflection points can be adjusted "live" when you are running the filter in "Preview Mode."  You will be able to hear the effects of modification that you make to the threshold line almost immediately.


Harmonic Reject

The Harmonic Reject filter, which is sometimes referred to as a "comb" or "multiple notch" filter is used to attenuate periodic noises that contain harmonics.  It is capable of attenuating either the odd or even harmonics of the selected fundamental frequency.  This filter is very effective for attenuating a form of hum (line frequency related) noise that is rich in harmonics.  This type of noise is sometimes referred to as "Buzz."  Noise, such as this can be introduced into an audio recording from sources such as light dimmers or switch mode computer power supplies.  "Buzz" can contain odd harmonics of the power line frequency, all the way through the entire audio spectrum.  A 60-Hertz square wave is, by definition, the fundamental component of the waveform plus all the odd harmonics of that frequency up to infinity.   The multiple bands (tines) of the Harmonic Reject filter will attenuate the fundamental as well as all of the harmonic by- products within the audio spectrum.  Should you encounter wide-band line frequency related noise that is asymmetrical, it can produce some even harmonics.  An example of noise containing even harmonics would be that produced by an unsymmetrical sine, trapezoidal or square waveform.  The Harmonic Reject filter can be placed in a mode in which it will attenuate the "evens" rather than the "odds" if you should encounter such noise. Before using the Harmonic Reject Filter, it is often useful to identify the fundamental frequency of the noise signal by using the Spectrum Analyzer found under the "View" menu.

Figure 31- The Harmonic Reject Filter

The following is a summary of the control parameters and range of adjustment provided for the Notch Filter:

1.	Frequency (Fundamental): 5 - 5,000 Hz

2.	Attenuation: 0 to 100 dB

3.	Filter Harmonics:

A.	Odd Only

B.	Even Only

4.	Maximum Harmonic Number: 1 to 500

This filter incorporates the "Keep Residue" feature.  This allows you to hear or keep only the noise component of the original signal.  This feature is useful for "tuning" the filter to the maximum level of noise, so that when you actually run the filter with the "Keep Residue" feature turned off, the noise left behind will be minimized.  Lastly, the keep residue function also allows you to produce "slot" filters, provided the slots that you need are harmonically related.  See Appendix 1 for more details about slot filters.

Important Note:

For severe 60 Hz buzz situations, run at least two passes of the filter.  The first pass should be run with a setting of 60 Hz, odd, and 500 harmonics and the second pass, run with a setting of 30 Hz, odd, and 500 harmonics.  This sequence can be repeated more than once for further buzz reduction.  Don’t forget that you can stack up 2 or more of these filters in the MultiFilter and apply them all at once.


 Dynamic Noise Filter

(Analog Noise Filter)

This is a digital simulation of a dynamic analog filter.  It is useful for dynamically attenuating "Hiss" from old record recordings or from old magnetic tape recordings.  It performs better than a fixed Low-pass filter because it only attenuates high frequencies when there is no high frequency information present above the setting of its "threshold" adjustment. Sometimes this technique is referred to as "single-ended noise reduction". The dynamic noise filter's Low-pass corner frequency is frequency modulated by a rectified envelope signal that represents the amplitude of the signal content above a particular low-pass corner. So, normally, the bandwidth of this filter is limited until some high frequency content is measured by its high frequency detector.  When this occurs, the bandwidth of the filter is opened up to allow the frequency of interest to pass through.  When the signal diminishes again below a threshold value, the filter closes back down to a smaller bandwidth.  The user has the ability to adjust a number of parameters with this filter, including noise threshold, filter frequency, attack time (the time constant associated with the signal whose job it is to increase the Low-pass filter frequency corner), release time (the time constant associated with the signal whose job it is to decrease the Low-pass filter frequency corner after a high frequency event has ceased) gain and filter slope (either 6 dB or 12 dB).  This filter should only be used on recordings that contain little or no impulse noise, or on recordings that have already been processed through the Impulse Noise filter first.

Figure 32- The Dynamic Noise Filter

The Dynamic Noise Filter provides the following slider controls:

•	Noise Threshold

The value of rectified and averaged High-pass signal Voltage above which the output of the Dynamic Noise Filter starts to raise its corner frequency.  Moving the noise threshold slider control vertically raises its value.  This control must be adjusted so that when there are no highs present in the source material, background "Hiss" is attenuated, but when "highs" (such as cymbal crashes or the pronunciation of the letter "S") the filter "opens up".

•	Filter Frequency

This is the 1st order High-pass corner filter frequency which drives the Dynamic Noise Filter Detector / Rectifier / Attack & Release Time Constant circuitry.  For modern reel-to-reel tapes, this parameter will be operated generally up in the 4 to 6 KHz range.  For early 78's, it will be operated in the 1 to 3 KHz range.  The Filter Frequency parameter range is from 200 Hz to 19.99 KHz.    Experimentation will be required to determine the best settings.

Important Note:

The frequency range of adjustment up to 19,999 Hz is only effective when utilizing a 44.1 KHz sampling rate or higher.  At a 22.05 KHz sampling rate, the maximum effective frequency setting will be 10 KHz, and at an 11.025 KHz sampling rate, this value will drop to 5 KHz.

•	Attack Time

The Attack Time slider adjusts the time constant (in milliseconds) associated with the rising edge of a High-pass signal envelope.  Fast music will require smaller values of attack time compared to slow music.  The range of adjustment for the Attack Time parameter is 1 to 300 milliseconds.

•	Release Time

The Release Time slider adjusts the time constant (in milliseconds) associated with the falling edge of a High-pass signal envelope.  This parameter will also require smaller values for fast music compared to the requirements of slow music.  Also, the release time will almost always be set to a period of time greater than the Attack time for the algorithm to sound natural.  The range of adjustment for the Release Time parameter is 1 to 500 milliseconds.

•	Gain

Gain controls the amount of dynamic High-pass filter signal that is summed back into the output of the filter.  This allows you to obtain upward or downward expansion of the high frequency portion of the audio spectrum.  The "neutral" setting for this would be 0 dB that represents no expansion or compression.  Setting this value greater than 0 dB will be produce a "Spectral Enhancer" function.  Values of 0 dB and lower produce a Single Ended Noise Reduction function used to de-Hiss a sound source.  This control is calibrated in dB.

Important Note:

The controls can be adjusted "live" when the preview mode button is clicked.


Dynamic Noise Filter Operating Procedure (Tutorial)

Most of the parameter settings for this filter will vary considerably depending on the content of your particular Wave file.  You will have to experiment to determine the values most to your liking.  The values used below in the Procedure Example will get you started.

1.	Highlight the portion of your Wave file on which you desire to apply the Dynamic Noise Filter.  (You may choose to highlight the entire file or any portion thereof.)

2.	Click on the "Filter Menu" with the left mouse button.

3.	Click on "Dynamic Noise Filter."

4.	Set the Noise Threshold slider all the way down; this is the minimum threshold position of the slider control.

5.	Set the Filter Frequency to around 1.5 KHz.

6.	Set the Attack Time to about 5 mSec.  (Unless you are attempting to obtain some sort of special effect, the Release Time should always be set to a value greater than or equal to the Attack Time.)

7.	Set the Release Time to around 50 mSec.

8.	Set the Gain Control to 0 dB.  (This control should only be set to higher numbers if a "Spectral Enhancement" effect is desired.)  This will modify the "effective bandwidth" of your recording by incrementally amplifying the high end of the spectrum when there are enough highs present to trip the detector.  This can be used to increase the "presence" of a recording, or to enhance the sound of a vocalist.)  If more noise reduction is desired set the gain control to higher negative values.

9.	Click on "Preview."

When the filter is operating properly, "Hiss" will be reduced, but when there is high frequency content on the recording, the filter should "open up" and pass through the "highs".  If the Threshold is set too high, the filter will never open up, and the Wave file will sound "dull" although "Hiss" may be reduced.  If the Threshold is set to low, the filter will always be opened up to full bandwidth, and there will be no noise reduction.  If the Attack time is set too long, there will be a delay heard before the filter changes bandwidth on musical high frequency transients such as cymbal crashes.  If the Release time is set too long, there will be a residual "Hiss" left behind after a high frequency musical event, which will decay out, but too slowly.

When you determine the best setting of the controls for your particular Wave file, click Run filter.  When the filter has completed its operation, the results will appear in the Destination Workspace.


Low Pass Filter

This filter is called a Low-pass filter because it only passes through signals that are lower than its set corner frequency. It attenuates high frequency signals above the corner frequency.

The effect can be similar to turning down the treble control on a home stereo except that the Low-pass filter is much more flexible. This filter can be somewhat useful for reducing hiss in a recording, but care must be taken not to reduce the "presence" of a recording by eliminating too much of the high end musical content at the same time.


Chebyshev Response and 4th Order Slope Added To Low Pass Filter

It is most useful where a recording does not contain any musical information above a certain frequency, and you wish to eliminate the high frequency noise that would otherwise be present.

This is a digital simulation of a conventional Low-pass filter.  It is created using an Infinite Impulse Response (IIR) algorithm having a Butterworth or Chebyshev characteristic (for the higher order slopes). Frequencies below the "corner frequency" are passed through to the output, and frequencies above the corner are attenuated. The degree to which higher frequencies are attenuated is determined by the slope (order) of the filter.  Four slopes are provided.  They are 6dB / Octave, 12 dB / Octave, 18 dB / Octave and 24 dB / Octave.  The higher the slope, the more attenuation will occur to frequencies above the corner frequency. The corner frequency is the frequency that you choose, and it is defined as the frequency at which the signal has been attenuated by 3 dB.  This filter can be somewhat useful for reducing hiss in a recording, but care must be taken not to reduce the presence of a recording by eliminating too much of the high end musical content at the same time. When used selectively, this filter can also be used either to "De-Ess" an overly sibilant vocal, or reduce harsh harmonic distortion products that may have resulted from occasional master recording overloading (clipping).  For forensic recordings, this filter can be used to remove most sounds whose frequencies are above the speech spectrum by setting the corner frequency to around 3000hz.

Figure 33- The Low Pass Filter

The higher order (12, 18, & 24 dB / Octave) Low-pass filters are of the Butterworth or Chebyshev types, depending on your choice.

The following is a summary of the control parameters and range of adjustment provided for the Low-pass Filter:

A.	Frequency: 5 - 19,999 Hz

B.	Filter Slope: 6, 12, & 18 dB / Octave

C.	Preview Mode Button: On / Off (The slider control can be adjusted "live" when preview mode is on.)

D.	Filter Type: Choice of Butterworth or Chebyshev

Important Note:

The frequency range of adjustment up to 19,999 Hz is only effective when utilizing a 44.1 KHz sampling rate.  At a 22.05 KHz sampling rate, the maximum effective frequency setting will be 10 KHz, and at an 11.025 KHz sampling rate, this value will drop to 5 KHz.


Low pass Filter Operating Procedure (Tutorial)

1.	Highlight the portion of your Wave file on which you desire to apply the Low-pass filter.  (You may choose to highlight the entire file or any portion thereof).

2.	Click on the Filter Menu with the left mouse button.

3.	Click on "Low-pass."

4.	Choose the "Frequency" above which you desire to attenuate all signals, utilizing the right mouse button in conjunction with the "Frequency" slider control.  When the control is all the way down, the setting will be 5 Hz, and when it is all the way up, the setting will be 19.999 KHz. (Useful settings will usually fall somewhere within the 3 KHz to 15 KHz range, depending on the source material and the goals of the audio restoration.)  If you desire finer frequency resolution, you may also use direct numeric entry of the value.

5.	Choose the Filter "Slope" which you desire.  Click on either 6 dB / Octave, 12 dB / Octave, or 18 dB / Octave.  The steeper the slope, the higher will be the degree of attenuation of all frequencies above the "Frequency" setting.

6.	If you desire to hear the results of your filter settings before creating a new "Destination" file, click on "preview."

7.	After a short delay, you will hear the effect of the settings that you have chosen.  (The system may seem to stutter if your computer is too slow to keep up in real-time with the algorithm.  This repeating pattern will not appear in the final Destination processing of the filter.)

8.	As the filter is running in either Preview mode or Run mode (Destination File Mode), you will see a dialog box which indicates the "% Done" of the filter algorithm on the selected portion of the Source Wave file.  Also, at the top of the Dialog box you will see indicated the "Total Samples to Process:"

9.	Keep adjusting the Frequency and Slope parameters, and testing the various settings using the "Preview" mode until you are satisfied with the results.

10.	When you are satisfied with a group of settings, you will be done with Preview mode.

11.	Click on Run, and the filter will process your Source Wave file through the filter algorithm, and create a Destination Wave file containing the output of the filter.

12.	When this process is complete, you will see the Destination File become highlighted in Yellow, at the same time that the Source File becomes unselected.


 Band Pass Filter

Band-pass filters are essentially a combination of a Low-pass filter and a High-pass filter. It attenuates both the high frequency and the low frequency portions of the audio spectrum. It is useful where the recording contains extraneous noise in the low frequency region such as rumble or thumps, and high frequency noise such as hiss.  This filter can also be very useful for improving the intelligibility of audio recordings, especially speech, by eliminating the unnecessary portion of the audio spectrum that is not used by speech frequencies to carry useful information to the listener.

Figure 34- The Band Pass Filter


Chebyshev Response and 4th Order Slope Added To Band Pass Filter

This is a digital simulation (IIR based) of a conventional analog Band-pass filter having a Butterworth or Chebyshev response when set to the steeper slope values. Band-pass filters are passive to frequencies within the Band-pass region, but they attenuate frequencies above and below the two corner frequencies.  Band-pass filters have both an upper and a lower corner frequency, and like the Low-pass and the High-pass filter, the corner frequencies are defined as the frequencies at which the signals either above the upper corner or below the lower corner are attenuated by 3 dB.  Four slopes are provided for the Band-pass Filter, just like the Low-pass and the High-pass.  They are 6 dB / Octave, 12 dB / Octave, and 18 dB / Octave and 24 dB / Octave. This filter can be very useful for improving the intelligibility of audio recordings, especially speech, by eliminating the unnecessary portion of the audio spectrum that is not generally occupied by vocal frequencies used to carry useful information to the listener.  The Forensics Menu "Brick Wall" filter has a much steeper version of this filter for dealing with extreme cases of out-of-band noise that needs to be eliminated.

Note:

The higher order (12, 18, & 24 dB / Octave) Band-pass filters are of the Butterworth or Chebyshev type depending on your choice.

Special effects can be produced with the Band-pass filter.  These special effects can be useful when producing movies or stage plays or shows and a particular sound producing device and its environment needs to be accurately reproduced through the "House" P. A. System.  Here are a few examples:

Simulation	Low Freq. Control	High Freq. Control	Slope
1930's Vintage Table Top Radio:	830 Hz	2000 Hz	18 dB / Octave
Modern cheap Table Top Radio:	265 Hz	6100 Hz	18 dB / Octave
Loud "Walkman" personal  stereo as heard by person nearby:	3650 Hz	9800 Hz	18 dB / Octave
Modern Stereo System as heard from the next room:	95 Hz	4100 Hz	12 dB / Octave
1950's Vintage Juke Box:	30 Hz	2700 Hz	12 dB / Octave
AM Transistor Pocket  Radio:	1395 Hz	2110 Hz	18 dB / Octave
Telephone Receiver sound from "off the hook":	2700 Hz	2895 Hz	18 dB / Octave
Night Club Band as heard from Parking Lot:	85 Hz	240  Hz	12 dB / Octave
Olde Acoustic Phonograph:	870 Hz	2390 Hz	18 dB / Octave
Public Address System at Outdoor Event:	300 Hz	3000 Hz	12 dB / Octave
Modern High End Audio  System:	15 Hz	19,999 Hz	6 dB / Octave
Bandpass Filter Response Limits:	5 Hz	19,999 Hz	-

You can create your own simulations of sound devices and acoustic environments through experimentation with the Band-pass filter parameters.  Using the above simulations, in conjunction with the FIVE/LIVE reverb, can further enhance various acoustical environments.  Once you discover the appropriate values, write them down for future reference.

Cascading this filter (using the MultiFilter) with others like the Virtual Valve Amplifier to add distortion, and the Reverb to add room acoustical effects can embellish these sound simulations.

The Band-pass filter can also be used as a tool to determine if any useful audio information exists in a particular portion of the audio spectrum; it becomes sort of an audible wave analyzer when used in this manner.  For more information on this mode of operation, refer to the "Using FIVE/LIVE as an Audio Waveform Analyzer" portion of the "How Do I" section of this manual or go to the section which explains the operation of the Spectrum Analyzer.

The following is a summary of the control parameters and range of adjustment provided for the Band-pass Filter:

•	Low Frequency: 5 - 19,999 Hz.

•	High Frequency: 5 - 19,999 Hz.

•	Filter Slopes: 6, 12, 18, & 24 dB / Octave.

•	Preview Mode Button: On / Off (The slider controls can be adjusted "live" when preview mode is on.)

•	Filter Type: Choice of Butterworth or Chebyshev

Note:

The frequency range of adjustment up to 19,999 Hz is only effective when utilizing a 44.1 KHz sampling rate.  At a 22.05 KHz sampling rate, the maximum effective frequency setting will be 10 KHz, and at an 11.025 KHz sampling rate, this value will drop to 5 KHz.


Band-pass Filter Operating Procedure (Tutorial)

1.	Highlight the portion of your Wave file on which you desire to apply the Band-pass filter.  (You may choose to highlight the entire file or any portion thereof.)

2.	Click on the Filter Menu with the left mouse button.

3.	Click on "Band-pass."

4.	Make an initial determination of what band of frequencies you desire to pass through the Band-pass filter.

5.	Utilizing the right mouse button in conjunction with the Low Frequency slider control, select the lower corner frequency of the range that you have chosen.  (The range for this control is 5 Hz to 19.999 KHz)

6.	Utilizing the right mouse button in conjunction with the High Frequency slider control, select the upper corner frequency of the range that you have chosen.  (The range for this control is 5 Hz to 19.999 KHz)  Note: If you desire finer frequency resolution for either the lower or the upper corner frequency, you may use direct numeric entry, instead of the slider controls.

7.	Choose the Filter "Slope" which you desire.  This slope will symmetrically effect both the upper and lower corner roll-off rates.  Click on either 6 dB / Octave, 12 dB / Octave, 18 dB / Octave or 24 dB / Octave.  The steeper the slope, the higher will be the degree of attenuation of all frequencies outside of the selected pass band range.

8.	If you desire to hear the results of your filter settings before creating a new "Destination" file, click on "preview."

9.	You will hear the effect of the settings that you have chosen, after a short delay.  (The system may repeat [stutter] if your computer is too slow to keep up in real time with the algorithm.  This repeating pattern will not appear in the final Destination processing of the filter.)

10.	As the filter is running in either preview mode or normal mode (Destination File Mode), you will see a dialog box that indicates the "% Done" of the filter algorithm on the selected portion of the Source Wave file.  Also, at the top of the Dialog box you will see indicated the "Total Samples to Process:"

11.	Keep adjusting the Low Frequency and High Frequency sliders as well as the Slope parameters until you achieve the effect you desire and are satisfied with the results.

12.	When you are satisfied with a group of settings, you will no longer need to invoke the Preview function.

13.	Click on Run, and the filter will process your Source Wave file through the Band-pass Filter algorithm, and create a Destination Wave file containing the output of the filter.

14.	When this process is complete, you will see the Destination File become highlighted in Yellow, at the same time that the Source File becomes unselected.

15.	Click on "Close"

Note 1:

If the low frequency control is set to a higher frequency than the high frequency control setting, a "no pass" filter will be created.  This is of little useful value, but is allowable by FIVE/LIVE.

Note 2:

You can select Chebyshev rather than Butterworth if you want steeper filter response at the expense of some ripple within the filter pass-band.


 High Pass Filter

A High-pass filter only passes signals that are above or "higher" than the corner frequency. It reduces the level of low frequency signals that are below the corner frequency.  The effect can be similar to turning down the bass control on a home stereo. This filter is very useful for reducing turntable rumble, muddiness, and any other extraneous low frequency noise in a recording.


Chebyshev Response and 4th Order Slope Added To High Pass Filter

This is a digital simulation (IIR based) of a conventional analog High-pass filter having a Butterworth or Chebyshev response (for the higher order slopes).  This filter attenuates frequencies below the High-pass corner frequency.  Just like the Band-pass and the Low-pass filter, this filter has four slopes available.  They are 6 dB / Octave, 12 dB / Octave, and 18 dB / Octave. This filter is very useful for reducing the effects of turntable rumble, or microphone seismic effects from creating "muddiness" on an audio recording.  It can also be used to eliminate any  offset that may have developed on a Wave file. To reduce turntable rumble, start with a setting of 60 Hz and 18 dB / Octave, and then adjust the parameters until you are satisfied.  This filter is also useful when selectively applied for reducing microphone "P" popping effects on the vocal track of multi-track recordings wherein an adequate windscreen had not been utilized in the session. Effective settings typically are 120 Hz with a slope of 18 dB / Octave (selectively applied to the event).

Figure 35- The High Pass Filter

Note:

The higher order (12,18,& 24 dB / Octave) High-pass filters are of the Butterworth or Chebyshev types, depending upon your selection.

The following is a summary of the control parameters and range of adjustment provided for the High-pass Filter:

•	Frequency:  5 - 19,999 Hz.

•	Slope:  6, 12, 18, 24 dB / Octave

•	Preview Mode Button: On / Off (The slider controls can be adjusted "live" when preview mode button is activated.)

•	Filter Type: Choice of Butterworth or Chebyshev

Note:

The frequency range of adjustment up to 19,999 Hz is only effective when utilizing a 44.1 KHz sampling rate.  At a 22.05 KHz sampling rate, the maximum effective frequency setting will be 10 KHz, and at an 11.025 KHz sampling rate, this value will drop to 5 KHz.


High-pass Filter Operating Procedure (Tutorial)

1.	Highlight the portion of your Wave file on which you desire to apply the High-pass filter.  (You may choose to highlight the entire file or any portion thereof).

2.	Click on the High Pass Filter. (Filter Menu)

3.	Choose the "Frequency" below which you desire to attenuate all signals, utilizing the right mouse button in conjunction with the "Frequency" slider control.  When the control is all the way down, the setting will be 5 Hz, and when it is all the way up, the setting will be 20 KHz.  (Useful settings will usually fall somewhere within the 15 Hz to 500 Hz range, depending on the goals of the audio restoration process.)  If you desire finer frequency resolution, you may also use direct numeric entry of the value.

4.	Choose the Filter "Slope" which you desire.  Click on either 6 dB / Octave, 12 dB / Octave, or 18 dB / Octave or 24 dB / Octave.  The steeper the slope, the higher will be the degree of attenuation of all frequencies below the "Frequency" setting.

5.	Click on Preview to Audition the results before processing.

6.	You will hear the effect of the settings that you have chosen, after a short delay.

7.	As the filter is running in either preview mode or normal mode (Destination File Mode), you will see a dialog box that indicates the "% Done" of the filter algorithm on the selected portion of the Source Wave file.  Also, at the top of the Dialog box you will see indicated the "Total Samples to Process:"

8.	Keep adjusting the Frequency and Slope parameters, and testing the various settings using the "Preview" mode until you are satisfied with the results.

9.	When you are satisfied, click on Run, and the filter will process your Source Wave file through the filter algorithm, and create a Destination Wave file containing the output of the filter.

10.	When this process is complete, you will see the Destination File become highlighted in Yellow, at the same time that the Source File becomes unselected.

Note:

For steeper filter response at the expense of passband ripple, choose Chebyshev rather than Butterworth response.


Removing DC-Offsets Using the High-pass Filter (Tutorial)

The High-pass Filter can be used to remove DC-Offsets from Wave files.  To do so, set the High-pass Filter to 20 Hz and 6 dB per octave, and run it on the Wave file needing correction.


 Notch Filter

A notch filter attenuates signals that are near its center frequency setting.  The degree to which it attenuates frequencies near the center frequency is determined by the bandwidth setting.  This filter is useful for removing 50 or 60 Hz hums from a recording. It is also useful for decreasing any sound system acoustic feedback that may be found on some live recordings.  A "Slot" filter is also provided within the Notch filter menu item for Forensics applications.  Multiple slots can be constructed using the FIVE/LIVE MultiFilter.

This is a digital simulation (IIR based) of a second order notch / slot filter.  It attenuates all frequencies near its center frequency setting.  The degree to which it attenuates frequencies adjacent to the center frequency is determined by the bandwidth setting.  This filter is useful for removing 50 or 60 Hz hum from a recording (or harmonics thereof).  It is also useful for decreasing any sound system acoustic feedback that may be found on some live recordings.  It can be used to attenuate the heterodyning "whistle" which is sometimes heard on AM broadcast radio reception.  Some audio restoration engineers also use this filter to remove some "Hiss" from recordings.  For this application, the filter's center frequency is set somewhere in the 8 to 12 KHz range, with a bandwidth of 0.25 Octave or less.  Experimentation is the only way to determine its effectiveness in minimizing "Hiss" from your particular source material.  Also, it is important to note that this method is not the most effective for "Hiss" removal.  Instead, consider using either the Continuous Noise filter or the Dynamic Noise filter.

Figure 36- The Notch Filter

The following is a summary of the control parameters and range of adjustment provided for the Notch Filter:

•	Center Frequency: 5 - 19,999 Hz

•	Bandwidth: 0.01 Octaves to 2.00 Octaves

•	Preview Mode Button: On / Off  (The slider controls can be adjusted "live" when preview mode is on.)

This filter incorporates the "slot filter" feature. Essentially, a slot filter produces the inverse response of a notch filter.  This allows you to hear or keep only the residual component of the original signal. This feature is useful for "tuning" the notch filter to the maximum level of noise, so that when you actually run the notch filter with the "slot filter" feature turned off, the noise left behind will be minimized.  The slot filter function is also useful in Forensics applications, wherein one is interested in isolating a very particular sound that exists in a very specific and narrow frequency band.  If multiple slots are required, use the MultiFilter with multiple notch/slot filters in the chain, or consider using the Harmonic Reject filter in "Keep Residue" mode, provided the slots, which are required, are harmonically related.

Note:

The frequency range of adjustment up to 19,999 Hz is only effective when utilizing a 44.1 KHz sampling rate.  At a 22.05 KHz sampling rate, the maximum effective frequency setting will be 10 KHz, and at an 11.025 KHz sampling rate, this value will drop to 5 KHz.


Notch Filter Procedure (Tutorial)

1.	Identify the frequency that you desire to reject from the Wave file recording.  The determination of the frequency of interest can be simplified by using the Band-pass filter as an Audible Spectrum Analyzer.

2.	Highlight the portion of your Wave file on which you desire to apply the Notch filter.  (You may choose to highlight the entire file or any portion thereof).

3.	Click on "Notch."

4.	Choose the Center Frequency based on the "problem" frequency that you have observed and desire to reject.  The Center Frequency range is from 5 Hz. to 19.999 KHz.  (The highest useful frequency is around 15 KHz).  If you desire the finest degree of frequency resolution possible, use direct numeric entry rather than the use of the slider controls.

5.	Choose the Bandwidth that you find to be the most effective in eliminating the desired frequency.  The bandwidth control is calibrated in Octaves, and has a range from 0.01 Octaves to 9.99 Octaves.  You should choose the smallest possible bandwidth that still accomplishes the job of rejecting the troublesome frequency.  Otherwise, you will start eliminating useful information from your recording.  Generally, useful ranges for Bandwidth will be in the 0.5 Octave to 0.1 Octave range.

6.	If you desire to hear the results of your filter settings before creating a new "Destination File, click on "preview."

7.	You will hear the effect of the settings that you have chosen, after a short delay.  (The system may repeat (stutter) if your computer is too slow to keep up in real-time with the algorithm.  This repeating pattern will not appear in the final Destination processing of the filter.)

8.	As the filter is running in either preview mode or normal mode (Destination File Mode), you will see a dialog box that indicates the "% Done" of the filter algorithm on the selected portion of the Source Wave file.  Also, at the top of the Dialog box you will see indicated the "Total Samples to Process:"

9.	Keep adjusting the Frequency and Slope parameters, and testing the various settings using the "Preview" mode until you are satisfied with the results.

10.	When you are satisfied with a group of settings, you will no longer need to use Preview mode.

11.	Click on Run, and the filter will process your Source Wave file through the filter algorithm, and create a Destination Wave file containing the output of the filter.

12.	When this process is complete, you will see the Destination File become highlighted in yellow, at the same time that the Source File becomes unselected.

13.	Click on "Close."

Note:

If only a small section(s) of your Wave file is in need of Notch filtering, as is often the case when acoustic feedback is encountered on a live recording, you can use "sync mode" and just filter the sector which contains the noise which your are attempting to reduce. "Sync mode" can be selected under the "View Menu."


Median Filter

The Median Filter can be used to substantially reduce "crackle" (small impulse noise) from a recording. Use a sample setting of 3 to 7 for this application.  A "weighting" control is also provided, which affects the "timbre" of the processed sound.

Figure 37- The Median Filter

There is no analog equivalent to the Median Filter.  This filter defines a window of samples, and for that window, determines which sample is the median value within the grouping (median meaning middle value). That value is the one that is passed along to the destination file, and then the window moves over 1 sample and re-evaluates the median, again passing the new median value to the destination file.  This filter is useful for improving the intelligibility of severely distorted signals and it is also useful for pulling signals out of a very poor signal-to-noise ratios situation (pulling signals out of the mud).  It is somewhat similar in sound performance to a high-order Low-pass filter.  (The median value of a string of sorted numbers is the one in the middle of the string.  In other words, if you have seven sorted numbers, the fourth number will be the median value).  The FIVE/LIVE Median Filter allows you to choose the number of samples over which the median value is determined by the algorithm.  The range is from 3 to 20 samples.  The higher this value, the higher the attenuation of high frequency signal and/or noise.  Also, the higher the setting, the longer the processing time required for the filter.  The most useful settings will generally be found to be from 3 to 7 samples.  Outside of that range, you may hear a significant degradation of the top end of your recording, depending on its bandwidth.  Even at 7 samples, you will notice some "fuzziness" inter-modulating with the upper end of the spectrum on some recordings. So always start with 3 to 5 samples when using the Median filter, and choose the smallest value that produces an effective de-crackling result.  It is also important to note that the higher the number of samples chosen, the longer it will take your computer to calculate the Median values to process your Wave file.  At some point, your system will likely process in non "real-time."


Weighting Function Added To Median Filter

A "Weighting" function is provided with this filter that shifts the position on the number line as to which value will be chosen as the "Median."  This feature is particularly useful in forensics audio application wherein extremely inarticulate speech needs clarification.  For example, the absence of consonant or sibilant sounds can render a recording undecipherable.  Recording bandwidth limitations can severely distort or eliminate the "hissing-consonants" which render speech understandable - - - a fact that has been known since the earliest ventures by Thomas Edison into the field of recorded sound.  With the use of the Median filter in conjunction with the weighting control, this problem can often be corrected.  The weighting control will essentially affect the "timbre" of the processed sound.  Trial and error will determine the best combination of "Samples" and "Weighting" to cure a particular forensics sound problem.

The following is a summary of the control parameters and range of adjustment provided for the Median Filter:

•	Samples: 3 - 20

•	Preview Mode Button: On / Off  (The slider control can be adjusted "live" when preview mode is on.)

•	Weighting: +/- 100 from the neutral value of 0 (0 provides no weighting to the Median value calculated by the filter)

Note:

De-Crackling will generally be best accomplished with a "Samples" setting around 3.  Intelligibility improvement of extremely distorted or garbled voice recordings will generally be accomplished with "Samples" settings anywhere between 5 through 17.  You will have to experiment to determine the best settings for solving a particular problem.


Median Filter Operating Procedure (Tutorial)

1.	Highlight the portion of your Wave file on which you desire to apply the Median Filter.

2.	Click on the Median Filter (Filter Menu)

3.	Choose the number of Samples over which you desire the median calculation to be performed.  The higher the number of samples chosen, the greater will be the attenuation of the higher frequency portion of the audio spectrum.  You can choose any integer value from 3 to 20 samples. (The most useful values will be found in the 3 to 7 samples range.)  The higher the chosen number of samples, the longer will be the process time requirement for the algorithm.  Changes in value are accomplished utilizing the slider control.

4.	If you desire to hear the results of your filter settings before creating a new "Destination" file, click on "preview."

5.	You will hear the effect of the calculation of the median value over the chosen number of "samples."

6.	As the filter is running in either preview mode or normal mode (Destination File Mode), you will see a dialog box that indicates the "% Done" of the filter algorithm on the selected portion of the Source Wave file.  Also, at the top of the Dialog box you will see indicated the "Total Samples to Process:"

7.	Keep adjusting the number of samples until you achieve your desired effect.

8.	When you are satisfied with a setting, click on Run, and the filter will process your source Wave file through the filter algorithm, and create a Destination Wave file containing the output of the filter.

9.	When this process is complete, you will see the Destination File become highlighted in Yellow, at the same time that the Source File becomes unselected.

10.	Click on "Close".

Note:

The "Weighting" control is used to effect the Timbre of the resultant sound of the Median filter.


 Averaging Filter

This filter sounds similar to that of a Low-pass filter, although it is somewhat more effective than a Low-pass filter in reducing not only "Hiss" but also "Crackle" from a sound source.  It is most effective on limited bandwidth sources such as old acoustic recordings made before 1925.  This filter is also useful for improving the intelligibility of highly garbled voice communications recordings.

Figure 38- The Averaging Filter

This is another filter that has no analog equivalent.  Its interface to the operator is similar to the Median Filter, with the difference being that instead of calculating the median value of a sample window to pass into the Destination workspace, the average value of a group of samples is passed through. You select the number of samples on which the average value is calculated with a slider control in its dialog box.  The greater the number of samples, the higher the degree of smoothing effect on the waveform. The higher that you set the degree of smoothing, the greater will be the loss in the higher end of the audio frequency spectrum.

The following is a summary of the control parameters and the range of adjustment provided for the Average Filter:

•	Samples: 2 - 100

•	Preview Mode Button: On / Off (The slider control can be adjusted "live" when preview mode is on.)


Average Filter Operating Procedure (Tutorial)

1.	Highlight the portion of your Wave file on which you desire to apply the Average Filter.  (You may choose to highlight the entire file or any portion thereof.)

2.	Click on the Filter Menu with the left mouse button.

3.	Click on "Average."

4.	Choose the number of Samples over which you desire the moving average calculation to be performed.  The higher the number of samples chosen, the greater will be the attenuation of the higher frequency portion of the audio spectrum.  You can choose any integer value from 2 to 100 samples.  The higher the number of samples chosen, the longer will be the processing time requirement for the algorithm. This is accomplished utilizing the slider control.

5.	If you desire to hear the results of your filter settings before creating a new "Destination" file, click on "preview."

6.	You will hear the effect of the averaging over the chosen value of "samples."

7.	As the filter is running in either preview mode or normal mode (Destination File Mode), you will see a dialog box that indicates the "% Done" of the filter algorithm on the selected portion of the Source Wave file.  Also, at the top of the Dialog box you will see indicated the "Total Samples to Process:"

8.	Keep adjusting the number of samples until you achieve your desired effect.

9.	When you are satisfied with a setting, you will no longer use the Preview Mode button.

10.	Click on Run, and the filter will process your source Wave file through the filter algorithm, and create a Destination Wave file containing the output of the filter.

11.	When this process is complete, you will see the Destination File become highlighted in Yellow, at the same time that the Source File becomes unselected.

12.	Click on "Close".


 Graphic Equalizer

The equalizer is a familiar filter that acts like an expanded tone control. The audio spectrum is broken into 10 bands, each being one octave wide. Each band's gain (volume) can be independently adjusted to achieve the desired audio result. This filter is useful for tonal shaping of the finished audio product or to enhance the bass or treble of a recording. It is also useful for improving the intelligibility of recordings or "Bringing Out" a particular instrument or vocal.

The Graphical Equalizer is the digital equivalent (which is IIR based) to the analog Graphical Equalizer found in many sound systems.  Its primary advantage is that it can be applied to a Wave file without having to resort to adding an analog step in your sound restoration process.  This results in decreased noise and distortion on your final product.  The equalizer has ten bands containing the following center frequencies:

31 Hz,  62 Hz,  125 Hz,  250 Hz,  500 Hz,  1 KHz,  2 KHz,  4 KHz,  8 KHz,  16 KHz

Figure 39- The Graphic Equalizer

The amplification and attenuation range for each band is + / - 12 dB.  An additional feature is provided to allow a "shelf" function (this is a "pull out" in the frequency response curve, sometimes referred to as the adding of a "zero" in order to form a pole-zero pair) below 31 Hz and above 16000 Hz.  A "Reset Levels" feature is provided.  Clicking on "Reset Levels" will return all of the Graphical Equalizer slider controls to their 0 dB position.  Since the graphic equalizer can add gain to your signal, a latching overload indicator is provided.  If, at any time during the processing of a file through the graphic equalizer, the output signal exceeds the dynamic range of the system, the indicator will turn red and latch until the filter is re-run.

Note 1:

The top equalizer band of 16,000 Hz is only effective when using a sample rate of 44.1 KHz or higher; it becomes ineffective at sampling rates of 22.05 KHz and 11.025 KHz.  The 8,000 Hz band will also be rendered ineffective when using a sampling rate of only 11.025 KHz.

Note 2:

The graphic equalizer controls can be adjusted "live" when the preview mode button is clicked.

Note 3:

Since the graphic equalizer can actually increase the gain of the system, it is possible to produce clipping which will result in unpleasant distortion products to appear in the Destination Workspace.  This usually occurs when excessively boosting the bass portion of the spectrum.  The overload indicator will change from green to red, and latch in that condition if there has been an overload. The latch will be reset, following a re-run of the algorithm, provided that the overload condition has been cleared by reducing gain in one or more bands.


Graphic Equalizer Operating Procedure (Tutorial)

1.	Click on the Filter Menu.

2.	Click on "Graphic Equalizer."

3.	Using your Mouse, adjust the frequency band slider control up or downwards as desired.  This can be accomplished either by directly pointing the cursor with the mouse and depressing the left mouse button to move the control, or by utilizing the up and down arrows associated with each of the 10 controls.

4.	When the Slider control for a particular band is in its center position, the band is neither being attenuated nor amplified.  Moving the slider upwards produces amplification of frequencies in the band up to 12 dB.  Moving the slider downwards produces attenuation of frequencies in the band of up to 12 dB.


Paragraphic Equalizer

This unique equalizer combines the flexibility of a parametric equalizer with the ease of use associated with a 10 band graphic.  The actual frequency response of the filter is graphically controlled and/or displayed.  Virtually any frequency domain transfer function that you can dream of can be created with this filter.  Many presets are provided to facilitate many unusual equalization situations.

Figure 40- The Paragraphic Equalizer

The Paragraphic Equalizer is a unique form of a parametric equalizer employing IIR techniques.  It combines the graphical representation of its transfer function (frequency response curve) with the versatility of a parametric equalizer.  Additionally, it provides you with up to 10 bands of equalization.  It differs from the graphic equalizer in that three parameters are adjustable for each band:

•	Frequency: (Hertz  - 10 to 20 KHz_

•	Amplitude: (attenuation or gain - +/- 20 dB)

•	Octaves: (Q or bandwidth - 0.05 to 3.0)

The following additional controls and displays are provided:

•	Number of filters: (1 to 10)

•	Output Gain: (+ / - 20 dB)

•	Reset Levels: (Resets Paragraphic to zero dB and factory default frequencies)

•	Overload indicator: (Illuminates when full-scale output {clipping} occurs.

The Parametric Equalizer displays its frequency domain transfer characteristic graphically. Therefore, it is referred to as the FIVE/LIVE "Paragraphic" equalizer.  It can be modified using your mouse by dragging the inflection point dots, and modifying the bandwidth using the octave control. You simply draw the shape of the response, which you desire, and the algorithm adjusts the parameters to match the response curve.  Each band is represented by a single square "dot" on the graph.  The "active" dot will be the larger one displayed.  That is the dot for which the parameters are being displayed numerically on the control panel.  You can use the mouse to drag any of the dots, which actually represents a frequency inflection point, wherever you wish.  If you want to sharpen or widen the response of any inflection point, use the octave control to achieve the desired curve for a highlighted dot.  It is often very useful to use the spectrum analyzer, found under the View menu, in conjunction with the Paragraphic Equalizer.  You will then be able to see the exact effect that you are imposing on the Wave file signal.

Under the Settings menu, you will find a number of useful audio restoration functions, including the RIAA curves.  Also, various inverse RIAA curves with a variety of turnover frequencies are available.  These features enable you to use a standard RIAA pre-amplifier to transfer acoustical and electrically recorded 78 RPM records to your hard drive, and re-compensate at another point in time, without having to purchase specialized hardware.  Also, of interest is the family of Fletcher-Munson Equal Loudness Contours at different sound pressure levels.  These can be used to compensate for the response of the human ear depending on the loudness level that you expect a particular audio piece to be auditioned.

Note 1:

As with all of the FIVE/LIVE filters, sample theorem dictates useful bandwidth for the algorithms.  The paragraphic equalizer will only have a useful bandwidth up to about 10 KHz with a 22.05 KHz sample rate, and about 5 KHz at 11.025 KHz.

Note 2:

Many of the Paragraphic equalizer presets such as the RIAA, Reverse RIAA and NAB curves are defined over the entire audio spectrum consisting of at least 20 Hz to 20 KHz band spread values.  Therefore, the use of any sampling rates less than 40 KHz will invalidate the accuracy of these curves.  We recommend using only 44.1 KHz or higher in order to properly realize these equalization curves.

Note 3:

Random white noise can be converted to pink noise by feeding it through the appropriate preset under the Paragraphic equalizer.


File Conversion

The File Conversion filter is not really a filter at all but a way to convert mono files to stereo and visa-versa. It can also be used to adjust the channel balance or reverse the channels of a stereo recording, or convert a mono source into a stereo file. It is useful in converting stereo recordings made out of phase (such as old vertically recorded acoustic discs) into a stereo or mono file that is compatible with modern systems.

Figure 41- The File Conversion Window

File Conversion includes two controls.  One set allows you to adjust the gain levels for each channel.  The second control, called Time-Offset, provides a means for azimuth correction, Forensics audio enhancement, and stereo simulation.  A final important use of the file conversion filter is to simply copy parts of the source file over the destination file. This is one way to revert back to the original source file (undo) following a bad filter application.

This will be the first step that you will perform in any sound restoration project.  This feature provides a number of file conversion options that can provide a certain degree of noise reduction in and of themselves. Gain adjustments can also be performed during the file conversion process.  The following File Conversion options are available:

 From Stereo to -

•	Mono (L + R): This option adds the Wave file Left Channel Input to its Right Channel Input before feeding it into the Destination workspace.

•	Mono (Left Only) & Mono (Right Only): These options choose only one of the two Wave file inputs to be used in the file conversion-processing step.

•	Mono (L - R): This option subtracts the Right Wave file signal from the Left signal before feeding it into the Destination workspace.

•	Stereo: This option maintains the Wave file in stereo through the file conversion-processing step.  It can be used to adjust the gains of the two signals if they are incorrect while maintaining their independence.

•	Stereo Reverse: This option will reverse the left and right channels during the Wave file conversion-processing step.  It can also be used to adjust the gains of the two signals at the same time, if they are incorrect.

•	L = L+R  R = L-R:  This option provides monophonic mix to the left channel, and provides the ambient signal from a stereo recording to the right channel.

From Mono to -

•	Mono:  This option merely provides a clone of the original Wave file.

•	Stereo: This option converts a monophonic single-track file into two single-track monophonic files.

Here are more details on the various File conversion options and their specific application:

•	Mono (L + R)

This is generally used to convert a lateral cut record (like a typical 78, or a monophonic LP) which is monophonic to start with, but which has been transferred to the hard drive with a stereo cartridge, and convert these two signals into one signal on which you will perform further processing.  The advantage of this simple conversion is that some of the noise content of the record will cancel out during this process, in particular, low end rumble, and even some higher frequency surface noise.  This process alone can provide up to 6 dB of signal-to-noise improvement (depending on the condition of your source) compared to the use of only one of the lateral groove walls (i.e. using the left only or the right only signal.)

Important Note:

It is advisable to set both gain controls to - 6 dB to avoid overloading of the Destination channel during this mixing process, unless your recording is extremely under-recorded to start with. Minus 6 dB is the default value for the two gain settings in the Mono (L + R) File Conversion feature.

•	Mono (Left Only) & Mono (Right Only)

Sometimes, 78-rpm laterals are worn unevenly due to years of improper tracking of the tone arm that played the particular record.  Therefore, it is sometimes useful to compare the Left Only groove wall with the Right Only groove wall to hear if that is the case.  If you hear a significant difference between one of the two groove walls, you should then compare the quieter of the two with the Mono (L + R) signal for comparison.  Choose the quietest of the three possibilities for your Destination file.

•	Mono (L - R)

This feature takes the algebraic difference between the left channel and the right channel audio signals and feeds it into the destination file.  It has three applications:

1.	If you have transferred vertical cut records such as cylinders or Edison Diamond Discs utilizing a stereophonic cartridge, and haven't previously extracted the vertical signal component from that signal, this feature will enable you to do so.  Just as you would have done with the laterals, it is useful to listen to the Left Only signal and compare it with the Right Only signal to make sure that no significant tracking damage has been done to the record over the years.  Choose the quieter of the two for subsequent comparison to the Mono (L - R) signal.  Generally you will find that the Mono (L - R) signal has the best signal- to-noise ratio for vertical (hill and dale) recordings.  Pathe´ (groove width modulated 78's) recordings should also be converted to monophonic utilizing the Mono (L - R) feature.

2.	This feature can also be used to compensate for gain imbalances between the left channel and the right channel of the analog equipment used to make the transfers into your computer system.  When listening to a lateral monophonic recording in Mono (L - R), you can adjust the gain control sliders until you hear a maximization of the noise and garble on the recording, and a minimization of the useful information content of the recording.  This will provide the best setting of the gain controls when you finally make the file transfer utilizing the Mono (A  + B) file conversion feature.

3.	It can be used to "cancel" a television or radio broadcast out of a surveillance recording.   The surveillance recording would have had to be recorded in stereo, with the surveillance signal on one track and also with a "reference" track containing the broadcast.  This technique will not completely cancel out the source radio or television source, but will attenuate it somewhat. Use the gain controls and preview to obtain the most effective degree of cancellation.

•	Stereo

This algorithm preserves a truly stereophonic Wave file in stereo through the sound editing and sound restoration process.  It allows you to adjust the channel levels to bring them more into balance if they are not balanced to begin with.

•	Stereo Reverse

This algorithm transposes the left and right channels from the source file before it is transferred into the destination file. This algorithm also allows you to, while reversing the channels, adjust the channel levels to bring them more into balance if they are not balanced to begin with.

•	Time Offset Feature / Azimuth Correction

The file conversion routine includes a "Time Offset" feature.  It is mounted horizontally on the file conversion control panel.  For normal file conversion operations, this control MUST be set to zero.  The time-offset algorithm provides you with the ability to retard or advance the timing between two stereo tracks.  This will work with a stereo-to-stereo conversion or a mono to stereo conversion.  The range of adjustment is + / - 20 milliseconds; when the control is set to its center (zero), the time offset between the two stereo channels will be zero milliseconds.

This feature has three applications:

1.	Analog Magnetic tape recording azimuth correction: When analog magnetic tapes are recorded or reproduced, the gap of the respective head (recording or playback) should ideally be perfectly normal (perpendicular) to the direction of the tape movement.  If, in either of the two mentioned processes, the respective head gap is off-normal (off-azimuth,) two types of signal degradation will occur.  The first phenomenon results in the loss of the high-end of the audio spectrum frequency response. The second effect produces a phase shift of one channel with respect to another thereby "smearing" a stereophonic image.  If you are reproducing a monophonically recorded cassette tape via a stereophonic playback machine, the effect of azimuth misalignment on high frequency loss can be somewhat improved by compensating using the "Time Offset" feature (the same applies when a monophonic half-track reel-to-reel tape is reproduced on a quarter track machine.)  To compensate for the effect of azimuth misalignment, adjust the "Time Offset" control until the best high frequency response is heard with your stereo system placed in monophonic playback mode while previewing.  If you are dealing with stereophonic source materials, it is hard to determine the correct phasing by merely adjusting the "Time Offset" for the best image.  But, if you place your stereo system in monophonic mode with a stereo tape source, and follow the same procedure just described (use the "Time Offset" feature to adjust for the optimal high frequency response), this will also correspond to proper left channel to right channel phasing, and will therefore produce the best stereo image.  Of course, you must place the system back in stereo mode after the file conversion has been completed in order to appreciate the results.

2.	Improving the Intelligibility of Forensics recordings:  It has been shown that audio signals, which are very difficult to discern, can be made more intelligible when the brain is presented with the same signal twice with a short time interval in-between.  The human brain processes the information that arrives at each ear independently by the left and right brain.  The information is then shared and compared between the left and right hemispheres. Comprehension of the information results from the interaction of both hemispheres communicating with one another.  When a delay is injected between the information heard by the left ear and the right ear, the intelligibility factor is improved.  This phenomenon was discovered by one of the British intelligence agencies (MI-5) in the late 1950's. The technique involves the use of stereophonic headphones (so that each ear is acoustically isolated from the other,) and an adjustable delay inserted between the two reproducers.  The Time Offset feature can be used for this application.  First, use the standard techniques for cleaning up the Forensics recording.  Then apply the monophonic signal to the file conversion algorithm using the "Time Offset" in preview mode with headsets to adjust for the best intelligibility.  This may improve your ability to transcribe conversations, which would otherwise be impossible to discern.  This technique does not work with loudspeaker reproduction.

3.	Stereo Simulation:  The "Time Offset" feature is one of several methods provided by FIVE/LIVE that will produce a stereophonic effect.  Merely start with a monophonic file, and convert it to a stereophonic file with some value of "Time Offset" applied.  Adjust the "Time Offset" control to produce the spatial effect that you desire while using preview.

•	Phase Inversion:

180-degree phase inversion can be added to either channel by checking the appropriate box.  This feature can be used to correct for out of phase stereo masters wherein the cutting head was connected incorrectly to the cutting head power amplifier.  Please note that if both boxes are checked, you are right back where you started.  You must only check one of the two boxes to correct for a phase-inverted channel.

Note:

The gain and the Time Offset controls on the file converter routines can be adjusted "live" after the preview mode button has been clicked.


 Cross Fade

The Cross-fade filter is used to join sections of different Wave files into a single Wave file. Rather than just abruptly ending one file and starting another, the Cross-fade filter will smoothly fade from one file to another. During the time that the files overlap, the destination file is gradually faded to silence, while the source file fades from silence to full volume. This filter is also available from the Edit menu as a paste function.

Note:

This menu item and the corresponding icon will only come alive after you have files in both the Source and Destination workspace.


Using the Filter Menu Cross-fader (Tutorial)

Warning:  This is not undo-able

1.	Open a Source File which you desire to be the segue song in a cross-fade sequence.

2.	Open a Destination File that you desire to be the first number in the cross-fade sequence.

3.	Highlight the Source File up to the end of the file.

4.	Highlight the Destination File segment at its ending where you desire the cross-fade to occur.

5.	Click on the Cross-fade feature under the Filter Menu.

6.	Set the Gain Controls as follows: (Default settings are correct)

•	File # 1 Level

•	Start = -100 dB

•	Stop = 0.00 dB

•	File #2 Level

•	Start = 0.00 dB

•	Stop = -100 dB

7.	Choose the Cross-fade timing that you desire.  Linear produces a pleasing result on most material.

8.	Click on "Do Cross-fade."

9.	The final results will reside in the Destination Workspace following processing. The entire Source File will be appended to the Destination file following the cross-fade sequence.


The Effects Menu

The Effects Menu houses a good portion of the audio enhancement tools located in FIVE/LIVE.  This is the place to head when you want to breathe new life into an old or poorly recorded file.


 Reverb

The Reverb effect is used to add a realistic room sound to a recording. The reverb is capable of simulating different size rooms, with different kinds of reflective surfaces and decay times. The reverb filter lets you control the overall room size, decay time, early reflection level, and mix between the original material and reverb sound.

Figure 42- The Reverb Effect Window

Reverb can be useful when dealing with recordings, which are completely "dead" as originally mastered.  As with the various other filters, the reverb effect can be applied globally or selectively (using sync mode) to a Wave file.  The reverb effect can also be used to convert a monophonic recording to a simulated stereophonic recording.  The following controls are provided on the FIVE/LIVE Reverb:

•	Room Size: (check box)

o	Small (Club)

o	Medium (Auditorium)

o	Large  (Concert Hall)

o	Very Large  (Stadium)

•	Reflections: (check box)

o	Bright: (Simulation of a very "hard" acoustical environment, as in a stone building)

o	Warm:  (Simulation of a typical auditorium or theater)

o	Dark:  (Simulation of a heavily draped auditorium)

•	Decay: Control Range 1 to 99 in relative units.

The decay control affects the dampening effect of the algorithm on the reverberated signal.  The higher this control is set, the longer the reverberation "dwell-time."  The lower that this control is set, the quicker will be the decay of the reverberated waveforms.

•	Output Mix: (Slider Control) Control Range: 0 to 100 in   percentage units

The Output mix determines the amount of the reverb effect that is fed into the system output.  When the control is set to zero (dry), there will be no reverb effect.  When the control is set to 99, there will only be the reverb effect, with the source signal bypassed.  Useful ranges of control are usually in the 5 to 25 range, but if you are looking for extreme effects, you can get them if desired.

•	Reverb Presets:

The Reverb is equipped with a number of descriptive presets.  This is a good place to start from when using the reverb effect.  Choose the desired acoustical environment (which can be selected and previewed "on-the-fly").  After you have found something close to the sound you desire, revert to the various controls to fine "tweak’" the reverb for the exact sound you are looking for.


Virtual Valve Amplifier

The Virtual-Valve Amplifier is a computer simulation of a number of vacuum tube amplifier circuits.  (Valve is the British term for electron tube.  We call it the "Virtual Valve Amplifier, because that sounds cooler than "Virtual Tube Amplifier.")  Its effect is to add "tube-warmth" to the sound of a recording.  This is sometimes desirable to apply to DDD (purely digital) recordings.  It can also be used to add subtle harmonics to very old recordings.  A harmonic exciter is also included with the Virtual Valve Amplifier.  It is important to note that the Virtual Valve amplifier is using real tube circuits, and real tube non-linear device characteristics to produce its effect.  The wide range of adjustability of this algorithm will allow you to create an amplifier that runs the gamut in sonic performance from "grit-guitar" to "high-end audiophile."

Figure 43- The Virtual Valve Amplifier

The Virtual Valve Amplifier (VVA) produces a variety of sounds associated with valve (electron tube) based amplifiers.  The effects run the range from a subtle "tube warmth" sound to extreme effects like "guitar amplifier overload" or "fuzz box."  The FIVE/LIVE VVA accomplishes these effects through the use of actual electron tube circuits, which are simulated by your computer.  The electronic models of the various tube amplifier circuits have been derived from the "large-signal" transfer functions of the various tubes and output transformers you can choose from. This data has been derived from extensive bench measurements of tube amplifier circuits under varying operating conditions.  As such, the effects will sound literally as would be heard if you were to process a signal through a physical electron tube amplifier.

However, with the VVA, you have a great deal more control over the various sounds that can be produced, since controls, which are not normally found on electron tube equipment, have been provided.  Parameters such as "Operating Point" (sometimes referred to as "Q" point by engineers) are usually fixed by the amplifier manufacturer.  "Drive" is determined by how loud you play a "physical" amplifier, but with the VVA, the output level remains constant independent of drive due to an internal gain compensation algorithm.  The following is a listing of the controls that are provided on the FIVE/LIVE VVA:

•	Drive Slider: 1 to 100

This control effects the degree of modulation applied to a given tube amplifier circuit and centered about the operating point.  The higher the drive level setting, the greater will be the production of predominantly even order harmonics due to the circuits asymmetrical non-linearity.  As a result, there will be more "effect" as this control is increased.  Also, the "depth" of the effect is determined in part by the degree of drive applied.

•	Operating Point (or Harmonic Control) Slider: -100 to zero (in the middle) to +100

•	VVA Mode:

The operating point control performs two different functions, depending on the Tube Type / Configuration selected.  When a triode or class A amplifier is chosen, it sets the operating point for the particular tube or amplifier configuration that you have chosen.  Operating point also determines the devices bias value at zero signal input.  The distributions of harmonics, which are introduced into the output of the amplifier, are determined to a large degree by the location of the operating point.  When the control is set to + 100, (all the way up) the devices are operating close to "saturation," and when the control is set to – 100 (down), the devices are operating close to "cutoff."  The non-linearity distribution is different near cutoff as compared to operation near saturation.  You can use the control to achieve variations in the desired "tube effect."  Most audio preamplifier tubes such as the 12AX7 are the most linear in the middle of their dynamic operating curve (control set to the middle position.)

•	Harmonic Exciter Mode:

When the system is placed into harmonic "Exciter" mode, the operating point control reverts to a "Harmonics Control" which varies the distribution of harmonics that are produced by the VVA.  The Harmonic Exciter is designed to provide the following audio enhancements:

A.	Synthesize the upper register harmonics that may have become lost through "generation loss" or due to the poor frequency response of the master recording.

B.	Add "presence" to a vocal recording.

C.	Create a more "up-front" sound on any modern recording.

When the control is set to +100, both even and odd harmonics are produced.  When the control is set to –100, only the first 3 to 4 even harmonics of the fundamental are produced.  Settings in between will produce varying combinations of the two extreme settings.  The system is placed into harmonic Exciter mode by checking the "Exciter" box listed under Tube Type / Configuration, located at the bottom of the VVA window.  The magnitude of the inserted Exciter effect is controlled by the "Mix" control.

•	Operating Point Indicator:

Vertical undulations are graphically presented proportional to signal level, drive, and operating point. The Operating point indicator will have a black background in standby modes of operation and a blue background with vertical undulations appearing in any of the operational modes of the VVA.  A yellow horizontal line during operation indicates the operating point center value. Also, the dynamic operating mid-point reference is indicated by a fixed white line on the indicator. The magnitude of the drive level to the amplifier is indicated by yellow undulations plus and minus about the operation point.  So both the effects of the drive and the operating point slider are indicated on the same display, for convenience.

•	Detail: 0 to 100

The detail control allows you to control the sensitivity of the VVA to the more delicate nuances of the musical material presented and processed.  The higher the setting, the greater the effect will be on the material.

•	Mix: 1 to 100

The Mix control affects the degree of VVA signal, which is re-inserted into the signal path.  At its maximum setting of 100 (wet), the dominant signal pathway is exclusively through the VVA, and when the control is set to 0 (dry), only the non-processed signal is fed through the system.  You can choose any level in between which appeals to your taste.

•	Range Checkbox:    -Sweet     -Warm     -Full Range

The Range control effects the spectral distribution of the harmonic by-products, which are passed through to the systems output.  Its most desirable setting is very much a function of the musical material, which is being processed, and the desired tube sound.

•	Advanced Controls Checkbox: On/Off

This enables the more advanced controls of the VVA, if desired.  If this control is not checked, defaults values will be chosen for some of the control settings, tube types, amplifier configuration, operating point and detail controls.  Although all of those settings are preset, you will still have control over the VVA Drive and Mix settings.

•	Bypass: On/Off

This control allows you to quickly compare the effects of the processed signal produced by the VVA to the unprocessed signal, while the program is in "Preview" mode.

•	Settings:  Listing

The VVA has a list of pre-sets, which will be a valuable starting point from which to fine tweak the adjustment controls to your desired taste.  These presets are somewhat descriptive to help you in making a choice.  The choices can be changed in real-time while running the program in Preview mode, so that you may compare the various presets.

•	Tube Type Checkbox: Checkboxes for the following Valves (tubes) or circuit configurations:

A.	Triode (12AX7)  - This configuration incorporates this high-mu dual triode into a typical RC coupled class A audio pre-amplifier configuration.  This tube was chosen, because it had been and still is the industry standard pre-amplifier valve.  It has a relatively flat linear operating region in the middle of its dynamic operating range, producing relatively lower levels of distortion compared to some of the other devices offered in the VVA.  But, by moving the Operating Point to either the saturation of cutoff extreme, more "tube-warmth" effect can be produced by this device.  This is the same device as the European ECC83.

B.	Triode (12AT7)  - This amplifier configuration utilizes the same type of RC coupled pre-amplifier circuit described above, only using a 12AT7 high-mu dual triode.  The primary difference is that the 12AT7 was designed primarily for RF mixing applications.  As a result, it has a large degree of non-linearity throughout its entire dynamic operating range, including the middle.  As a result, you will be able to obtain a higher level of even harmonic distortion (the pleasing distortion) in which to add back into the signal path of the VVA.  This is the same device as the European ECC81.

C.	Triode (12AU7)  - This amplifier configuration is simulating the driver / phase inverter stage of a push-pull power amplifier.  It utilizes the 12AU7 medium-mu dual triode, and, like the previously described circuits, is biased class A and is RC coupled.  This device also has a significant non-linearity in the middle of its dynamic operating curve. (In power amplifiers, some of this non-linearity is removed via the use of negative feedback, and decreasing the mix control level on the VVA simulates this phenomenon).

D.	Pentode (6EJ7)  - This single stage, high-gain microphone amplifier configuration utilizes a sharp-cutoff pentode.  It can produce a very pleasant "tube-warmth" effect when the operating point is properly set.  This device is the same as the European EF183.

E.	2 Stage Class A  - This is an 8 Watt class A power amplifier, consisting of a 12AU7 medium-mu triode driving a single 6L6GC beam power pentode audio output valve.  Its effects are distinctive due to convolution of the non-linearity of the triode interacting with those of the pentode, with both devices operating in class-A mode.  The 6L6GC is similar in performance to the industrial 5881, and also the European KT-66.

F.	2 Stage Class AB  - This is a 25 Watt class AB power amplifier, consisting of a 12AU7 phase inverter / driver, pushing a pair of 6L6GC beam power pentodes.  Because the circuit is push pull, the output devices produce a more symmetrical and reduced even-order distortion characteristic distribution.  The operating point is fixed at the factory, and cannot be adjusted for this amplifier configuration.

G.	2A3 Push-Pull –The 2A3 is what some people refer to as a "retro – triode."  It was invented in the 1930’s, had a directly heated cathode, and produced a high power output at its time of development. It was often found used in theatrical applications and public address systems. The "Push-Pull 2A3" VVA setting uses the 2A3 triode implemented in a "push-pull" class AB1 power amplifier circuit designed to produce 15 Watts of output power.  This configuration exhibits a more linear output transfer characteristic compared to its Pentode push-pull counterpart. We have included the 2A3 tube in this particular configuration in the Diamond Cut VVA because a musician friend of ours (Les Paul) recommended that we do so because of its unique characteristics.  He explained to us that he used a push-pull pair of these devices as the power amplifier to "cut" all of the records that he released from his own home studios.  The reason that he used these was the extremely clean sound that they produced.  The particular devices that we used to create the 2A3 VVA models were of the "dual – plate" variety.  The devices used were taken from new (unused) but old stock and were manufactured for the military by RCA Victor in 1953.

H.	2A3 Single-Ended –This is a single ended class A power amplifier implemented using the 2A3 power triode.  It exhibits reasonably good linearity and about 4 Watts of audio in a "single-ended" class-A configuration.  Its dominant distortion products are "evens." This is the only power triode in the Diamond Cut VVA suite of tubes.

I.	Exciter  - This check box enables the Harmonic Exciter feature of the  FIVE/LIVE VVA.  The exciter uses a simulation of a vacuum tube rectifier (6X4) to produce harmonics.  Asymmetry between the positive and negative going transfer function establishes the relationship between the degree of even and odd harmonics produced. For more details on its performance, please refer to the Harmonic Exciter description under the Operating Point Control description.

J.	Transformer Class AB  - This check-box enables a push-pull, transformer coupled, 6L6GC based, class AB, 20 Watt power amplifier having a 12AU7 based driver / phase inverter stage.  It produces a distortion dominated by odd order components.

Just like the other filters and effects; the VVA is equipped with a set of descriptive presets.  This is always a good place to start from when using the VVA.  After you have found the preset, which most closely resembles the sound you are looking to achieve, you can go back a fine "tweak" the control more precisely.  After you have found a group of settings that you would like to keep, use the Save Settings feature to give it a name so that you can recall it in the future.


Dynamics Processor

The dynamics processor provides you with the ability to control the dynamic signal content of the audio envelope of a Wave file.  Included are compression, downward expansion, noise gating, and de-essing.

Figure 44- The Dynamics Processor

The Dynamics Processor provides you with three functions related to the control of the dynamic range of an audio signal.  The functions are as follows:

•	Expander

This system is a downward expander.  When the signal is below the threshold setting, the dynamic range of the signal is increased depending on the value of the ratio setting.  In other words, the incremental attenuation of the Wave file signal is proportional to the ratio setting below the threshold value.  The higher one sets the value of this ratio, the greater will be the degree of downward expansion applied to the signal.  Signals above the threshold value are passed through the system with no processing applied.  When the ratio control is set to its maximum value (control set all the way up), the system will behave like a Noise Gate.  When more modest values of the ratio control are used, the system can produce some improvement in dynamic range and the signal-to-noise ratio of a Wave file.  The Expander has the following controls available:

A.	Expander/Gate Checkbox: On/Off

Checking this box will enable or disable the expander/gate function of the Dynamics Processor.

B.	Threshold:  -50 dB (control down) to 0.00 dB (control up)

This control establishes the signal level below which the expander performs its process on the Wave file signal.

C.	Ratio: 1.00 (control down) to 29.99 (control up)

This control determines the degree of downward expansion applied to the Wave file for signals that are below the threshold value setting.  The higher the number chosen, the greater will be the signal expansion effect.

D.	Expander bar-graph: Horizontal meter indicating from 0 to –40 dB.

This meter indicates the actual value of downward compression in dB, which is being applied to the Wave file signal.

E.	Attack: 199.9 mSec to 0.1 mSec

This control is used for both the expander and the compressor functions of the dynamics processor.  It determines the time constant associated with the onset (delay) of any of the dynamic processor effects.

F.	Release:  4.0 Seconds to 0.05 Seconds

This control is also used for both the expander and the compressor function of the dynamics processor.  Its setting determines the delay time associated with the decay of the particular process chosen.

•	Compressor

This system is an upward compressor.  When a Wave file signal is above the threshold setting, the dynamic range of the signal is decreased, the degree of which depends on the value of the ratio setting.  In other words, the incremental attenuation of the signal is proportional to the ratio setting when it is above the threshold value.  The higher that one sets the ratio value, the greater will be the degree of compression. When this value is set to its maximum, the system will behave like a limiter.  Signals below the threshold value are passed through the system with no processing applied.  When the ratio control is set to its maximum value (control set all the way up), the system will produce the largest degree of compression. The Compressor has the following controls available:

A.	Threshold: -50 dB (control down) to 0.00 dB (control up)

This control is similar to the threshold control for the expander, but establishes the signal level above which the compressor performs its process on the Wave file signal.

B.	Ratio: 1.00 (control down) to 29.99 (control up)

This control determines the degree of compression, which is applied to the Wave file for signals that are below the threshold value setting.  The higher the number that is chosen, the greater will be the effect on the signal.

C.	Expander bar graph: Horizontal meter indicating from to  +40 dB.

This meter indicates the actual value of compression in dB that is being applied to the Wave file signal.

D.	Attack: 199.9 mSec to 0.1 mSec

This control is used for both the expander and the compressor functions of the dynamics processor.  It determines the time constant associated with the onset (delay) of any of the dynamic processor effects.

E.	Release:  4.0 Seconds to 0.05 Seconds

This control is also used for both the expander and the compressor function of the dynamics processor.  Its setting determines the delay time associated with the decay of the particular process chosen.

Important Note:  The Punch and Crunch Filter provides a somewhat different approach to compression and expansion.


De-Esser Added To Dynamics Processor

A de-esser is a form of compressor, which is only reactive to the frequencies associated with the pronunciation of the letter "s" (ess.)  It is necessary to perform this function on over-modulated signals in the "s" frequency range that is adjustable from 1,000 Hz to 10,000 Hz.  This occurs due to poor mike technique, a poor initial mix, improper mike channel equalization, or insufficient "padding" of the mike input circuit during the recording session.   When the frequencies in the sensitive band are detected and are above the threshold setting, compression will be applied to the degree determined by the compressor ratio control.  The attack and release controls are not active when the compressor is in "De-esser" mode.  To place the compressor in the De-esser mode, click on the box by the same name.

One global control is provided in addition to all of those mentioned above.  The output gain allows you to correct for overall effects (attenuation or gain) that any of the dynamic processor functions may have on the overall output signal level.  Presets have also been provided to get you started with reasonable setup parameters for the various dynamic processor functions.

•	Automatic Level Control (ALC or AGC)

The Dynamics Processor includes an automatic level control feature (ALC).  Sometimes, these algorithms or systems are referred to as automatic gain controls or AGC’s.  This feature provides upward expansion of signals below the threshold line and downward compression of signals above the same threshold.  This feature is useful in Forensics applications where there is a large variation in signal levels between several different parties that may be communicating with one another.  It is also useful for the broadcast of live sporting events (if you have the LIVE version of the product) in which the crowd reaction is of interest when the announcer is not speaking. This feature is activated simply by clicking on the "ALC" box in the Dynamics Processor.  The threshold, attack, and release controls are still active when this function in invoked.

Reverse File

The Reverse file feature does just that - - - it converts a Wave file so that it will play in reverse.  This has several uses:

1.	Sometimes, it is beneficial for removing stubborn ticks or pops with the impulse filter.  By running a reversed file through the impulse filter, sometimes clicks that were otherwise too difficult to detect, may be found and removed.  Of course, when the process has been completed, you must re-reverse the file again, so that it may be heard in the normal forward direction.

2.	The Reverse File feature may be used for reversing metal stamper recordings that have been transferred on standard turntables that were not capable of running in reverse.  Simply record the metal stamper in reverse (using the appropriate bi-radial stylus), and then run the "Reverse File" feature to correct the transfer for forward playback.

3.	It provides an interesting effect.  Of course, it can also be used to make sure that there are no demonic subliminal messages recorded in reverse on the music that your children love to listen to!


Channel Blender

The channel blender provides the ability to reduce muddy bass from vinyl recordings, decrease the "ping-pong" effect from early Stereophonic recordings, minimize Multi-path distortion from FM stereophonic broadcast recordings and even remove vocals from stereo recordings.

Figure 45- The Channel Blender

The Channel Blender serves 4 purposes:

1.	In the early days of stereo, channel separation was the rage.  Sound engineers often literally segregated the recording artists into separate recording studios in order to maximize the channel separation.  This later became know as the "Ping-Pong" effect.  In other words, with stereo separation, if a little was good, and more was better, than too much was thought to be just enough.  The Channel Blender can be used to reduce the extreme stereo separation found on some of these early stereophonic recordings, restoring them to a more natural sound.

2.	Rumble on Vinyl recordings is dominated by the vertical displacement component of the master recording and playback stylus.  Since bass is acoustically non-directional below about a hundred Hertz, this rumble can be reduced by summing the signals and then adding them back into the main stereophonic channels below a certain crossover frequency.  The Blend to Mono feature performs this function when it and the "below" function are checked. This can add clarity and improved bass definition to vinyl recordings, which sound muddy due to excessive rumble.  And keep in mind that rumble is not just a product of the turntable from which you are playing a record, but also involves the system which mastered it in the first place.  So, even though you may have a very expensive turntable, you will still encounter recordings that are laden with rumble. The recommended frequency for this feature is around 125 Hertz with the "below" box checked.  But experiment to determine the best results for the material that you are dealing with.

3.	FM stereo multi-path distortion, when it occurs, is dominant in the last two octaves of the audio spectrum.  By placing the channel blender in Blend to Mono above the corner frequency setting, you can reduce this distortion with a tradeoff of channel separation at the upper end of the audio spectrum.  Try corner frequency settings starting at around 5 KHz with the "above" box checked.

4.	Lastly, ambience can be enhanced on a stereo recording by phase inverting one of the channels summing by the L-R rather than the L+R information back into the main signal path. This is accomplished by phase inverting one of the two channels.

The Channel Blender has the following unique controls:

•	Left and Right Channel Blend controls: These two controls take the summed or differenced signal and adds it back into the respective left and / or right channels.  When these controls are set to 0, there is no blending effect.  At a setting of 100, the blending is maximized.  Two check boxes are located in the Left and Right blend control panels.  This produces a 180-degree phase inversion of either of the channels before the summation takes place.  Therefore, you can blend in L+R (with the phase inversion boxes not checked) or you can blend in the L-R signal (with ONE of the two Invert Phase boxes checked.)  The L-R signal contains the ambience information on most stereophonic recordings.  If both Invert Phase boxes are checked, the signal reverts back to L+R, so if ambience enhancement is desired, only check one box.

•	Blend to Mono Checkbox: This checkbox sums the signal to monophonic above or below the indicated frequency. You can select a crossover frequency anywhere between 10 and 10,000 Hertz.

•	Above: This blends to mono all frequencies above the corner frequency setting. This is used to reduce multi-path distortion from FM broadcasts.

•	Below: This blends to mono all frequencies below the corner frequency setting.  This is used to reduce rumble and muddy bass on vinyl stereophonic recordings.


Punch and Crunch

Punch & Crunch is a multi-channel dynamic compressor or expander.  It allows you to modify the dynamics of a piece of audio in a very natural sounding manner.  In some cases, you may want to restore lost transient response by applying the expander mode, but in some other cases you may want to compress the signal in order to produce more "broadcast presence."

Punch and Crunch is a four band dynamic expander (Punch) and compressor (Crunch) and automatic level control (ALC).  It is useful for a number of applications such as the following:

1.	Adding dynamic range or "Punch" back into severely compressed radio broadcast or vinyl recordings.

2.	Adding "dial presence" or "Crunch" (compression) to radio broadcasts, without suffering the "pumping" effect found in conventional wide-band dynamic compressors.

3.	Decreasing the dynamic range of classical music so that it can be more "listen-able" in restaurant or automotive environments by applying the compressor function.

4.	Improving the signal-to-noise ratio and dynamic range of old 78-RPM recordings by using Expander mode.

5.	Improving the intelligibility of forensics recordings.

6.	Special Effects creation.

7.	Etc.

Figure 46- Punch and Crunch

It works by breaking the audio spectrum into four separate bands.  Each band is independently expanded or compressed when its signal exceeds the graphical display of its particular threshold line.  The degree to which the bands are expanded or compressed is modified by using the ratio control.  The actual compression or expansion of any particular band is shown by horizontal bar graphs for each band that are calibrated in dB.  The bands are broken into the following "buckets":

•	Band 1: 0 to 125 Hz

•	Band 2: 125 Hz to 900 Hz

•	Band 3: 900 Hz to 4,000 Hz

•	Band 4: 4,000 Hz to 20,000 Hz

The following controls are provided on Punch and Crunch:

•	Graphical Display of the four bands.  Each band is represented on this graph and the incoming signal present on each band will modulate the vertical displacement of each band.  Threshold for each band can be dragged with the left mouse button to the desired position.  When the threshold is dragged all the way to the top, that band will have no dynamic compression or expansion effect.  When a band is dragged all the way to the bottom, it will have a maximum compression or expansion effect.

•	Graphical Display of the Expansion or Compression of each band.  This graph is horizontally modulated and located beneath each of the four bands.  It is calibrated in dB.  It will tell you the amount of compression or expansion being applied to its associated band.

•	Ratio:  This controls the degree of compression or expansion applied to the system.  When the system is operating in compression mode, you can choose up to 30:1 compression.  When the system is operating in expansion mode, you can choose up to 15:1 expansion.

•	Attack:  This determines the time constant associated with the onset of this effect and is calibrated in mSec.

•	Release: This determines the time constant associated with the decay time for this effect and is calibrated in Seconds.

•	Output Level:  This allows you to adjust the output level of the system.  Use this in association with the Overload indicator to minimize clipping distortion.

•	Mode:  This allows you to choose either Expansion (Punch) or Compression (Crunch) modes of operation.  Additionally, you can choose ALC (Automatic Level Control) mode, which is often useful in Forensics Audio applications. It can be used to improve the intelligibility of speech having wide variations in signal level resultant from multiple persons speaking simultaneously on a recording.


New Punch and Crunch Narrowband mode: Ideal For Speech Processing

This shifts the location of all of the crossover frequencies closer together on the audio spectrum.  When this is used in conjunction with the expander mode, it is useful as a noise reduction and speech enhancement tool in Forensics applications such as cleaning up noisy telephone conversations.  Before using Narrowband mode for recorded speech, it is recommended that you first process your signal through one of the speech filters (typically 300 to 3,000 Hz). In Narrowband mode, the spectrum is broken down into the following "buckets":

•	Band 1: 0 to 400 Hz

•	Band 2: 400 Hz to 800 Hz

•	Band 3: 800 Hz to 1,600 Hz

•	Band 4: 1,6000 Hz to 20,000 Hz


 Change Speed

The Change Speed Filter is designed for use in several applications:

•	It can be used to correct the speed of a recording.

•	It can be used for fractional speed mastering.  This feature is important in any of the following three instances:

1.	You have available a 45 RPM turntable but do not own a 78 or an 80 RPM turntable.

2.	You want to play 80 RPM records, but only have a 78.2 RPM turntable with no

3.	Your turntable will play all speeds, but the record you are attempting to transfer is so warped that the stylus skips off the record due to the vertical undulations.

•	It can make speech transcription easier by slowing it down to the rate at which you can scribe.  Also refer to the Stretch & Squish filter for this functionality.  Note:  The Change Speed tool will alter the pitch of the audio, while the Stretch and Squish Tool will lengthen or shorten an audio file without changing the pitch.

•	It can be used to produce interesting special effects.

Figure 47- Change Speed

The following is a summary of the control parameters and the range of adjustment provided for the Change Speed filter:

•	Starting Pitch Control: -50% to +100%

•	Ending Pitch Control:  -50% to +100%

•	Display Pitch Range (3 ranges):

1.	+/-1%

2.	+/-10%

3.	+100 / -50%

•	Shape (Pitch vs. Time):

Straight Line (2 Green Cursors)

Curved Line  (4 Green Cursors)

•	Graphical linear or curvilinear Pitch inflection points (green square cursors on graph)

The Graph shows you how you have programmed the speed to change as a function of the selected Wave file time axis.  You can use the mouse to drag the two green cursors to establish the time relationship that you desire.  Often, a flat line is appropriate; however, sometimes the speed of the cutting lathe would slow down towards the end of the recording. The reason this occurred is that some of the early recording lathes used wind up mechanical motors with governors rather than electrical hysteretic synchronous motors.  To correct this defect, a pitch decrease (negative pitch slope) is necessary towards the end of the recording. When the curve shape is selected, two additional green cursors appear. The new green cursors can be moved both vertically and horizontally allowing you to create numerous curvilinear pitch vs. time relationships.  Curvilinear correction is useful for fixing the speed of non-capstan based tape recordings that have been transferred using a capstan-based machine.   It can also be used to create interesting special effects when used in conjunction with the looped preview or looped play mode.


New Stretch and Squish Filter Provides Time Compression/Expansion

The Stretch and Squish filter is a variation on the theme of the Change Speed filter.  It allows you to change the speed of a piece of audio while maintaining its pitch.  You can either slow down the beat or cadence of a piece (Stretch) or speed it up (Squish).

Figure 48- Stretch and Squish

Stretch & Squish is actually a special case of the speed change filter.  It has the ability to maintain a relatively constant pitch while modifying the cadence or beat interval of the Wave file that is being processed.  Its primary purpose is for Forensics applications in which a spoken word recording needs to be slowed down for transcription to the written word.  However, it can also be used to learn a fast guitar riff by slowing it down, or for other special effect situations.  It has a range of adjustability that will allow you to half the cadence in "Stretch" mode which means that it is slowed down.  Also, it has the ability to double the cadence of a Wave file in "Squish" mode.  Stretching or Squishing a Wave file is accomplished with the Stretch control.  If you set the Stretch control to 1.000, the file will play or preview at normal speed.  However, if you decrease the number to its lower limit of 0.5000, you can double the speed of the recording.  If you increase the number to its upper limit of 2.000, it will now play at half speed. Also, a Size control is provided which allows you to vary the update rate of the algorithm.  This control affects the overall sound quality of the filter, which is subjective in nature and left up to the user. The following controls are provided on the Stretch & Squish Filter:

•	Ratio:

0.500 to 1.000 speeds up the file and represents "Squish" mode

1.000 to 2.000 slows down the file and represents "Stretch" mode

•	Size Control:

20-240 (mSec)

a.	20-100 (Only useful for special effects)

b.	100-240 (Useful for Speech and Music without significant distortion)

•	Overlap Control:  0.2 to 0.6  (Determines the normalized degree of frame overlap and must be adjusted qualitatively.  The best sound will typically be found between .3 and .5)

Note:

This filter produces digital artifacts.  Artifacts are normal for this type of mathematical manipulation when applied to a Wave file.  The greater the amount of "Stretch" or "Squish" applied, the greater will be the degree of artifact generated by the routine.  Quality will decrease as you increase the amount of compression or expansion.


Filter Sweeper

The Filter Sweeper is a novel system providing a variable frequency response versus time capability.  The sweeper can be applied to an entire Wave file or any portion by using our selective filtering capability.  Three filter types are available in the Filter Sweeper including a sweepable Low-Pass, High-Pass, and Notch.  Besides producing interesting special effects, the Filter Sweeper has a number of very useful audio restoration applications.  Here are two examples.

The hum frequency of a recording sometimes changes as a function of time.  This audio defect may be due to the fact that the original analog tape recorder was battery powered and as the battery became weaker during the recording process, its speed decreased.  This mechanism will have caused the resultant hum frequency to increase versus time when played back later on a constant speed system.  To correct this defect, you can apply the Filter Sweeper set for operation in "Notch" mode. Set the "Start Frequency" to the appropriate lowest hum frequency value of the defective Wave file, and the set the "Stop Frequency" to the highest value of hum frequency.  These two values of frequency can be determined by using the Spectrum Analyzer while using the "looped play" mode during the appropriate sections (beginning and end) of the defective Wave file.  The "Sweep Type" will need to be determined empirically, because the appropriate compensation curve will depend on the type of capstan servo motor regulator that was used in the offending tape recorder in conjunction with the type of batteries that had been employed.

Figure 49- The Filter Sweeper

Another application of the Filter Sweeper is what we call "masked" Fade-Ins and/or Fade-Outs.  These "masking" techniques not only can be used to produce very interesting effects, but also provide a very useful noise reduction tool.  How can a swept filter produce noise reduction?  The principle is based on a simple musical concept coupled with the non-linear nature of the human sense of hearing.  Some musical material commences with very simple progressions (including only one or two musical instruments), then moving into a series of complex riffs and refrains (maybe including a few crescendos) and then often diminishing in complexity towards its ending.  Much of the noise in such a recording is hidden by the complexity of the musical material found it the central body of the piece.  The noise on the recording is heard to be most prevalent to the human ear at its beginning and its ending during the compositions simple parts.  (In actual fact, the noise level is probably relatively constant throughout the entire recording).  That is why we refer to this phenomenon as "masking.  This perception is a trick that the human ear plays on the listener which you can take advantage of as an audio restoration engineer.   By first selecting an area at the beginning or end of the piece and then using the various Filter Sweeper presets labeled "Masked Fade-In" or "Masked Fade-Out", you can smoothly contour the response of the restoration system during these quiet intervals at the beginning and ending of the musical rendition.  If there is excessive hiss present during these time periods, use the Low-Pass Sweeping filter.  If there is a high degree of low frequency noise (like rumble and or HVAC noise), use the High-Pass Sweeping filter.  If there is a combination of both of these types of noises, you will have to apply two passes of the Swept filter to obtain the optimal results, running one operation with the Low-Pass and another with the High-Pass.  Since some musical material is simple at its beginning and end, reduced frequency response will often not be observed by the listener due to the application of this new technique.  The "Sweep Type" that you choose will depend on the type of musical progression that you are dealing with, and is also somewhat subjective.

The Filter Sweeper provides the following controls:

•	Filter Type:

1.	Low Pass (First Order Butterworth)

2.	High Pass (First Order Butterworth)

3.	Notch (Second Order with 0.1 Octave Bandwidth)

•	Sweep Type: (determines the shape of the sweepers frequency versus time relationship)

1.	Linear

2.	Logarithmic

3.	Exponential

•	Start Frequency: (determines the frequency value in Hz from which the sweep begins.)

Range: 10 – 19,999 Hz

•	Stop Frequency: (determines the frequency value in Hz at which the sweep terminates over the determined highlighted area of the Wave file.)

Range: 10 – 19,999 Hz


The Forensics Menu

Surveillance, two-way radio, and noisy telephone communications place special burdens on noise reduction software.  FIVE/LIVE comes complete with a special set of filters aimed at Forensics applications that can be found under the Forensics Menu.  Besides being useful in Forensics applications, these filters can be used for a variety of non Forensics-related material.

Noise encountered in Forensics applications can be categorized into several basic categories.

•	Out-of-band noise:

This type of noise is rejected with the brick-wall band-pass filter.

•	In-band repetitive noise at a level lower than the target signal:

This type of noise is rejected with the brick-wall band-stop filter.

•	In-band random noise at a level lower than the target signal:

This type of noise is rejected by the standard Continuous Noise filter found under the Filter menu.

•	In-band repetitive noise at a level equal to the target signal:

This type of noise can be attenuated by the Adaptive filter set up for the Keep Residue (coherent) mode of operation.  If this noise is in the form of a "buzz" you should also consider trying the Harmonic Reject filter found under the Filter menu.

•	In band random noise at a level equal to the target signal:

This type of noise can be reduced by using the Adaptive filter set up for the Normal (random) mode of operation.

Several of our forensics filters can also be used in those extreme circumstances such as when the noise level is significantly greater than the "good" signal level.  In this instance, the ability to hear and understand the underlying speech is typically the goal – not a complete restoration.  See the Note below.

Note

It is important to note that the Forensics filters are not optimized specifically for "high fidelity" applications, but more for improving the intelligibility of speech or the discernment of subtle sounds buried in noise.


 The Adaptive Filter (LIVE ONLY)

This filter is used largely in audio applications where the ambient noise environment is constantly changing and the filter co-efficients must automatically change to maintain good intelligibility of an audio signal.

Figure 50- The Adaptive Filter

The Adaptive filter adjusts itself to remove a modeled signal representing the unwanted time domain waveform while preserving the target signal.  It uses the least mean squared calculation of the waveform signal to provide correction coefficients for its internal "leaky" filters.  It works best with a reference signal (in other words, a stereo or binaural source) containing only the noise to be rejected. This second channel or reference signal can be obtained from a second surveillance track with its microphone located near the noise source in the room such as a Jukebox or a television set.   However, it can also use its own signal as a reference in conjunction with the time delay function, which is provided for monophonic situations.  Additionally, the adaptive filter provides either the main processed signal or a keep-residue mode output signal for rejecting a wide variety of different types of noise sources.  Sometimes it will be found that the "keep residue" mode signal is more useful than the main output signal. Trial and error is sometimes the best way to determine the best mode to use.  The following controls are included with the adaptive filter:

•	Convergence- (Adaptation Speed) Slower adaptation speeds produce better noise rejection, while faster adaptation speeds produce better adaptation in quickly varying ambient noise conditions.

•	Filter Length (Samples) - The larger that this number is set for the more signal inflection points can be modeled in the time domain signal in order to be rejected or maintained.

•	Reference Signal - (signal to be compared to when a stereo recording is available) (choose the one that contains the reference signal to be used)

1.	Right Channel

2.	Left Channel

3.	Time Delay

•	Time Delay (Samples) for use in time delay reference mode only when no reference signal is available.  This essentially allows the signal being adapted to be its own reference.

•	Adapt / Freeze button - Selects adapt or freeze coefficients mode of operation.  Usually this is activated in Adapt mode for the first 5 seconds of noise.  In some cases where the sound ambient in constantly changing, one may choose not to "freeze" this filter.

•	Keep Residue button- This allows the operator to use the error signal rather than the output signal from the Adaptive filter.  This feature is useful for attenuating continuous loud or varying tones that may have been used to mask a Forensics recording.

New Adaptive Filter Normalize Allows Greater Automatic Filtering (LIVE ONLY)

This button, when checked, places the system into a "Normalized Least Mean Squared" mode, which is useful when attempting to clean up material with widely varying signal amplitude components.

Important Note:

Because of the wide range of convergence allowed in the Adaptive Filter, certain audio signals may cause the filter to become unstable and cut out.  In these cases, try changing the Convergence parameter until the audio is restored.  Usually lower convergence is more stable, but adapts more slowly to changes in the audio.  Lengthening the filter may also increase stability.


 Brick Wall Filter

These filters differ from their "HIFI" counterparts is that they are very steep digital filters used to attenuate signals that are interfering with a poor quality audio signal.  These are FIR (Finite Impulse Response) filters that exhibit a very high degree of out of band attenuation.  These filters are predominantly used for forensics audio applications.

Figure 51- The Brick Wall Filter

The Brick Wall Filters include the following choices of filter shapes:

1.	Low-pass:  Only allows signals below the corner frequency to be fed to its output.

2.	High-pass: Only allows signals above its corner frequency to be passed to its output.

3.	Band-pass:  Allows only the signal at its chosen frequency and within its bandwidth limit setting to be passed to its output.

4.	Band-stop:  Rejects all signals at its chosen frequency and within its bandwidth limit setting.

Several controls are provided:

•	Frequency (Hertz) (Range is from 3 Hz to 20,000 Hz)

•	Length (Samples up to 4094) - The larger the value of this parameter, the greater the degree of rejection past the filters corner frequency setting.  However, the higher the setting for length, the larger will be the computational demands on your computer.  So, large values of "length" will take longer to process or may make your machine "stutter" in preview or live real time modes.

•	Upper Frequency Control- Sets the upper limit for the filter's bandwidth.  (Range is from 3 Hz to 20,000 Hz)*

•	Lower Frequency Control- Sets the lower limit for the filter's bandwidth.  (Range is from 3 Hz to 20,000 Hz)*

*Note - Only applies to the Band-pass and the Band-stop filter.


 Polynomial Filter (LIVE ONLY)

This allows the mathematicians, scientists, and engineers to create their own transfer function using a polynomial expression.  For those not so inclined, there is a plentiful assortment of presets to choose from.

This system realizes a transfer function in terms of the input to output signal ratios.  It is useful for canceling the non-linearities that may have been introduced during a recording process.  Essentially, this feature can be useful in some circumstances for reducing harmonic distortion that was created by recording non-linearities. The method of data entry for the Transfer Function is in the form of the coefficients associated with a 5th order polynomial for which the actual transfer function graph is plotted automatically.  You can enter the coefficients numerically for each term or you can use one of the numerous presets that are provided to get you started with a particular function.  It is useful to have some mathematical background in order to effectively use this filter using co-efficient data entry.  However, trial and error is the best method for finding a setting that will reduce the distortion of a particular recording since the recording non-linearity is not generally known ahead of time.  Furthermore, sometimes adding non-linearities can be useful to enhance the intelligibility of extremely muffled conversations.  Limited instantaneous dynamic expansion and compression can also be realized by using this feature, for which there are several presets provided for convenience.  Interesting distortion creation and frequency multiplication is also possible with this algorithm.

Figure 52- The Polynomial Filter

1.	The Transfer Function is given in the general form by the following polynomial expression:

Y = A5*X^5 + A4X^4 + A3X^3 + A2X^2 + A1X + A0

You have a field in which you can enter each of the co-efficients the polynomial, including:

A5 =_____ A4 = _____ A3 =_____ A2  =_____ A1 =_____ A0 =_____

Note:

Values of A can be positive or negative.

•	Input (Gain):

Range: 0.001 to 1.999  (1.000 represents unity gain)

This controls the level of the signal being applied to the input of the polynomial expression.

•	Offset (DC):

Range: -1.00  to  0  to +1.00  (0 represents no offset value)

This control effects the DC offset applied to the input signal to the equation.

•	Output (Gain):

Range: 0.001 to 5.000  (1.000 represents unity gain)

This control effects the output of this system after the polynomial equation has been applied to the signal .

•	Mix:

Range 0.000 to 1.000

This control affects the degree to which the processed signal is added to the input signal for presentation to the systems output.  0 represents no polynomial effect and 1.000 represents complete polynomial effect on the output.


 32,000 Band Spectral Filter (LIVE ONLY)

This filter allows you to create a very high-resolution frequency response contour using up to 32,000 bands of equalization by using FFT techniques.  The user interface system is intuitive and allows you to zoom-in on a particular portion of the spectrum that needs accurate and specific frequency response contouring. By using the right mouse button, you can add bands, or delete them simply by pointing and clicking on the graph.  This is very useful in Forensic audio applications for removing in-band and out-of-band extraneous noises because of its high degree of frequency selectivity and its very steep slope characteristic.  The following controls are provided:

Figure 53- The Spectral Filter

•	Graphic representation of the Frequency Response:  Drag the blue inflection points (dots) with your mouse in order to establish the desired frequency response.

•	Range (dB): This parameter determines the vertical axis range for the Spectral Filter.  The following ranges are provided:

1.	+40 / -100 dB

2.	+20 / - 40 dB

3.	+ / - 40 dB

4.	+ / - 20 dB

5.	+ / - 10 dB

6.	+ / - 3 dB

•	FFT Size (number of Frequency Bands):

1.	256 (128 Bands)

2.	512 (256 Bands)

3.	1,024 (512 Bands)

4.	2,048 (1,024 Bands)

5.	4,096 (2,048 Bands)

6.	8,192 (4,096 Bands)

7.	16,384 (8,192 Bands)

8.	32,768 (16,384 Bands)

9.	65,635 (32,768 Bands)

•	Zoom In: Click on the Magnifying Glass Icon with the "+" sign or drag the mouse over the area of the spectral graph on which you desire to focus.

•	Zoom Out: Click on the Magnifying Glass Icon with the "-" sign.

•	Gain Control Slider: Adjust the Spectral filter overall output over the range of –40 dB to + 40 dB.

•	Bypass Check box:  Allow you to hear the source material with the Spectral Filter out of the system.

•	Keep Residue: Allows you to hear what you are removing when using the Spectral Filter.

•	Reset: Returns all off the blue inflection points (bands) on the graphical display to a "flat" response.

•	Presets: Save or Delete buttons

•	Right Mouse Button:

To use this feature, select the function you desire with the right mouse button and move the band with the left mouse button.  Here are the functions available for the Spectral Filter by using the Right Mouse Button:

1.	Add Point (adds a frequency band on the graph where you are pointing with the mouse)*

2.	Delete Point (deletes a frequency band on the graph where you are pointing with the mouse).

3.	Reset Point Count (to system default value)

*Note:  Adding frequency bands can also be accomplished by double-clicking the left mouse button.


Spectrograph

 Ideal For Voice Prints and Acoustical Analysis (LIVE ONLY)

A Spectrogram provides a method for displaying waveform data including Time, Frequency and Intensity all on the same graph.  Time is represented on the horizontal (X) axis, frequency is represented on the vertical (Y) axis, while intensity (loudness) is represented by color or gray scale. The Diamond Cut Spectrograph displays itself in the Destination window when this is checked.  The spectrogram is calculated for whatever waveform is displayed, highlighted and zoomed in on in the Source window and is time aligned with the same.

Figure 54- The FIVE/LIVE Spectrogram View

Display Controls:

•	Right Vertical Slider: Controls the amount of gain applied to the Intensity modulation of color or gray scale.

•	Left Vertical Slider: Controls the relative position of the displayed Spectrogram.

•	Zoom In*: Allows you to zoom in on a portion of the Wave file as displayed in the source window.

After zooming is completed, the system will re-calculate the spectrogram for the zoomed-in

displayed data in the destination window.

•	Zoom Out*: Allows you to zoom out on a portion of the Wave file as displayed in the Source

Window.  Similarly, the system will re-calculate the spectrogram for the zoomed-out displayed data in the destination window.

Display Preferences:

You can choose between a number of preferences associated with the spectrogram under the Preferences menu found under "edit," or by double clicking on the spectrograph display area.  The following preferences are available to you:

•	Frequency Axis Selector:

A.	Linear

B.	Log

•	FFT Size:

Choose between 32, 64, 128, 512, 1024, 2048, and 4096.  Small values provide fast FFT update time, while large FFT sizes provide improved frequency resolution.  The frequency resolution is the FFT size / 2.

•	Color Palette:

You have the choice of the following color gradients:

1.	Grayscale

2.	White to Blue

3.	White to Red

4.	White to Green

5.	White to Red to Blue

6.	White to Green to Red

7.	White to Red to Blue to Black

8.	White to Yellow to Red to Black

9.	White to Yellow to Green to Aqua to Blue

•	Display Frequency Range

1.	Enter Value for Minimum Frequency in Hz

2.	Enter Value for Maximum Frequency in Hz

•	Display Frequency Labels

This turns the Frequency Labels along the Vertical axis On or Off

*Note:

For more information on methods for Zooming-In and Zooming-Out, please refer to that section of the Users Guide.

Spectrographs are useful for applications like spectrographic voice recognition or comparison (sometimes referred to as "voice-printing.")   Physiologically, speech is produced by the interaction of two mechanisms consisting of resonance and articulation. Resonance is produced by the nasal, pharyngeal and oral passages while articulators are produced by the jaw muscles, lips, teeth, tongue, and the soft palate.  Typically, samples that are 2.5 seconds in length with the frequency display range showing information from 100 Hz to somewhere between 3 to 6 KHz are used for vocal comparisons.  The so-called English "cue words" often used for comparison are as follows: {The, To, And, Me, On Is, You, I, It, A}

To perform a comparison, it is necessary to observe the same words contained on the two specimens (the display tile feature is helpful for this purpose).  It is beneficial if the same recording equipment was used to record both samples of data to be compared, however impractical in most situations.  It is also beneficial if the same ambient sound conditions are presented on the two samples.  If there is a lot of background noise, consider applying one of the Speech filters before measuring the spectrogram.  Speech filters can be found in the Band-pass filter preset menu and the Forensics Brick Wall filter preset menu.  See those specific sections of the users guide for details.   Lastly, the emotional state of the persons making the expressions should be similar.  If one specimen, for example, has the person screaming, and the other has the person sobbing or whispering, it will be difficult to draw any conclusions with a degree of certainty.  Please note that spectrographic "voice-printing" is not admissible evidence in all court systems in the United States.   Contact the court system or a legal expert in your area for details.


 DeClipper

The De-Clipper is provided to reduce the distortion, which results from an over-driven or clipped signal.  When viewed in the time domain via the Source display window, this problem often looks like the signal has a "crew cut" or is "maxed-out".

Figure 55- The DeClipper

The De-Clipper can be used to repair signals, which were either clipped by digital or analog mechanisms.  It performs its magic by detecting signals with very low or zero values of slope (user adjustable from 0 - 0.5) above a settable threshold amplitude value.  When this condition is detected, the routine mathematically interpolates a new signal and replaces the zero slope portion of the bad waveform with one containing curvature.  This results in decreased distortion.  If the material being de-clipped has been directly clipped by the digital recording process (in other words, the signal is clipped at full scale output as indicated on the destination window), then you must first decrease the overall gain of the Wave file by 3 dB before applying the De-Clipper.  This sort of de-clipping can be accomplished with very low values of slope.  If the signal was clipped previously to transfer to the digital domain by an overloaded analog amplifier, the signal can be de-clipped by raising the slope control until the "total clips fixed" display starts incrementing.   The following is a listing of controls available on the de-clipper:

•	Threshold

The threshold determines the amplitude below which signals will be applied to the de-clipper detector.  Lower values imply a more aggressive response.  The range for this control is from 10 to 100 on a relative and normalized percentage scale.

•	Slope

The Slope control determines the "flatness" of the clipped waveform that will be interpreted as a "clipped" event.  Low numbers like zero, imply a perfectly flat line as will be found in digital clipping.  Higher numbers represent slight slopes associated with analog clipping.  The range for this control is from 0.000 to 0.500 with 0 representing zero slope and 0.500 representing a 45 degree slope.

•	Strength

The Strength control affects the curvature of the applied interpolated waveform.  Adjust this for the best sounding (or looking) replacement waveform.  The range for this is from 1 to 5, with 5 being the most aggressive.

•	Total Clips Fixed:

This is a numerical display indicating how many "clips" were detected and fixed by the de-clipper.  It is useful when using "preview" mode to assure that the routine is detecting the clipping while adjusting the various controls.

•	Bypass

This bypasses the de-clipper so that you can preview and hear the before and after results of the de-clipping process in an instant.


De-Clipper Operating Procedure (Tutorial)

1.	Set the Threshold control to its maximum value (all the way up).

2.	Set the Slope to a value close to zero for digital clipping, but higher for analog clipping.

3.	Preview the filter and watch the Total Clips Fixed display.

4.	Reduce the level of the Threshold until it begins to show increments on the Total Clips Fixed display.

5.	Adjust the Strength for a minimization of distortion as heard in preview mode.

6.	Run the Filter.

7.	Done

Note:

You can look at the results produced by the De-clipper after you have run the Filter.  You will notice that the flat-topping has been replaced with smooth rounded waveforms if the controls have been set properly.


Additional Forensics Features

There are additional tools that will be of value for forensics work which are not found in the Forensics menu.  Some of them are as follows:

1.	Automatic Level Control (ALC or AGC) (Dynamics Processor):

Sometimes, these algorithms or systems are referred to as Automatic Gain Controls or AGC's.  This feature provides upward expansion of signals below the threshold line and downward compression of signals above the same threshold.  This feature is useful in forensics applications where there is a large variation in signal levels between several different parties that may be communicating with one another.  It is also useful for the broadcast of live sporting events (if you have the LIVE version of the product) in which the crowd reaction is of interest when the announcer is not speaking.  Simply clicking on the ALC box in the Dynamics Processor activates this feature.  The threshold, attack, and release controls are still active when this function in invoked. Many Forensics specialists have also found the automatic level control feature (ALC) in the Dynamics Processor to be very useful in their work. This feature is particularly useful for correcting near-party / far-party loudness discrepancies of recorded phone conversations and other situations where signals vary greatly from time to time.

2.	Punch and Crunch Effect operating in Narrow band mode:

This is useful for improving the intelligibility of signals having very small values of dynamic range in the speech portion of the spectrum.  It is advised that a speech filter is run first before applying this filter for enhancement.  For more information, please refer to the section on the Punch and Crunch effect.

3.	Continuous Noise Filter:

This can be very useful for decreasing "in-band" noise found on Forensics recordings.  A noise fingerprint will have to be taken from a section of the recording having only noise and no signal in order to use this filter.  For more details, refer to the section on the Continuous Noise Filter.

4.	Harmonic Reject Filter:

This is very useful for attenuating buzzes found on some recorded phone conversations.  Several passes through this filter are often required for severe noise situations.  Also, it is worth trying both odd and even harmonics of the fundamental buzz frequency.  The fundamental buzz frequency can be found by using the Spectrum Analyzer located under the "View" menu.

5.	Impulse Filter:

This is very useful for eliminating static from body mike recordings transmitted via an RF carrier.

6.	Slot (which is a subset of the Notch) Filter:

This can be useful for isolating specific sounds buried in a recording.

7.	Median Filter:

This is useful for bringing out the subtle articulations of speech in low-bandwidth or muffled voice recordings, especially when used in conjunction with its "Weighting" function.

8.	Spectrum Analyzer in "High Resolution" mode:

This is useful for tape authenticity verification and other forensics analysis functions.

9.	Source Time Display window in conjunction with the Timer found in the View Menu and the Markers found in the Marker Menu:

This is useful for precisely measuring the time between multiple events such as gunshots.

10.	Live Mode is useful for surveillance situations in which signals must be cleaned up "real time."

11.	Live Log to Disk Mode is useful for recording conversations selectively while in Live, real time mode.

12.	High Resolution Spectrum Analyzer


The Marker Menu

FIVE/LIVE provides you with movable red timing markers that can be placed within either the Source or Destination workspace.  They are the Wave file equivalents to bookmarks.  You will find them to be useful tools when utilizing such editing features as the Copy, Cut, Mute, etc. features of the product. The Marker menu is also complimentary to the CD Prep menu. It is very useful when you need to take a very long Wave file and Chop it up into smaller files manually for indexing onto a CD-R. The following commands are provided under the Marker Menu:

Add Markers: This feature adds up to 100 markers to the desired Workspace.  Clicking here will drop a marker at the beginning of a selected area.  If no area is selected, by default, the marker is dropped at the far left of the Wave file.  This command can also be accomplished by use of the right click menu.

Clear All Markers: This feature erases all existing markers.

Highlight Marked Area: This feature highlights in Yellow the area of a Wave file located between two previously defined red markers.

Drop A Marker: This command allows you to drop a marker during playback so you can audition your audio while "marking it up" for in-depth study later.  Markers can also be dropped by simply hitting the "M" key on your keyboard during playback.

Go To Next Marker: Exactly as the name implies…allows you to quickly navigate to the next marker on the right.  This can also be accomplished by simply pressing the "N" key on your keyboard.

Go To Previous Marker: Same as above, only this command takes you to the previous marker on the left.  This can also be accomplished by simply pressing the Shift + "N" keys on your keyboard.

 Label Marker Helps Clean Up Your Work Area:

By default, FIVE/LIVE just places a number on each marker.  If you use several markers, it may be necessary to label them more accurately.  The Label Marker command allows you to be more precise with your markers.  Right clicking on your mouse and activating the Label Marker command can also accomplish this same task.

Lock Markers Prevents Accidental Movement

Now you simply click on this menu item and your markers cannot be accidentally moved while you’re working on your file.  In order to move them, you’ll need to press the Ctrl button plus drag the mouse.


The CD Prep Menu

This menu contains the tools that help you prepare for the final stages of your restoration project.  The CD Prep Menu does everything but copy your files to CD.


Quantize for CD Audio

This feature moves a marker to a multiple of 2352 Bytes to provide computability with CD data grouping so that glitch-less indexing can occur. This feature is particularly useful when chopping a large (continuous concert type) Wave file into pieces for transfer to CD-R.  If you have a Wave file open, merely click on this menu item and the file will be properly quantized.


Using Quantize on a Single File (Tutorial)

1.	Bring up desired Wave file in the source workspace

2.	Click on the CD prep menu item.

3.	Click on Quantize for CD Audio

4.	Done


Using Quantize and Chop File on a Large File (Tutorial)

1.	Bring up desired Wave file in the source workspace

2.	Place markers at all of the locations at which you desire to break the file into individual Wave files.  These are usually placed at the silent portions of a recording, between cuts.

3.	Click on the CD prep menu item.

4.	Click on Quantize for CD Audio.  All of the markers shall be moved into the proper positions for CD quantization.

5.	Chop file into pieces using the command by the same name.

6.	Done


Chop File into Pieces

This command breaks a long Wave file into smaller Wave files as defined by the locations of your various markers.   This command will remain grayed out until your file has markers laid.

Figure 56- Chop File Into Pieces

If you’ve chosen to record a full album length file onto the hard drive and are now ready to break the large file into the smaller "song" files, this feature, used in conjunction with either your markers (manually finding the beginning and ending of your "songs") or with the Find and Mark Silent Passages tool (automatically finding the song tracks by the silence in between two songs…described below) allows you to accomplish this task.   It allows you to choose from any or all of the following options:

•	Preserve Original File: After you’ve ‘broken up" your main Wave file, do you still want to keep it as a master?

•	Create Play List:  This launches the Play List editor, which allows you to set up your CD’s running order, allows you to add, audition or remove specific titles.  The play list also allows you to export your cue sheet for other CD burners in the .cue format.  For more info on the playlist, refer to the earlier chapter on the File Menu.

•	Open Files After Chopping: In most cases, you’ll want to audition your "cuts" for accurate beginnings, endings, volume continuity, etc. before actually burning your CD.  This checkbox automatically opens your new "cuts" so that can take place.

•	File Naming: This will list the base name of your Master Wave file and then you can choose whether the resulting "chopped" pieces will be named with letter or number suffixes.

•	Directory for Resulting Files: Allows you to choose a storage facility for your resulting files.


Find and Mark Silent Passages

FIVE/LIVE includes a feature, which will automatically find and mark the silent passages of your Wave file.  This is particularly useful when you desire to process an entire Vinyl record album in one shot through the various algorithms, and then break them up into separate Wave files at the end of the process.  You have the ability to select the threshold of silence, and the time duration of the silence.  After you have invoked this feature, you will see all of the markers moved to the silences between cuts.  You can move the markers manually, if you are not satisfied with the separations that were automatically determined by the program.  After this has been completed, you can chop the file into pieces, and separate Wave files will be created.

Figure 57- Find and Mark Silence


Gain Normalize

The Normalize Gain feature searches an entire Wave file looking for the peak signal level.  Then, it adjusts the gain applied to the file so that the overall level is below that value.  This will provide the best signal to noise ratio and a reasonable volume balance for each "cut" on your final master.  Normalize Gain should be applied before burning a CD-ROM or making your final tape.

Important Note:

In Classic Editing mode, Gain Normalize will search the entire file for the peak and adjust the gain based on that peak…it will ignore any selected area.  In Fast Edit mode, Gain Normalize will abide by a selected area, find the peak there and adjust the selected area only with the appropriate gain.


  Normalized Gain Scaling

It allows you to scale the gain of a Wave file to values other than 0 dB (full-scale output).  The range of adjustment provided is +/- 20 dB, which corresponds to a gain factor range of +/-10.  If you apply gain scaling above 0 dB, some portion(s) of the Wave file will be clipped.  This could be useful in a situation wherein a single transient pulse or two are dominant in the Wave files amplitude, and "clipping" it is irrelevant to you.  If you apply scaling below 0 dB, all resultant signals shall be below full-scale output.


The View Menu

The View menu allows you to access the commands that effect the manner in which files, controls, and parametric displays are presented to the user. To activate a particular feature, click on it with the left mouse button.  Once it has been activated, a check mark will appear next to the command.


Spectrum Analyzer

This command brings up the floating Spectrum Analyzer. The Spectrum Analyzer is capable of resolving frequency increments as small as 0.02 Hz when set appropriately.  This Analyzer can be used with any of the filters or effects.  It is connected to the output of the filter or effect, so that you can see, in the frequency domain, how you have affected the file.  If you want to compare the output of the filter or effect to the input, use the Bypass function on the filter or effect window.  Since the Spectrum Analyzer utilizes constant Hertz per frequency band, it will display white noise as a flat, horizontal line.  This is unlike octave weighted real time audio analyzers in which white noise produces a diagonal, positive sloped line, and pink noise produces a flat horizontal line.  Conversely, pink noise displayed on the Spectrum Analyzer will be displayed as a negative sloped diagonal line.  The Spectrum Analyzer displays the algebraic sum of the left and right channels.  Keep in mind that the use of the Spectrum Analyzer will slow your system down slightly. Therefore, when you are done using it, shut it down.

Figure 58- The Spectrum Analyzer

The following displays are provided on the Spectrum Analyzer:

•	Frequency vs. Amplitude Graph:  The vertical axis indicates 0 dB at the top and ranges down to -100 dB at the bottom.  The horizontal axis indicates frequencies from 10 Hz (left) to a little over 20 KHz (right.)

•	Two digital readouts indicate the frequency and amplitude of signals feeding the spectrum analyzer.  One signal is the peak value and the other is user defined.

The following controls are provided:

•	Display Mode:

1.	Fast:  Shows the spectrum in almost real time.

2.	Slow:  Shows the spectrum with slower ballistics.

3.	Averaging: On/Off:  This allows the system to provide you with the average signal spectrum rather than a real time display.  The averaging interval will be as long as the analyzer is left in operation.

•	FFT Size:  11 selection alternatives (64, 128, 256, 512, 1,024, 2,048, 4,096, 8,192, 16,384, 32,768, & 65,536). This selection determines the frequency resolution of the spectrum analyzer.  The higher the number selected, the better the frequency resolution of the display.

•	Range: This scales the vertical axis to 100 dB, 50 dB, or 20 dB full scale.  This feature along with the offset control allows you to hone in on a particular signal.

•	Hold Button:  This is a "toggle" function and will allow you to freeze or un-freeze the spectral display update.

•	Clear Button:  This clears the display.  It is especially useful when using averaging mode so that you can clear the display after a long averaging interval, allowing you to move onto a new area of the Waveform to be averaged.  Clearing the display will re-initiate the averaging process as well as clearing the display.

•	Show Peak Button:  This feature will automatically find the peak amplitude signal and display its Frequency and Relative Amplitude value in the upper right hand corner of the Spectrum Display screen.  The marker and display for this feature are red in color.

•	User Controlled Marker:  You can place a marker anywhere you want on the spectral display by clicking the left mouse button on the peak that you are interested in measuring.  To accomplish this, merely point the mouse cursor to the peak of interest and click the left mouse button.  A green marker will appear at that location and a yellow digital display of the frequency and relative amplitude of the signal that you pointed to will appear in the upper left hand corner of the spectral display.  To read another value, merely click the mouse again, pointing to a new spectral line.  The marker and the display will then be updated.

•	Frequency Resolution:  The Spectrum Analyzer has the ability to display frequencies with the following values of resolution as indicated on the chart below.  Note the interaction between the FFT Size selected and the available Resolution in Hz that are available:  "OK" indicates available ranges.  "NA" indicates Not Available resolution values by Frequency Band.  It is very important to note that small values of FFT Size produce poor values of Resolution, but the response will be very fast.  Conversely, large values of FFT Size produce good values of Resolution, but the response will be very slow.  This latency time (or display refresh / update) is a function of the Spectrum Analyzer frequency resolution setting.  The latency expressed in Seconds, is displayed in the right-hand column for each value of resolution.

Resolution Chart

0.02	0.04	0.08	0.17	0.34	0.68	1.35	2.96	5.38	10.77	21.53	43.07	86.13	172.27	344.53	689.06	Res. Hz		FFT Size
NA	NA	NA	NA	NA	NA	NA	NA	NA	NA	OK	OK	OK	OK	OK	OK		64
NA	NA	NA	NA	NA	NA	NA	NA	NA	OK	OK	OK	OK	OK	OK	NA		128
NA	NA	NA	NA	NA	NA	NA	NA	OK	OK	OK	OK	OK	OK	NA	NA		256
NA	NA	NA	NA	NA	NA	NA	OK	OK	OK	OK	OK	OK	NA	NA	NA		512
NA	NA	NA	NA	NA	NA	OK	OK	OK	OK	OK	OK	NA	NA	NA	NA		1,024
NA	NA	NA	NA	NA	OK	OK	OK	OK	OK	OK	NA	NA	NA	NA	NA		2,048
NA	NA	NA	NA	OK	OK	OK	OK	OK	OK	NA	NA	NA	NA	NA	NA		4,096
NA	NA	NA	OK	OK	OK	OK	OK	OK	NA	NA	NA	NA	NA	NA	NA		8.192
NA	NA	OK	OK	OK	OK	OK	OK	NA	NA	NA	NA	NA	NA	NA	NA		16,384
NA	OK	OK	OK	OK	OK	OK	NA	NA	NA	NA	NA	NA	NA	NA	NA		32,768
OK	OK	OK	OK	OK	OK	NA	NA	NA	NA	NA	NA	NA	NA	NA	NA		65,536
64	32	16	8.0	4.0	2.0	1.0	.5	.250	.125	.0625	.0312	.0156	.0078	.0039	.002		(In Secs)	Latency

Note 1:

Because the FFT creates solutions in both the real and imaginary planes, the actual number of frequency bands created is the FFT Size / 2.

Note 2:

NA = Not Applicable

•	Offset Slider Control:  This allows you to move the centering of the spectral display up or down.  It is of particular value when the "Range" control is set to a high sensitivity value such as 50 dB or 20 dB, and the signal appears to be off of the screen.  By using the Range control and the "Offset Slider" control, you can zoom-in on a signal of interest.

•	Window Selection:  Seven window selections are provided so that you can establish the appropriate tradeoff between Stop Band attenuation and Lobe Width.

A.	Blackman

B.	Hanning

C.	Hamming

D.	Rectangular

E.	Kaiser-Bessel

F.	Triangular

G.	Bessel

The following chart shows the performance of several of the provided window functions.  Others are provided with the software for completeness.

Window Name	Side-Lobe Peak Amplitude in dB	Main Lobe Band-pass Width	Minimum Stop-Band Attenuation in dB
Blackman	-57	12 pi / N	-74
Hanning	-31	8 pi / N	-44
Hamming	-43	8 pi / N	-53
Rectangular	-13	4 pi / N	-21
Window Name	Characteristics	Application
Blackman	Exhibits the best amplitude resolution and accuracy, but has the poorest frequency resolution	Very Good for measuring single frequency signals in order to look at its higher order products.
Hanning	This exhibits fair frequency resolution but poorer amplitude accuracy compared to the Rectangular Window	Good for measuring Narrow Band random Noise, measuring Periodic Signals and for measuring transient bursts wherein the signal levels prior to and just after the event are much different in amplitude.
Hamming	This window is very similar to Hanning, however it exhibits slightly better frequency resolution and accuracy compared to Hanning	Same Applications as Hanning.
Rectangular	Exhibits the best frequency resolution, but has the poorest amplitude resolution and accuracy.  In essence, Rectangular is not a window, per se, at all.	Good for measuring sine waves that have similar amplitudes and whose frequencies are loose, for measuring broad band noise with slowly varying frequency spectrum and for measuring Transient bursts wherein the signal levels before and after the key event are similar in amplitude
Note:

Hanning is the recommended default window to use for general applications.

•	Display Type:  Choose between Line or Bar (Graph) modes.


New Distortion Analyzer in Spectrum Analyzer (THD Mode)

This selector box converts the Spectrum Analyzer into a high performance Distortion Analyzer when enabled.  A pure Sine wave obtained from either a very high quality external signal generator or the "Make Waves" generator must be used as the stimulus for the device under test (DUT) or algorithm under test.  The THD meter measures the fundamental signal component separating it from the rest of the total harmonic signals presented to the system.  It also "notches" out the fundamental and sums the remaining harmonics.  Then it takes the ratio of the fundamental to the sum of the measured harmonics and expresses it as a percent on the digital readout.  Even and Odd order distortion components are produced by transfer function non-linearity.  Even products are dominantly due to transfer asymmetry.  Note:  The Show Peak checkbox must be checked before enabling the THD display.

Important note:

When not using the Distortion Analyzer, turn it off to minimize CPU utilization.

•	Ultra High resolution Forensics mode:  Set the FFT Size to 4,096 or greater. If you then click on 0.67 Hz or smaller, the system will re-scale the horizontal (X) axis of the spectral graph to indicate 700 Hz full scale rather than 20 KHz when you run the spectrum analyzer.  You can resolve frequency increments as small as 0.02 Hz as indicated by the settings shown on the resolution chart.  To augment this capability, use the "Range" control setting set to 20 dB and the "Offset" control to zoom in on the signals of interest.  This combination of features are particularly useful when trying to determine if a forensics recording is a "dub" of the original by looking for two discrete "hum" frequencies on the spectrum.  For more information on tape authentication, please refer to the "How do I?" section of this users manual.

Important note:

When using the Spectrum Analyzer in Ultra High resolution mode, it will take a long time for the system to integrate.  A message will appear at the bottom of the display during this interval stating "Sampling, Please Wait."  Refer to the resolution chart for latency details.


Measuring the Spectrum of only the Left or the Right channels of a Wave file (Tutorial)

The Spectrum Analyzer displays the algebraic sum of both channels in normal operation.  If you want to obtain the spectrum of only one channel, use the following procedure:

1.	Bring up the Spectrum Analyzer.

2.	Set its parameters appropriate for the measurements that you wish to make.

3.	Bring up the File Conversions feature found under the Filter menu.

4.	Choose Stereo to Left only or Stereo to Right only, depending on the channel of interest.

5.	Click on Preview, and only the chosen channel will be presented to the Spectrum Analyzer.


Measuring the % Total Harmonic Distortion (%THD) of an Audio component (Tutorial)

The Spectrum Analyzer contains a special feature which allows you to measure the % Total Harmonic Distortion of a piece of electronic equipment, whether it be a CD player or an audio amplifier with a fairly high degree of accuracy.  This will require, however, the use of a professional grade, high performance sound card and CD player in order to achieve accurate results.  There are two procedures that you can choose from.  The first procedure is simple to perform, but less accurate than the second procedure which is more complicated.

•	Procedure #1 (Simple Method):

1.	Create a 5 minute stereo 44.1 KHz sampled 1000 Hz* Sine waveform Wave file by using the Make Waves feature.

2.	Name the file, and make sure you know the path on which it was stored.

3.	Use your ROM burner software to make a CD of this Wave file.

4.	Connect the output of the CD player to the component under test’s input

5.	Connect the output of the device under test (DUT) to your sound card’s line input.‡

6.	Launch FIVE/LIVE

7.	From the View menu, enable the Spectrum Analyzer.

8.	In the Spectrum Analyzer, enable the THD meter feature.

9.	While operating in "Live" mode, play the CD.

10.	 You should see a dominant spectral spike at 1 KHz.

11.	 The Distortion Analyzer will calculate the %THD and display it in the THD meter window.

12.	 Done

•	Procedure #2 (Accurate Method):

1.	Perform steps 1-3 from the above tutorial

2.	Connect the output of the CD player to the line input of your sound card.

6.	Launch FIVE/LIVE.

7.	From the View menu, enable the Spectrum Analyzer.

8.	In the Spectrum Analyzer, enable the THD meter feature.

9.	While operating in "Live" mode, play the CD.

10.	You should see a dominant spectral spike at 1 KHz.

11.	The Distortion Analyzer will calculate the CD and sound card %THD and display it in the THD meter window. Write this value down and call it %THD(I) (instrumentation).

12.	Disconnect the CD player from the line input of the sound card and then    plug it into the electronic device under test’s (DUT) line input.

13.	Connect the output of the DUT to your sound cards line input.‡

14.	In the Spectrum Analyzer, observe the THD meter feature and write down the %THD indicated.  Write this value down and call it %THD(T) (total).

15.	Calculate the DUT distortion using the following equation:

%THD(DUT) = (((%THD(T))^2) – ((%THD(I))^2))^1/2

Note:

Lower frequency Wave files can be used.

‡ Note:

If a power amplifier is being tested, appropriate resistive loading must be connected to its output, along with an appropriate attenuator between the amplifiers output and the sound card input.  DO NOT connect the output of a high-output audio power amplifier directly to the input of your sound card.  If you do so, the result could be enough to make a grown man cry since it will destroy your sound card.  We are not worried about women here, since they would not be so dumb as to make this mistake.


XY Display

The X-Y Vector Display displays the phase relationship between two waveforms. This feature is primarily used to align the azimuth of analog tape deck recording and playback heads.  To do so, a known pre-recorded azimuth alignment tape is required containing a fixed frequency tone.  The X-Y Vector display in conjunction with your azimuth tape and the Time Offset feature can be used to fine-tune a recorders azimuth alignment without the necessity of having to take a screwdriver to your tape deck's head alignment adjustment screws.  The goal here is to correct the azimuth of the Wave file using the File Conversions Filter with the Time Offset slider control.  Preview the Wave file with the X-Y Vector Display showing, and adjust the Time Offset slider control until a 45 degree positive sloped (up and to the right) straight line is seen on the display.  That will be the optimum value of azimuth correction.

Figure 59- X-Y Vector Display Showing Proper Azimuth Alignment

The following features will be found on the X-Y Vector display:

•	X-Axis displacement (Horizontal), which corresponds to the Left Channel Input

•	Y-Axis displacement (Vertical), which corresponds to the Right Channel Input

•	X-Axis gain control slider (Horizontal in position)

•	Y-Axis gain control slider (Vertical in position)

Here is a listing of some vector displacements, which can be observed on the X-Y Vector Display.  They have the following meaning and are sometimes referred to as Lissajous figures:

1.	Straight line at 45 degrees with a positive (up and to the right) slope = Signals are in phase and are Monophonic.

2.	Straight line at 45 degrees with a negative (down and to the left) slope = Signals are out of phase and are Monophonic. Ex.  An example of a situation that could cause this would be a miss-wired Phono cartridge.

3.	Straight horizontal line only = Monophonic Right channel only signal.

4.	Straight vertical line only = Monophonic Left channel only

5.	Straight line at 45 degrees with a negative (down and to the left) slope = Signals are 180 degrees out-of- phase

6.	Circle = Signals are 90 degrees phase shifted

7.	Frozen figure "8" = Signals are frequency phased locked to one another but 2:1 in frequency ratio.*

8.	Moving figure "8" = Signals are not frequency locked, but are abut 2:1 in frequency ratio.*

9.	Random pattern of squiggly lines on the screen = Stereophonic audio signal

10.	Etc.

*Note:

In these examples, the right channel would be twice the frequency of the left input.  If the figure 8 were lying on its side, then the left channel would be twice the frequency of the right input.


Time Display

The Time Display view displays all of the time related parameters associated with a source or destination workspace.  When activated, a display window will appear containing four sets of timing numbers.  This window can be dragged and placed anywhere in your workspace.  The following time related parameters are displayed:

1.	Cursor Location (largest numerals)

2.	Start Location of a highlighted area (small numerals)

3.	Stop Location of a highlighted area (small numerals)

4.	Span- this represents the total time of a highlighted area (small numerals)

Figure 60- Time Display Window

You can monitor "Samples" rather than "Time" in this display if desired.  Go to the Preferences Menu and choose the "Display" option.  Under that category, you will find a "Display Time Format" choice.  Click on "Samples".


Output VU Meter

Two 100 segments VU meters can be displayed which will indicate the output of any filter multiple filters that are being used by LIVE or FIVE.  These meters indicate the level of   the left and right channels and have both average and peak reading ballistics.  They are calibrated to indicated values from -50 dB to 0 dB, with 0 dB being full-scale output.  Any signal above that level will be clipped by the system. Signals that are clipped are indicated by the illumination of a small red "LED" indicator at the top and in the middle of the meter display.

Peak Hold Added to VU Meters

The meter includes two peak indicators.  The white horizontal bar is designed to "hold" the indication of an overload for two seconds after it occurs so that it will be more obvious to you that a clipping event has occurred. The green horizontal bar is a peak hold indicator and will hold the value of the audio peak until such time as the reset button, located at the bottom of the VU meters, is activated by the mouse. These VU meters can be activated or de-activated under the View Menu.  Also, they can be "dragged and dropped" anywhere on your desktop workspace using your left mouse button.

Figure 61- VU Meters


Volume Control

This feature allows you to view and adjust the sound card primary volume controls.  Two controls are presented:

1.	Main Volume Control

2.	Wave Volume Control

Important Note:

This volume control is only useful for sound cards that make use of the Windows Mixer.  For any card that doesn’t use the Windows Mixer, adjustment of volume will take place using the proprietary mixer of that product.


Fast Edit History

When working in Fast Edit mode, this cool little window is like having your own personal secretary.  It keeps track of every edit and allows you to navigate quickly from one to the other.  If you decide to start at midpoint in this list and want to eliminate all edits done after this point, just double click on that edit and the system will ask if you’re sure…then eliminate all edits after this point.

Figure 62- Fast-Edit History View


Zoom In

Zoom In allows you to magnify any portion (no matter how small) of your Wave file.  It can be used to with either the Source, Destination or Fast Edit workspace.  The Zoom-In process may be repeated any number of times for a really close and detailed look of your audio waveforms.  However, only the last 5 zoom levels are remembered.   You can also access this feature by using the Z hotkey or the Right Click Menu.


Zooming-In & Zooming-Out on portion of a Wave file (Tutorial)

To Zoom-In and Zoom-Out on a portion of a Wave file, use the following procedure:

1.	With your mouse, left click on the workspace location that you desire to magnify.

2.	As you begin to drag the mouse pointer towards the right-hand portion of the screen, you will see two black timing reference markers appear in the workspace.

3.	The first marker will remain at the location at which you began to perform the "mouse-drag" operation, indicating the location of the "start" position of your zoom region of the Wave file.

4.	When you get to the end of the interval of the Wave file portion you wish to Zoom-In on, release the left mouse button, and the second line will remain at that location.  This line will have indicated the location of the "end" position of your selected zoom-in region of the Wave file.

5.	Click the mouse on the Zoom-In icon

6.	The selected area will be re-displayed with the time axis expanded.

7.	To zoom back out, merely click with the left mouse button on the Zoom-Out button on the toolbar.

8.	After you have clicked on the Zoom-In icon, you will notice that the Time Axis Slider control at the bottom of the file will move to the relative position of the Wave file where you began your zoom operation. By using either the slider directly with your mouse, or by using the arrows at each end of the slider, you will be able to move through the original relative position of the entire Wave file, displaying the same level of time magnification that you had just established through the use of your Zoom-In control.


 Zoom Out

This key performs the inverse function of the Zoom-In key.  It allows you to progressively back out of a Wave file which you had previously Zoomed-In on.  Hitting this key allows you to backpedal through your last 5 zooms.   You can also access this feature by using the X hotkey or by using the Right Click Menu.  For a brief tutorial, see above.


Zoom Out Full

 This feature allows you to quickly zoom back to square one.  It will display your full waveform on the screen.  You can also access this feature by using the Ctrl + Z hotkey or the Right Click Menu.


Zoom to Markers

This feature is only activated when markers are dropped.  When used, it fills the screen with the material located between your two markers.


Sync Files

This feature, when enabled, mutually synchronizes the Filter and other commands such as Zoom-In and Zoom-Out between the Source and the Destination Wave files.  This feature is useful for selective filtering or viewing of a portion of a Wave file.  For example, if you run a particular filter, and then find a sector (or sectors) that needs further filtering, you can highlight only the portion that needs the additional processing, and apply the filter accordingly.  Sync mode will also accommodate different filters being applied to different sectors.  Also, when enabled, highlighting and Zooming-In on a particular area of a source file will correspondingly Zoom-In on the same timing co-ordinates in the Destination file.  The converse is also true such that Zooming-In on particular set of co-ordinates in a Destination file will also cause the Source file to contain the same Zoom-In co-ordinates.  This feature is of particular use when you want to visualize how a particular filter may have modified a waveform from your Source file.  Clicking on it with your mouse enables this feature.  When it is active, a check mark will appear to the left of it.  Note:  The Sync Files function is only useful in Classic Editing Mode.

To use Sync Mode to apply selective filtering to a Wave file, it is first necessary to create a Destination file from your Source file.  This can be done through the normal processing procedure of any of the filters, or it can be accomplished with one of the file conversion options such as Mono-to-Mono or Stereo-to-Stereo.  Thereafter, merely highlight the portion of the Source file that needs filtering, click on the appropriate filter, select the appropriate filter values and run the filter.  Only the highlighted sector of the two files will be enacted upon by the filter, with the results appearing in the Destination file.  If another filter (or the same filter that you had just been working with) is then required to be applied to another portion of the Wave file, just repeat the outlined process.

Non-Sync mode of operation

In non-sync mode, the highlighted section of the source file is read and processed by the filter.  The processed section is then written to the destination file starting at the beginning of the file.  If a destination file already exists, it will be overwritten (a prompt warns you of this).  This mode is useful when only a section of the source file needs to be extracted, or for testing a filter's settings before processing an entire file.


Selective Filtering with Sync Mode (Tutorial)

1.	Open the desired Source Wave file.

2.	From the Source Wave file, create a Destination Wave file utilizing one of the File Conversion alternatives or one of the FIVE/LIVE Filters.

3.	Activate Sync Mode by clicking on it in the View Menu.  A check mark will show that it’s active.

4.	Highlight the portion of your Source Wave file that needs Selective Filtering.

5.	Left click on the "Filter Menu."

6.	Left click on the filter that you desire to run on the selected portion of your Source Wave file.

7.	Adjust the Filter parameters accordingly utilizing "Preview Mode" if necessary to get the parameters to sound appropriate to your taste.

8.	Click on "Run Filter."

9.	The highlighted portion of the Source Wave file will be processed and inserted into the same time-slot in the Destination Workspace.

10.	If further selective filtering is required on a different portion of the Source Wave file, repeat steps 5 through 9 until you are satisfied.


Enabling or Disabling Toolbars In FIVE/LIVE

The following toolbars may be turned on or off by the user by simply checking or unchecking them from the View menu item.


Channel Toolbar

Just beneath the filter and effects toolbar and towards the left of the display can be found the Channel Selector Toolbar.  This allows you to choose which channels are to be operated on by the system. Your choices are as follows:

1. LR (Both channels)

2. L (Left channel)

3. R (Right channel

The Channel Selector applies to all Filters, Effects and Editing functions accept for tools that change the overall length of the file.  Generally, one would tend to leave this feature in the LR (Both channel) position unless dealing with special editing or sound restoration issues.    Note:  You can click on any of the Channel controls while playing to select either the Left, Right or Both channels.  When Previewing a filter, both channels will always be previewed, but only the channel selected before the preview starts will be affected by the filter.


File Toolbar

The main FIVE/LIVE File Toolbar resides along the top of the program window.  It contains the most commonly used file functions of the program.  Clicking on them with the left mouse button activates them.


Filter and Effects Toolbar

The Filter and Effects Toolbar also resides near the top of the program window.  It contains buttons for the most commonly used filters and effects. This toolbar, like the others, "floats" and can be dragged and dropped anywhere within the FIVE/LIVE window using the mouse.  Clicking on them with the left mouse button activates them.


Forensics Toolbar

The Forensics Toolbar contains buttons for the most commonly used Forensics filters.


Play Controls Toolbar

This toolbar contains the "Transport" controls for recording and playback of the audio.


Status Bar

The Status Bar resides along the bottom of the program window.  Four parameters are displayed therein:

•	Program Mode is displayed in the left-hand field.  Initially, it will indicate Ready.  Clicking on the various toolbar buttons (from left to right) will enable the following functions and indicate the following operation on the status bar:

A.	Open an existing Waveform audio file

B.	Save the active Document

C.	Delete a Section of a Wave file

D.	Copy selection and put it on the clipboard

E.	Paste Clipboard contents over the selected area

F.	Context Sensitive Help

G.	Rewind file position to beginning

H.	Pause Playback of the selected file

I.	Fast Forward to the end of the file

J.	Start Recording an Audio file

K.	Stop Playback of the Current file

L.	Play the selected File

M.	Zoom-In to the selected area of the sound file

N.	Zoom-Out to the last selected sound file

Note:

Clicking on the various menu items will also activate a one-line help text file, which describes each item.

•	Channels Used: The field to the right of the "Program Mode" field shows the number of channels being used and will indicate either mono or stereo.

•	Sample Rate: The next field moving towards the right shows the sample rate in KHz.

•	Bit Rate: The next field moving towards the right shows the bit rate of the current file.

•	Highlighted Length: The next field tells you the length of time for a highlighted portion of the Source or Destination workspace.

•	Available Hard Drive Space: The next field indicates the Available Hard Drive Space in Mbytes.  This is useful to determine whether or not there is enough storage space to perform your processing.


File Info

The following information will be displayed regarding the current highlighted Wave file when "File Information" is clicked:

1.	File Name

2.	File Path

3.	File Size (Bytes)

4.	Length (Time)

5.	Channels (Mono/Stereo, etc.)

6.	Sample Rate (KHz)

7.	Bits Per Sample

8.	Last Modified (Date)


Rebuild Peak File

Peak files (.pkf) provide a graphical representation (1 sample per 200 and extracting the peak value in that window) of the Wave files and are stored along with any Wave file created or operated upon by FIVE/LIVE.  After many editing processes, it may be desirable to update this file to more accurately represent the present state of the displayed Wave file in the source or destination workspace.  To do this, click on this function, and you will see that the displayed Wave file is updated, accurately representing your edits.  Peak files may be deleted by the user at any time.


The Window Menu

The Window Menu contains all of the commands that affect the manner in which the files are displayed to the user.  The following commands are available:


Cascade

The "Cascade" command arranges all of the open Windows so that they appear on the display one behind the other.  The most recently opened Window will appear closest to you, with the remainder arranged in the order in which they were opened.


Tile

The "Tile" command arranges all of the open Windows so that they appear on the display in a matrix configuration (in the same plane) on your display.  This feature is used when you need to see all of the workspaces concurrently.


Arrange Icons

The "Arrange Icons" command allows you to use the left mouse button to arrange the "iconized" windows in the pattern in which you would like to see them.


Close All

As the name implies…click here to close all windows.  The system will verify your choice, ask if you want to save and then close each window.


Open Files

A listing of all of the open files is provided at the bottom of the Window Menu.  The file that is active will have a check mark indicated to the left of the file name.


The Help Menu

With this product, we strive to provide you with the best possible help.  To that end, in addition to this in-depth manual, we also include a computerized help file that is always available at the click of a mouse.


Tip of the Day

This feature is normally displayed on the splash screen.  It can also be invoked under this menu item.  It provides useful hints on the operation of the software.


Restoring a Recording

This feature links you to a section of the help file that describes in step-by-step details regarding the restoration of an old recording.


Restoring the Demo

This feature links you to a section of the help file that describes in step-by-step detail the process for restoring the demo Wave file that is included with the software package.  This is an excellent exercise to go through to get an initial idea of the operation of the system.


Contents

The FIVE/LIVE program contains an extensive Help file.  These topics are printable from within the help file for your convenience.  The following types of information are available:

•	FIVE/LIVE program Overview

•	Tutorial Information on each filter

•	Step-by-Step procedures for utilizing each filter and editing feature

•	Audio Restoration process procedures

•	Practical Hints on recording procedures

•	Extensive Glossary with definitions, tables of values, and appendix information.

•	Licensing Agreement, Warranty, and program History information


Context Sensitive Help

This button will provide you with on-line context sensitive Help from the FIVE/LIVE Help file.


Using Context Sensitive Help (Tutorial)

Method #1

1.	Point the mouse pointer to the portion of the  FIVE/LIVE program display about which you would like some Help file information.

2.	Depress the special function key F1.

3.	Help file information will appear in a window.  This information is usually only summary procedural information on the particular topic.  For more detailed information, refer to the actual Help file, and look for tutorial information on the topic of interest.

4.	When done, click on the  - button at the top left-hand side of the Help window.

Method #2

1.	Using the left mouse button, click on the button with the "?" contained within its perimeter on the toolbar.

2.	A black, mouse controlled "?" will appear next to the mouse pointer.

3.	Move the pointer to the area of the program display about which you would like some Help file information.

4.	Click on the left mouse button.

5.	Help file information will appear in a window.  This information is usually only summary procedural information on the particular topic.  For more detailed information, refer to the actual Help file, and look for tutorial information on the topic of interest.

6.	When done, click on the  - button located at the top left-hand side of the Help window.

Note:

Under some circumstances, Method #2 will not activate the context sensitive Help file.  When this occurs, use method #1.

The Help file search engine is broken down into two categories they are:

Search for help on

This allows you to search for a topic or view the topics listing in alphabetical order.

Important Note:

Features which have not been implemented in your particular version of FIVE/LIVE but which may be provided in different releases of the program will be described in italics.


About Dcart

The following information is obtainable under the "About DC-Art" feature:

•	Program Name and Version #

•	Copyright Information including Copyright Dates

•	Upgrade Contact Information

•	Program Registration Information

•	System RAM Size

•	Hard Drive Disk Space Available for use


Analog tape recording transfer tips

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP

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Tape Dropout Repair (reel-to-reel):

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Enhancing old reel-to-reel tapes recorded at slow speeds

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Archival Recording Philosophy & Methodology

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


CDR Prep from a commercial cassette tape

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


CDR Prep from a vinyl record

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Characterizing frequency response/equalizing audio response (LIVE only)

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Convert white noise into pink noise

1.	Under the Edit menu, click on "Make Waves".

2.	Select "Random" noise, with the amplitude set to -10 dB, and the length set to 5 or 10 seconds.

3.	Click "OK", and a random white noise file will be created in the Source window.

4.	Click on the "Filter" Menu and then the Paragraphic Equalizer.

5.	In the preset menu at the bottom of the Equalizer, find and click on the "White to Pink noise Converter" and then run that filter.

6.	Pink noise will now be found in the Destination window.


Declicking a vinyl LP record

This procedure makes use of the Expert Impulse Noise Filter.  You may also want to try our new powerful EZ Impulse Noise Filter for quick and pain-free editing.

1.	Download the Vinyl LP Wave file into FIVE/LIVE using the "Open" command.

2.	Highlight the portion of your Wave file on which you desire to apply the Impulse Noise Filter.  (You may also choose to highlight the entire file.)

3.	Click on the Filter Menu and then "Expert Impulse Noise".

4.	Click on the "Vinyl LP" checkbox.

5.	Set the Threshold Control to its minimum setting of 1.

6.	Set the "Tracking" Control somewhere in the 25 to 30 range to start.

7.	Set the "Size" control to a value somewhere in the range of 10 to 15 samples.

8.	Run the filter or use Preview mode, and determine if the Wave file is being adequately de-clicked.  If it is not, lower the tracking control.  If it is de-clicking, but producing distortion on the sibilant sounds, then raise the control. Continue this process until a good balance is established of minimum sibilant distortion and minimum click feed-through.  If it is de-clicking, but leaving a bit of an artifact behind, increase the value of the "Size" control.

9.	When you determine the best setting of the control for your particular Wave file, click on Run filter.  When the filter has completed its operation, the results will appear in the "Destination" workspace.

Important Note: Vinyl LP mode works best on Wave files that have been sampled at 44.1KHz.  The setting examples given below are based on Wave files that have been recorded at a 44.1KHz sampling rate only.

Important Note 2:  Unlike all of the other  FIVE/LIVE controls, Vinyl LP mode cannot be turned on or off "live" in Preview Mode.


Manually De-clipping an over modulated Wave file

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Decoding TDMF(touch tone) signals into their alpha-numeric values

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Forensic Tape Authentication

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Gain Riding Procedure

1.	Highlight the portion of your Wave file during which you would like to "slew" (change) the gain up to or down to a new value.  (If this is to be done on a relatively small portion of the Wave file, it may be useful to Zoom-In on the sector of interest.)

2.	Press the Play button to audition your selection.

3.	Click on either "Fade-In" or "Fade-Out." (Edit Menu)

4.	Choose the type of slew curve you prefer, either Linear or Logarithmic.

5.	Set the "Start Level" slider control to unity gain.

6.	Set the "Stop Level" slider control to the desired new gain for the segment of interest.  This can be up to + 6 dB (amplification) or up to - 96 dB (attenuation).

7.	Click on OK.  The gain "slewing" will be performed during the highlighted portion of the Wave file.

8.	Highlight the portion of the Wave file that you desire to remain at the new gain setting.  This will be the "dwell" sector for the gain riding procedure.

9.	Again, click on either "Fade-In" or "Fade-Out."

10.	Next, change the "Start Level" slider to the same value as the "Stop Level" slider.  This will be your new gain setting (dwell).

11.	Click on OK.  The gain modification will be applied to the highlighted portion of your Wave file for its highlighted duration or dwell time.

12.	Next, select the portion of your Wave file during which you desire the gain to slew back down to its original value (or another new value).

13.	Again, click on either Fade In or Fade Out.

14.	Change the "Stop Level" slider back down to unity gain (or whatever new gain you desire) leaving the "Start Gain" setting where it had been.

15.	Click on OK.  The gain will slew back to the unity gain value during the highlighted time interval.


Selective De-Clicking with the Impulse Filter and "Sync Mode."

1.	Place FIVE/LIVE into "Sync" Mode (View Menu).

2.	Automatically De-Click the entire Wave file in the standard manner utilizing the Expert or EZ Impulse Filters.

3.	Listen to the result in the Destination Workspace.  Listen for locations that contain pops or clicks which were too severe for the algorithms to conquer.

4.	Zoom-In on the pop or click that you desire to remove.

5.	Lower the setting of the Threshold Control (or the Tracking Control if you are working with Vinyl LP's), and run the filter.  Observe in the Destination Workspace to see if the increased sensitivity was able to remove the impulse.

6.	Keep repeating the above process until you are satisfied that the impulse pop or click has been removed and replaced with a reasonable looking and sounding replacement waveform.


Nudging the Highlighted portion of the workspace

1.	To Nudge the right-hand side of a highlighted portion of a FIVE/LIVE Wave file, merely use the left and right arrow keys on your keyboard.

2.	To Nudge the left-hand side of a highlighted portion of a FIVE/LIVE Wave file, depress the Shift key while operating the left and right arrow keys.

3.	To change the resolution of the Nudge feature (the number of samples per nudge) go the preferences section of the Edit Menu, and enter the desired value of samples for each nudge.


Record transfer to hard drive technical hints

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Removing a lead vocal from a stereo recording

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Restoring an old 78 rpm recording  - In Depth

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Restoring a recorded telephone conversation

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Rumble reduction

Rumble is a low frequency random noise found on record recordings.  Generally, the noise in the spectrum below 20 Hz is considered rumble.  However, with very old 78 and 80-rpm recordings, rumble can often be heard with higher frequency content.  FIVE/LIVE provides you with three possible methods for substantially reducing rumble on a recording.  But first, it is of course important to perform your transfers with a low rumble turntable to begin with.  There is no sense in making the problem worse by using a cheap turntable (especially the types which use a thrust bearing which is of the ball race variety).  Turntables whose platters are thrust supported on a single ball (or point) will offer the lowest degree of turntable rumble.

Method #1:

If you are transferring a lateral 78, or a lateral monophonic LP or Monophonic 45 RPM record, transfer it to your hard drive via a stereo cartridge.  Then, using the File Conversions feature, convert the file from "Stereo" to "Mono L+R."  Since most of the rumble on these types of records are contained in the vertical vector only, the Mono L+R algorithm will cancel out this noise signal, and only preserve the lateral (horizontal) component, which does not contain rumble.

Important Note:

This technique will not work on Hill and Dale or Pathe´ (groove width modulated) or stereophonic recordings.

Method #2:

Use the Continuous Noise Filter as prescribed in an earlier tutorial.  It is only important to realize that the signals you are dealing with will be those at about 60 Hz on down.  So when you experiment with the threshold line position, only adjust it in the lowest octave to eliminate rumble.  When adjusted correctly, you should be able to preserve the last octave of the audio band in terms of actual signal, while rejecting the lower signal level rumble.

Note: This technique will work on all types of recordings, not just monophonic lateral recordings.

Method #3:

Use the High-pass Filter, with settings of 60 Hz on down, and the slope control set to 18 dB / Octave.  Experiment until you achieve the desired results using preview mode.

Note: This technique will also work on all types of recordings, not just monophonic lateral recordings.  However, it is not as good as method number 2 because it will also eliminate some of the low bass of your recording as well as its rumble content.


Simulate stereo from a mono source

1.	Open the Monophonic File to which you desire to add "Stereo Simulation."

2.	Click on "File Conversions, found under the Filter Menu.

3.	In the "Source File" box, click on "Mono."

4.	In the "Destination File" box, click on "Stereo."

5.	Click on "Run Filter."

6.	When the conversion process is completed, close the File Conversions dialog box

7.	Under the File Menu, click on "Make Destination the Source."

8.	When the "Save As dialog box appears, click on "OK" if you are satisfied with the automatically assigned temp file name.

9.	Under the Effects Menu, click on "Reverb."

10.	Using the "Preview" mode button, listen to the effect that the reverb is having on the file.  Adjust the various reverb parameters until you achieve the desired stereo effect.

11.	When you are satisfied with the reverb settings, click on "Run."


Manual Splitting & recombining a stereo Wave file

Though we have recently implemented a new automated Split and Recombine tool, some users may prefer performing this process manually.

1.	Bring the desired stereo file up into the Source Workspace.

2.	Under the Filter menu, select the File Conversions feature.

3.	Click on the "From Stereo to Stereo" function.

4.	Lower the Left Amplitude control to -96 dB (control all the way down).

5.	Run the Filter

6.	Make the Destination the Source and then rename the file.  (It may be a good idea to add an extension to the file name denoting that it is the right channel only)

7.	Double click on the Icon just to the left of the File Menu selector to return you to the window that contains the original stereo Wave file located in the source workspace.

8.	Double click on the Source Workspace

9.	Lower the Right Amplitude control to -96 dB (control all the way down).

10.	Run the Filter again.

11.	Make the Destination the Source again renaming the file.  (This time you may consider adding an extension to the file name denoting that it is the left channel only).

12.	Perform whatever independent signal processing you desire on each of the two files (channels).  If only one channel requires processing, just process that particular Wave file.

13.	When you are done with the independent channel processing, bring up the final version of the right channel into the source window.

14.	Under the Edit menu, click on "Copy."

15.	Bring the final form of the left channel into the source window.

16.	Under the Edit menu, click on Paste and then Paste Mix.

17.	Click on OK - the resultant file in the source workspace is the re-combined stereo Wave file.


Using FIVE/LIVE as an audio waveform analyzer

FIVE/LIVE can be used as a waveform analyzer.  It is capable of both time and frequency domain analysis.  This waveform information may be useful to know when trying to determine what useful frequency range is present in a Wave file.  It also will allow you to measure resonance or acoustic feedback, and periodic noise sources such as 50 or 60 Hz Hum (or harmonics thereof).  Once this information is known, it will be easier to set up the parameters for the Notch, High-pass, Low-pass or Band-pass Filters, so that you can retain useful portions of the spectrum, while rejecting unwanted noise signals.  This information is also useful in Forensic applications for measuring the time between gunshots or other events of critical importance.  The waveform's, either time or frequency domain can be printed out using the Print Screen procedure.

THE REST OF THIS SECTION IS NOT INCLUDED IN THE DEMO HELP


Trouble Shooting

This is your one stop shop for problems or questions that you may encounter while working with our software.  This list of problems and solutions constitutes about 90% of the overall problems our techs deal with.  Chances are…if you have a problem with our product, the solution is listed below.  Please read this before contacting us.

This section lists common questions and problems users have with FIVE/LIVE.

Q: How do I control the recording level of the audio signal?

A: In Windows, there is a speaker Icon in the lower right hand corner of the Taskbar. Double-click on this Icon to bring up the control panel for your sound card. There are level controls for the Mic or Line inputs of the sound card.   For Sound Blaster type cards, double click the yellow speaker icon and then choose Options/Properties.  Click on Recording, and then click OK.  You'll now have your input mixer up.  Also be sure that the correct input is selected (Mic, Line, or Aux) for your particular recording setup.

Q: I’m trying to record and I hear the audio, but the Record meters are not jumping.

A:  Most sound cards come with the Line Input not selected.  This causes a "no recording" symptom.  But first, make sure that the sound card is installed correctly.  Just go to the Control Panel/Multimedia and make sure that the sound card is selected as the default playback and recording audio device.  If it is, the sound card is most likely working okay.  Next, bring up the Preference dialog box and select the "Sound Card" Tab and make sure that the inputs and outputs are set to the correct sound card.  If the system still is not recording, double click on the yellow speaker icon on the tray and choose Options/Properties and click on the Recording radio button.  Click OK.  This brings up the input mixer.  Make sure that the slider labeled "Line Input" is selected (it usually will not be).  When you do this, the meters in the Record window will start modulating immediately, indicating that the system is now working fine.

Q: I get a Windows error message when I try to record that says something about "this is beyond the capability of this device" or "The specified format is not supported".  What's this?

A: This just means that your sound card is not capable of recording with the sample rate or bit width you have selected.  For example, if you try to record a 24-bit file with a 16 bit sound card, you'll get this message.  Most sound cards will work with 44.1khz, 16 bit, stereo recording settings.  This could also mean that Windows is assigning Wave files to be played by some device other than a sound card…like a modem.  Make sure Windows is aware that you want Wave files assigned to your sound card’s output.

Q:  I click on Play and don't hear anything but the VU meters are jumping fine in the program.

A:  If the cursor is moving across the screen during playback and the VU meters are jumping, you can be pretty sure the audio is indeed playing.  Usually, this just means that you don't have your speakers plugged into the soundcard output where the audio is appearing.  Check under Edit/Preferences/Soundcard to see what the output device is set to and make sure your speakers are plugged into that set of outputs.

Q: My levels are low during recording.

A:  Maybe not.  You should normally not record all the way up to 0db (the red area) with digital systems.  Remember, you'll likely need some "head room" in order to be able to increase volume with some of our tools such as the Paragraphic EQ.  Peak levels of -4 or –6 db will give great results.  If your levels are REALLY low, check to make sure you have the right type of signal going into your sound card.  You cannot plug a turntable directly into a soundcard, for example - you'll need a preamp for this.

Q: FIVE/LIVE now contains a Fast Edit mode and the classic two-file mode.  Which one should I use?

A:  Beginners tend to like the Fast Edit mode since it works more like other audio editing programs.  It's intuitive, simple and fast.    However, as you learn the program we think you'll find the classic two-screen mode to be the most powerful audio restoration environment.  Our "Source and Destination" way of working is optimized for those situations where you want the best restoration possible and need to apply various filters to different portions of a file.

Q: I clicked Fast Edit in the Preference box and I'm still in the Classic mode.

A:  You need to close the program and restart it for this change to take effect.

Q: I am getting crashes when I try to play audio.  What do I do?

A:  FIVE/LIVE uses the new Windows Driver Model (WDM) method of communicating with sound cards.  This is new to many sound card manufacturers and some of these new sound card drivers may have bugs in them.    If your are getting a crash when you try to play or preview a filter, please check to see if a new sound card driver is available from the manufacturer of your card.  If this doesn’t do the trick, look under Edit/Preferences/Sound Cards and check the box which says "re-initialize on play".   This makes the sound card return to a known state every time you play and may help you.

Q: Why does Preview mode sound like it is stuttering?

A: All of the filters in FIVE/LIVE require a fair amount of processing power. If your computer cannot complete the processing fast enough to keep up with the audio stream, then the preview mode will stop and start in short bursts that sound like stuttering. This effect can be reduced or eliminated by increasing the number of "Preview Buffers" in the Edit/Preferences/Soundcard dialog box.   Regardless of stuttering in Preview, all "run filter" operations will be stutter free.  Putting many filters in the MultiFilter box will likely cause stuttering on even very fast machines.  You may also want to check and see that you’re using your sound card’s newest drivers.

Q: I noticed that a Live Mode is now included in FIVE/LIVE.  What's this for?

A:  This allows you to take a "live" audio stream and process it without first recording it to hard disk.  With this function, you can make your computer an integral part of your home stereo system, ham radio station or SWL station.   Imagine adding the sound of a $15,000 vacuum tube amplifier to all your audio without having to record and process it.  As an example - take your CD player and plug its outputs into your sound card input.  Take the sound card output and plug it into your home stereo receiver.  Now run FIVE/LIVE and choose MultiFilter.  Put some enhancement modules and/or noise filters into the signal path and click on "live preview".  All these tools will be applied to every CD you play in real time.

Q: I get an error when I try to use Live Preview mode.  What's up?

A:  Your soundcard must be able to play and record at the same time.  If you have a cheap soundcard, it's time to upgrade!

Q: I have a Whackmaster 6000 sound card I bought in 1992.  Is this OK?

A: Who knows?  We don't remember that year at all.  Just about all of the new sound cards include a new type of driver called WDM.  This is Microsoft's effort to fix a lot of earlier problems with various sound card implementations.  If your really old soundcard works, then you're OK.  If not, it's time to invest in more modern hardware that will also likely offer much better audio and lots of new features.

Q: I have a 192kHz 24 bit sound card.  Should I record using these settings?

A: Remember, your recordings will never be better than your originals.  If you are recording vinyl records and are making CDs, we recommend 44.1, 16, stereo.  If you have high quality masters or are making DVD audio disks, then go with the higher settings.  FIVE/LIVE doesn't care how you record, but higher sample rates and wider bit widths will result in larger files and more processing time.

Q:  How do I record my records using a turntable?

A:  A phono cartridge puts out a very tiny signal.  You cannot plug a turntable directly into your soundcard since it doesn't amplify the incoming signal enough.  You'll need a preamp.  If you have a stereo receiver with a control labeled "phono" on the front, you're all set since this has a preamp built in.  If your receiver doesn't have this (many modern ones don't) or your receiver is too far from the computer, get a stand-alone phono preamp.

Q: How do I connect my computer into my existing stereo system?

A:  On your receiver or master component, take the "tape out" or other line level output and plug it into the line level input of your soundcard.   You can also take the line out of your soundcard and plug it into the "tape in" of your receiver though most folks just connect the soundcard output to their computer speakers.

Q:  My computer crashes in Preview mode with the MultiFilter.

A:  If there is any weakness in your system, a MultiFilter preview will likely find it.  This can require thousands of calculations per second (or more) while audio is flowing.  The audio can be presented to your sound card in a start/stop fashion as well (though the buffering on your sound card and in FIVE/LIVE makes for smooth playback).  This environment is seldom tested for by sound card designers and you can uncover bugs in the drivers.  Make sure you have your latest sound card and video card drivers and try reducing the amount of video acceleration your video card is set for (this reduces the overall system overhead a bit).

Q: Will increasing the amount of RAM in my computer make FIVE/LIVE run faster?

A: FIVE/LIVE does not require huge amounts of RAM. If your computer system has 128 Mbytes of RAM, then further increases will not appreciably speed up the program. The software uses disk-based processing so hard drive speed and raw processor speed will generally have a greater effect than increased RAM beyond a certain minimum.  A faster processor WILL make a big difference in the number of simultaneous filters you can use.

Q: I want to purchase a system optimized for use with the software.  What sort of system will provide me with the fastest performance of all of the various FIVE/LIVE filters, effects, and editing features?

A: You should purchase a computer with the fastest clock speed in your price range.  Look for hard drives with fast access times and rotational speeds.  Purchase a high performance sound card.  Lastly, turn off all superfluous programs that may be running in background.

Q: How do I avoid producing dropouts during recording or playback?

A:  Make sure that you have reviewed all of the following:

1.	Make sure that you are using the latest drivers for your sound card.  They can usually be obtained from the card manufacturers or Microsoft's web site.

2.	Make sure that the screen saver and all power management functions will not kick-in during recording or playback.  By default, the screen saver has a 1-minute timeout, so after 1 minute of no keyboard or mouse activity, the screen saver will kick-in.  This flurry of disc activity will put a glitch on the recording or playback of Wave files.

3.	Turn off all power saving features, or set their timers to a value of time greater than the longest musical selection that you want to record or play.

Q: I have a vinyl LP that is very noisy, and still has too many clicks after processing.  What can I do?

A: Try running the Ez-Impulse filter or running the Expert-Impulse filter twice or more.  First run it with the Tracking control set to zero, and adjust the threshold control to remove just the largest clicks.  When done, make the destination the source, and re-run the filter with the Threshold set back to 1, and adjust the tracking control to get the smaller clicks. Another thing to try is to use the File Reverse feature, and then process the vinyl recording through the Impulse Noise filter.  When done, re-reverse the file.  Also, consider trying HQ mode, which is especially good for detecting very small clicks when adjusted correctly (small size values around 5).  If you still have clicks left, it's likely because the clicks are longer than the program is allowed to automatically identify and repair.  Use the Interpolate function to get rid of any remaining clicks.

Q: I have a record with one major scratch on it running from the center to the outside in a spiral pattern.  How do I eliminate the loud "click" which I hear which occurs once per revolution of the record?

A: You should use the Impulse Noise filter in the following way.  Set the Tracking control to its minimum value.  Set the Threshold to its maximum value.  Set the "samples" to about five.  While in Preview mode, slowly decrease the Threshold control until you see the click counter increment once for each click, which is occurring.  Do not increase the Threshold control any further than necessary.  Next, increase the "Samples" control until the click is not longer audible.  This technique is also useful for getting rid of the clicks produced by cracked 78-RPM shellac records.  It is even possible to take a broken 78, glue it together, and then after transferring it, remove the clicks, which occur at the breakage points.

Q: I just transferred an old stereo recording to my hard drive for processing, but I am not sure whether or not the left and right channels are in their correct positions or transposed.  How can I determine if they are correct?

A: If the material is contemporary pop, it is difficult to make this determination.  However, if the music is orchestral in nature there are some obvious orientations to listen for.  For example, the left channel should be dominated by any cymbal crashes or the tinkling sound of a triangle.  Also, when Violins come in they should dominate the Left channel. Double basses should be more prominent on the right side of the soundstage. French horns should be heard behind the woodwinds, while trumpets and trombones should generally project from the right rear. If this is not the case, the channels are probably reversed and should be corrected by applying the File Conversion Filter Stereo Reverse feature.

Q: Some of my vinyl records sound more distorted as the play moves towards the end of a side.  Why is this?  What can be done?

A: This distortion can be due to several sources of problem.  A poorly mastered lacquer could have been the culprit.  Not all mastering engineers were careful to "hold down" the dynamic range as the recording progressed, creating a great deal of groove wall over-modulation.  However, a more likely scenario is that the record was played with a worn stylus.  Have someone who is familiar with styli check it out under a microscope to be sure that it is not worn out or damaged.  If it checks out to be okay, then your vinyl recordings may have been damaged in past times as a result of being played many times with a defective stylus.  One thing that can be done is to selectively apply a Low-pass filter to the last few cuts of the recording to reduce this annoying anomaly.

Q: How do I generate a simulated stereo Wave file from a monophonic Wave file?

A: Start with a monophonic file that has been de-noised, and convert it to stereo using the File Conversion Filters.  Some stereo effect may be added here by applying a little "Time Offset" during this process.  Next, make the destination the source, to get a new source file.  Run the Reverb effect with a Small or Medium hall, setting the decay to a low number and the early reflections level nearly to zero.

Q: Does the order in which I process noise out of a Wave file matter?

A: Yes.  Always remove clicks and pops with the Impulse Noise filter before de-hissing a recording using either the Continuous Noise filter or the Dynamic Noise filter.  Never reduce the bandwidth of an audio signal before applying the Impulse Noise filter (and this applies both in the analog side of your signal path as well as the digital side.)

Q: I have an analog tape recording with clipping distortion due to over-modulation during the recording process.  Is it possible to "soften" the clipping sound in order to reduce the harshness produced during the overloads?

A: Clipping distortion can usually be substantially reduced by applying the De-clipping filter.  This is the preferred method.  However, clipping distortion can sometimes also be reduced by utilizing the Impulse Noise filter.  If the clipping distortion occurs at the peaks of the waveform, set the tracking control set to its minimum value, and the threshold set to maximum. Highlight a segment of the recording that contains distorted and non-distorted material.  In preview mode and with accuracy optimization checked, adjust the Threshold control until the clipping distortion is reduced.  In some cases, it may become necessary to run the Reverse NAB curve before following the above procedure.  After the distortion has been reduced with the impulse filter, it will be necessary to run the NAB curve to re-correct the recordings equalization.  These two curves can be found in the factory preset listing under the Paragraphic Equalizer filter.  The reason for the above two steps is that the saturation overload occurs at the tape-to-tape head interface.  The resultant overload is then phase shifted during playback by the NAB equalization circuit in the tape recorder.  Utilizing the Reverse NAB curve places the clipping distortion closer to the peak of the waveform, where is actually occurred.

Q: I enjoy removing "clicks" and "thuds" manually from my recordings using the Interpolate feature.  Sometimes these signals are so small that, although I can hear them, I can't see them in the waveform.  What can I do to make them more visible?

A: In order to improve the view of these signals, first put the system in Sync mode which is found under the View menu.  In order to see the "clicks" more clearly, run the High Pass filter with it set to 6 dB / Octave and 10 KHz on your Wave file.  The "clicks" will now be more visible in the Destination window and they will be time aligned with the Source Window.  Similarly, if you want to view "thuds", run the Low Pass filter with it set to 6 dB / Octave and 100 Hz on your Wave file.  As with the clicks, the "thuds" will be more visible in the Destination Window.


Reporting a Problem

Enhanced Audio has operators available during most days from 10:00 AM to 4:00 PM EST.  We provide technical support for registered users only.  First contact is usually handled via email at support@tracertek.com. When you contact us, please include:

•	Sound Card type

•	Version of the software (Found Under Help/About Software)

•	Version of Windows

•	Being specific as possible, outline the problem, how it occurs and its frequency

The reason we prefer email for first contact, is that it gives our techs a chance to study the problem here, before we give you an answer, rather than experimentation while you’re waiting on the phone.  We normally have an answer for you within the next business day.


Contact Information

Enhanced Audio Inc.

3600 Board Rd.

York, PA  17402

Phone 717 764 9240

FAX   717 764 9254


Tracer Technologies

Exclusive worldwide distributor of the Enhanced Audio product lineup as well as quality sound cards, software, preamps, speakers, headphones, MP3 players, multi-channel recording devices, microphones and other musician’s tools.

Tracer Technologies

3600 Board Rd.

York, PA  17402

717 764 9240

fax 764 9254

www.tracertek.com

info@tracertek.com


Hot Key Index

Here is a list of all known Hot Keys/Combos and their real world equivalent.

"1"            	Play selected area + 0.25 seconds on each side

"2"            	Play selected area + 0.5 seconds on each side

"3"            	Play selected area + 1 second on each side

"4"            	Play selected area + 2 seconds on each side

"A"            	Select displayed area (same as double click)

CONTROL+"A"    	Select entire file

CONTROL+"C"    	Copy selection to clipboard

"D"            	Select destination

CONTROL+ "E"   	Select/deselect the pencil (need to be zoomed in far

enough to see waveform)

"I"            	INTERPOLATE Full File

"J"            	INTERPOLATE LEFT channel only

"K"            	INTERPOLATE RIGHT channel only

"L"            	PLAY LOOPED

"M"            	DROP MARKER

CONTROL+"M"    	MUTE

"N"            	GO TO NEXT MARKER

CONTROL+"N"    	NEW File,

SHIFT+"N"      	GO TO PREVIOUS MARKER

CONTROL+"O"    	OPEN File

"P"            	Open PREFERENCES

CONTROL+"P"    	PRINT

CONTROL+"R"    	Brings up Record screen

"S"            	SELECT SOURCE

CONTROL+"S"    	FILE SAVE

CONTROL+"U"      	UNDO LAST EDIT

CONTROL+"V"       	PASTE

ALT+BACKSPACE	UNDO

SHIFT+DELETE    	CUT

END         	REWIND TO Start

F1          	Launch HELP,

SHIFT+F1	      	CONTEXT SENSITIVE HELP

F6 	         	NEXT PANE

SHIFT+F6        	PREV_PANE

HOME        	FORWARD TO END

CONTROL+INSERT  	EDIT COPY,

SHIFT+INSERT    	EDIT PASTE,

LEFT ARROW        	NUDGE RIGHT edge of selected area to the left

RIGHT ARROW       	NUDGE RIGHT edge of selected area to the right

SHIFT+LEFT ARROW   	NUDGE LEFT edge of selected area to the left

SHIFT+RIGHT ARROW  	NUDGE LEFT edge of selected area to the right

RETURN     	Default button (this is the cancel button on all filters)

SPACEBAR       	Play Wave

"X"            	ZOOM OUT

CONTROL+"X"        	CUT

"Z"            	ZOOM IN

CONTROL+"Z"        	UNDO

SHIFT+"Z"          	ZOOM OUT X4


Sync Mode/Non Sync Mode Explanation Chart

 The following diagram illustrates the standard filtering process of FIVE/LIVE.

Sync Mode

Sync mode is the default mode of operation for the Classic Editing screen of FIVE/LIVE. In Sync mode, both the Source and Destination files track each other. If you zoom into a section of the Source file, the Destination file will zoom to the same section. When you process the Source file using a FIVE/LIVE filter, the program reads the Source file, processes it, and writes it to the Destination file at exactly the same position as the Source file. This means that if you want to reprocess a section in the middle of a song, just highlight the section in the Source file that needs processing and run the filter again.

Non-Sync Mode

 In Non-Sync mode, the highlighted section of the Source file is read and processed by the FIVE/LIVE filter. The processed section is then written to the Destination file, starting at the beginning of the file. If a Destination file already exists, it will be overwritten (a prompt warns you of this). This mode is useful when only a section of the Source file needs to be extracted, or for testing a filter’s settings before processing an entire file.


Language translation (Deutsch/Español)

FIVE/LIVE program terminology translations from English to German and Spanish

English	Deutsch	Español
About Diamond Cut	Über DC-Art	Acerca del DC-Art
Accuracy	Genauigkeit	Acuracia
Add Marker	Marker hinzufügen	Añadir marcador
Advanced Controls		Controles avanzados
Amplitude (dB)	Amplitude (dB)	Amplitud (dB)
Arrange Icons	Symbole anordnen	Arreglo de Icones
Attack (mSec)	Anstiegsflanke (mS)	Atacar (mSec)
Attenuation	Dämpfung	Atenuacion
Average	Mittelwert	Promedio
Averaging Filter	Mittelwertfilter	Determinacion del Promedio del Filtro
Band Pass	Bandpass	Banda de frecuencias alimentadas
Bandpass Filter	Bandpaßfilter	Filtro de Paso de Banda
Bandwidth	Bandbreite	Anchura de Banda
Bright	Hell	Brillante
Bypass	Bypass	Sobrepasar
Cancel	Abbrechen	Cancelar
Cascade	Fenster kaskadieren	Cascada
CD Preparation	CD Vorbereitung	Preparacion del CD
Center Frequency	Mittenfrequenz	Centro de Frecuencia
Change Speed	Geschwindigkeit ändern	Cambio de velocidad
Chop File into pieces	Datei aufteilen	Recortar el fichero en pedazos
Clear all Markers	Alle Marker löschen	Despejar todos los marcadores
Click Filter	Klickfilter	Golpecito seco en el Filtro
Clicks / Second	Klicks / Sekunde	Golpecitos  en seco / Segundo
Close	Schließen	Fin
Close Destination	Zieldatei schließen	Cerrar Destinacion
Close Source	Quelldatei schließen	Cerrar las fuente de datos
Contents	Inhalt	Contenidos
Continuous Noise	Dauerrauschen	Sonido Continuo
Continuous Noise Filter	Rauschfilter	Filtrador Continuo de Sonido
Copy	Kopieren	Copia
Crossfade	Crossfade	Interseccion del aumento / disminucion del volumen alto
Curved	Kurve	Curvado
Cut	Ausschneiden	Cortar
Dark	Dunkel	Oscuro
DC-Art Progress	DC-Art Prozeß	Progreso del DC-Art
Decay	Ausklingzeit	Decaimiento
DeEsser	DeEsser	Normalizador de seseos
Delete	Löschen	Suprimir
Delete Files	Dateien löschen	Suprimir ficheros / archivos
Demo	Demo	Demostracion
Detail	Detail	Detalle
Device I/O Selection	Device I/O Auswahl	Seleccion del Dispositivo I / O
Drive	Drive	Impulsar
Dry	Trockenes Signal	Seco
Dynamic Noise	Dynamisches Rauschen	Sonido Dinamico
Dynamic Noise Filter	Dynamisches Rauschfilter	Filtro Dinamico de Sonido
Dynamics Processor	Dynamikprozessor	Procesor Dinamico
Early Level	Vorreflexion	Primera reverberacion de Amplificacion
Edit	Bearbeiten	Editar
Effects	Effekte	Efectos
Ending Pitch	Endtonhöhe	Terminando el Grado de Inclinacion
Even Only	Nur geradzahlige	Paridad Par
Exciter	Exciter	Excitador
Exit	Beenden	Salida
Expander/Gate	Expander/Gate	Ensanche / Puerta
Fade In	Einblenden	Aumento de Volumen
Fade Out	Ausblenden	Disminucion de Volumen
Fast Forward	Schneller Vorlauf	Advanzar / adelantar
File	Datei	Fichero / Archivo
File Conversions	Dateikonvertierungen	Traducciones del fichero
File Information	Dateiinformation	Informacion de Archivo
File Name	Dateiname	Nombre del fichero / archivo
File Toolbar	Datei Werkzeugleiste	Archivo de paneles
Files of Type	Dateityp	Modelos de ficheros
Filter	Filter	Filtro
Filter Freq.	Filterfrequenz	Filtro de frecuencia
Filter Harmonics	Oberwellen filtern	Filtro Armonicos
Filter Slope	Filterkurve	Filtro Inclinado
Filter Toolbar	Filter Werkzeugleiste	Filtro de instrumentos
Find and Mark Silent Passages	Stille Passagen finden und markieren	Encontrar y marcar los pasages silenciosos
Frequency	Frequenz	Frecuencia
Frequency Response	Frequenzgang	Frecuencia de respuesta
Frequency Spectrum	Frequenzspektrum	Frecuencia de espectros
From Mono to	Von Mono nach	Desde el Mono al
From Stereo to	Von Stereo nach	Desde el Stereo al
Full Range	Volle Bandbreite	Margen Entero
Gain	Verstärkung	Amplificacion
Gain Change	Verstärkungsänderung	Cambio de Volumen
Gain Normalize	Pegel normalisieren	Regularizar el sonido
Graphic Equalizer	Graphischer Equalizer	Equalizador Grafico
Harmonic Reject	Harmonische ausfiltern	Reyeccion Armonica
Harmonic Reject Filter	Oberwellen Unterdrückungsfilter	Filtro Armonico de Reyeccion
Help	Hilfe	Ayuda
High Frequency	hohe Frequenz	Frecuencia Alta
High Pass	Hochpass	Paso Alto
Highlight Marked Area	Markierten Bereich anwählen	Destacar las Areas Marcadas
Impulse Filter Parameters	Impulsfilter	Filtrar Parametros de Impulso
Impulse Noise	Impulsrauschen	Sonido de Impulso
Keep Residue	Rest behalten	Guardar residuo (resto)
Large	Groß	Grande
Left	Links	Mando izquierdo
Look In	Hineinsehen	Mirar En
Low Frequency	niedrige Frequenz	Frecuencia baja
Low Pass	Tiefpass	Avance de alimentacion de frecuencia baja
Lowpass Filter	Tiefpaßfilter	Filtro de pasaje bajo
Make Destination the Source	Ziel zur Quelle machen	Convertir los datos salidos del Fichero en datos nuevos de entrada de  Fichero
Make Waves	Wellenformdarstellung	Hacer ondas
Marker	Marker	Marcador
Maximum Harmonic	Maximale Oberwelle	Armonia Maxima
Median	Zentralwert	Media
Median Filter	Mittelungsfilter	Filtro Medio
Medium	Mittel	Mediano
Mix	Mischung	Mixto
Mono	Mono	Mono
Mute	Stummschalten	Mudo
No	Nein	No
Noise Threshold	Rauschschwelle	Umbral de Sonido
Notch	Notchfilter	Hindumieno
Octave	Oktave	Octavo
Odd Only	Nur ungeradzahlige	Paridad impar
Open	Öffnen	Abrir
Open as Read Only	Schreibgeschützt öffnen	Abrir tal como se lee - Solo
Open Destination	Zieldatei öffnen	Salida abierta de archivo
Open Playlist	Spielliste öffnen	Lista de titulos
Open Source	Quelldatei öffnen	Abrir la fuente de datos
Operating Point (Quiescent Point)	Arbeitspunkt	Punto de Ininiazion
Optimization	Optimierung	Optimizacion
Output	Ausgang	Salida
Output Mix	Ausgangsmix	Mezcla de salidas
Overload	Übersteuerung	Sobrecarga
Paragraphic Equalizer	Parametrischer Equalizier	Equalizador Paragrafico
Paste	Einfügen	Pegar
Pause	Pause	Pausa
Pause File	Datei anhalten	Pausa del fichero / archivo
Pentode	Pentode	5 Elementos
Percent Done	Prozent erledigt	Porcentage acabado
Pitch Range	Tonhöhenbereich	Margen del Grado de Inclinacion
Play Controls	Transportkontrolle	Uso de los Controles
Play File	Datei spielen	Activar el fichero
Play Looped	Endlosschleife	Ejecutar el registro y repetir
Preferences	Voreinstellungen	Preferencias
Preview	Vorschau	Vostazo/Chequeo
Quantize for CD Audio	Für CD Audio quantisieren	Cuantificar para el audio CD
Ratio	Verhältnis	Proporcion
Record	Aufnahme	Registro
Record File	Datei aufnehmen	Registro del fichero / archivo
Reflections	Reflexionen	Reflejos
Release (mSec)	Abfallzeit (mS)	Librar (mSec)
Reset	Zurücksetzen (Reset)	Reajustar
Reset Levels	Pegel zurücksetzen	Reajustar los Niveles
Restoring a Recording	Aufnahme restaurieren	Restaurando el Registro
Restoring the Demo	Demo restaurieren	Restauracion del muestreo
Reverb	Hall	Reverberacion / Eco
Reverse	Rückwärts	Inverso
Reverse File	Datei rückwärts abspielen	Invertir fichero / archivo
Rewind	Zurückspulen	Rebobinar
Rewind File	Datei zurückspulen	Rebobinar el fichero / archivo
Right	Rechts	Mando derecho
Room Size	Raumgröße	Tamaño de espacio
Run Effect	Effekt abspielen	Comenzar el funcionamiento de Efecto
Run Filter	Filter anwenden	Comenzamiento del filtro
Same Source as	Gleiche Quelldatei wie	El mismo dato que el
Sample Noise	Quantisierungsrauschen	Ejemplo de Sonido
Samples	Samples	Ejemplos
Save	Speichern	Guardar
Save as Type	Speichern als	Guardar tal como se ha mecanografiado
Save Destination as	Zieldatei spiechern als	Guardar destinacion como
Save In	Speichern in	Guardar En
Search for Help on	Suche Hilfe über	Busqueda de ayuda en
Select all	Alles auswählen	Seleccionar todo
Shape	Form	Form / Forma
Shift Threshold	Schwellwert verschieben	Cambio del Margen
Small	Klein	Pequeño
Source	Quelle	Fuente de Datos
Spectrum	Spektrum	Spektrum / Espectros
Spectrum Analyzer	Spektrumanalysator	Analizador de espectros
Speed	Geschwindigkeit	Velocidad
Starting Pitch	Anfangstonhöhe	Comienzo de Frecuencia
Status Bar	Zustandsleiste	Condicion de los paneles
Status Toolbar	Zustandsleiste	Compartimiento / Tablero de los paneles
Stereo	Stereo	Stereo
Stereo Reverse		Invertir el Stereo / retroceder atras Stereo
Stop File	Datei stoppen	Parar el fichero / archivo
Straight Line	Gerade	Linea Recta
Sweet	Süß	Agradable
Sync Files	Dateien synchronisieren	Sincronizacion de archivos
Threshold	Schwellwert	Margen
Tile	Fenster anordnen	Diseño de Azulejos
Time Offset	Zeitversatz	Cambio de tiempo
Tip of the Day	Tip des Tages	Consejo del dia
Total Clicks Processed	Anzahl der bearbeiteten Klicks	Total de golpecitos secos (pinchazos/clicks) procesados
Total Samples to Process	Anzahl der zu bearbeiteten Samples	Total de ejemplos para procesar
Tracking	Gleichlauf	Rastreando
Transformer Coupled Class AB	Transformator gekopplete Class AB	Transformador Acoplado de Clase AB
Triode	Triode	Triodo
Tube Type/Configuration	Röhrentyp/Konfiguration	Clase de tubo / configuracion
Undo	Rückgängig	Deshacer
Very Large	Sehr groß	Muy Grande
View	Anzeigen	Vista
Vinyl LP	Vinyl LP	Vinilo LP
Virtual Valve Amplifer	Virtueller Röhrenverstärker	Amplificador de valvula virtual
Warm	Warm	Calido
Wet	Effektsignal	Mojado
Window	Fenster	Ventana
Yes	Ja	Si
Zoom to Markers	Zoom zwischen den Markern	Enforcar rapidamente
Zoom In	Zoom In	Enforcar
Zoom Out	Zoom Out	Desenfocar
# of Filters	Filteranzahl	Numero de filtro
% Change	% Veränderung	Porcentage de Cambio
% Selected Time	% Zeit	Porcentage del Tiempo seleccionado
2 Stage Class A	Zweistufige Class A	2 Procesos - Clase A
2 Stage Class AB	Zweistufige Class AB	2 Procesos - Clase AB
German Translation: Courtesy of Konstantin Themelidis

Spanish Translation: Courtesy of Monica Sanz Aznar (Monica Hash)


Preset Listings

Adaptive Filter

1.	Basic Forensics Adaptive Filter

2.	Basic Forensics Adaptive Filter with Unity Convergence

3.	Basic Normalized Adaptive Filter

4.	Binaural Filter with Left Channel Reference

5.	Binaural Filter with Right Channel Reference

6.	Sharp Adaptive Filter with delayed Monophonic Reference

7.	Warbling Tone Rejection Filter

Averaging Filter

1.	Heavy Handed Averaging Filter

2.	Half Maximum Value Averaging Filter

3.	Maximum Value Averaging Filter

4.	Simple Cylinder Recording De-Hisser

5.	1950's Jukebox Sound

6.	Light Surface Noise Remover

Band-pass Filter

1.	1930 Vintage Table Top Radio

2.	1950's Vintage Juke Box

3.	AM Pocket Transistor Radio

4.	Loud Walkman heard by person nearby

5.	Modern "High-End" Audio System

6.	Modern Cheap Table Top Radio

7.	Night Club as heard in Parking Lot

8.	Olde Acoustic Phonograph

9.	Public Address System at Outdoor Event

10.	Speech Filter

11.	Stereo System heard in adjacent room

12.	Telephone "off-the-hook" sound

13.	Megaphone

14.	Metal Horn based Acoustical Phonograph

Brick Wall Filter

1.	1000 Hz Band-Stop Filter

2.	200 Hz High-pass Brick wall filter

3.	5000 Hz Low-pass Brick Wall Filter

4.	6000 Hz Low-pass Brick Wall Filter

5.	9000 Hz Low Pass Brick Wall Filter

6.	Alpha Brainwave Bandpass Filter

7.	Beta Brainwave Bandpass Filter

8.	Theta Brainwave Bandpass Filter

9.	Theta to Beta Brainwave Bandpass Filter

10.	Basic Forensics Speech Filter

11.	60 Hz Band-stop Filter

12.	Gentle Slope Forensics Speech Filter

13.	Steep Slope Forensics Speech Filter

14.	DTMF Band-pass Filter

15.	Sub-Sonic Filter (Steep)

Change Speed Filter

1.	Fractional Speed Mastering: 33.3 to 45 RPM

2.	Fractional Speed Mastering: 45 to 78.2 RPM

3.	Fractional Speed Mastering: 45 to 78.8 RPM

4.	Fractional Speed Mastering: 45 to 80 RPM

5.	Fractional Speed Mastering: 78.2 to 80 RPM

6.	0.5% Speed Decrease

7.	0.5% Speed Increase

8.	1.00 % Speed Decrease

9.	1.00 % Speed Increase

10.	1.50 % Speed Decrease

11.	1.50 % Speed Increase

12.	3.00 % Speed Decrease

13.	3.00 % Speed Increase

14.	6.00 % Speed Decrease

15.	6.00 % Speed Increase

16.	1.00 % Pitch Shift Upwards

17.	1.00 % Pitch Shift Downwards

18.	Rise and Fall

19.	Sin Wave Function

20.	Sin Wave Function – Small change

21.	Tape Recorder VOX Start Compensation

22.	Tape Recorder VOX Stop Compensation

Channel Blender

1.	Early Stereo Anti "Ping-Pong" Effect

2.	FM Stereo Multi-path Distortion Filter

3.	Stereo Ambience Signal Only

4.	Vinyl LP Bass Clarifier

5.	FM Stereo Noise Filter

6.	Lead Vocal Attenuator 1

7.	Lead Vocal Attenuator 2

8.	Lead Vocal Attenuator 3

9.	Vinyl LP Rumble Cancellation Filter

10.	Severe FM Stereo Noise Filter

11.	Dreamy Veil

Continuous Noise Filter

1.	Flat-line Response

2.	CrO2 (High Bias) Cassette Tape Fingerprint (no NR)

3.	CRO2 (High Bias) Cassette Tape Fingerprint (with NR)

4.	Fe (Normal Bias) Cassette Tape Fingerprint (no NR)

5.	FE (Normal Bias) Cassette Tape Fingerprint (with NR)

6.	FeCRO2 (High Bias) Cassette Tape Fingerprint (no NR)

7.	FeCRO2 (High Bias) Cassette Tape Fingerprint (with NR)

8.	Metal (Metal Bias) Cassette Tape Fingerprint (no NR)

9.	Metal (Metal Bias) Cassette Tape Fingerprint (with NR)

10.	Demo Audio Wave file De-Noise

11.	Edison Diamond Disc Fingerprint

12.	Lazy Man's Noise Sample

13.	Pathe´ 80 RPM Acoustic Fingerprint

14.	Typical 1940's Acetate Noise Fingerprint

15.	Typical 1940's Shellac 78 Fingerprint

16.	Typical 33.3 RPM Vinyl Noise Fingerprint

17.	Typical 45 RPM Vinyl Noise Fingerprint

18.	1 7/8 ips reel-to-reel tape fingerprint

19.	3 ¾ ips reel-to-reel tape fingerprint

20.	3 ¾ ips reel-to-reel tape fingerprint (w/ enhancement)

21.	7 ½ ips reel-to-reel tape fingerprint

22.	15 ips reel-to-reel tape fingerprint

23.	78 RPM, 8 inch Audiodisc Acetate Transcription

24.	78 RPM, 10 inch Audiodisc Acetate Transcription

25.	78 RPM, 10 inch Presto Acetate Transcription

26.	1890 Edison 2 Minute White Wax Cylinder

27.	1902 Columbia 78 RPM Fingerprint

28.	1903 Acoustical Lateral Fingerprint

29.	1904 Columbia Phonograph Record Fingerprint

30.	1904 Standard Disc Record 78 RPM Fingerprint

31.	1905 Victor 78 RPM Fingerprint

32.	1906 Edison 2 Minute Wax Cylinder (Standard)

33.	1912 Edison 2 Minute Celluloid Cylinder

34.	1919 Edison Blue Amberol 4 Minute Cylinder

35.	1928 Edison Lateral Cut 78 RPM Fingerprint

36.	1930’s (Early) Victor 33 RPM Program Transcription

37.	Fingerprint

38.	1932 Victor Pre-Grooved Home Recording

39.	1933 HMV (His Masters Voice / English) 78 RPM Fingerprint

40.	1939 English Columbia 78 RPM Fingerprint

41.	1940 Bluebird 78 RPM Fingerprint

42.	1940’s Victor 78 RPM Fingerprint

43.	1948 Capitol 78 RPM Fingerprint

44.	1952 Vinyl 78 RPM Fingerprint

45.	1954 Monophonic 10 inch diameter LP Fingerprint

46.	1952 Mercury 78 RPM Fingerprint

47.	1973 Columbia Vinyl LP Fingerprint (RIAA Equalization)

48.	Micro-cassette 1 7/8 ips Ferris Oxide Fingerprint

49.	Micro-cassette 1 7/8 ips Metal Fingerprint

50.	Micro-cassette 15/16 ips Ferris Oxide Fingerprint

51.	Micro-cassette 15/16 ips Metal Fingerprint

De-Clipper

1.	Analog Tape Recorder De-Clipper

2.	Digital Wave file De-Clipper

3.	Operational Amplifier De-Clipper

4.	Rate Occurrence De-Clipper

5.	Severe Overdrive De-Clipper

6.	Transistor Amplifier De-Clipper

7.	Vacuum Tube Amplifier De-Clipper

8.	Very Strong Interpolation Curvature

Dynamic Noise Filter

1.	Demo Audio Wave file De-Hiss

2.	Cassette Tape De-Hisser

3.	Spectral Exciter/Enhancer 1

4.	Spectral Exciter/Enhancer 2

5.	3 ¾ ips Tape Hiss Attenuator

6.	Brass Instrument Enhancer

Dynamics Processor

1.	1 KHz De-Esser

2.	2 KHz De-Esser

3.	3 KHz De-Esser

4.	4 KHz De-Esser

5.	5 KHz De-Esser

6.	6 KHz De-Esser

7.	7 KHz De-Esser

8.	8 KHz De-Esser

9.	9 KHz De-Esser

10.	10 KHz De-Esser

11.	ALC with Very Long Time Constants

12.	Automatic Level Control

13.	Automatic Level Control with Expander

14.	Basic 2500 Hz De-Esser

15.	ALC with Long Time Constants

16.	ALC with Very Long Time Constants

17.	Automatic Level Control

18.	Automatic Level Control with Expander

19.	Basic 2,500 Hz De-Esser

20.	Elevator Music Compressor

21.	Heavy Compression

22.	Heavy Downward Expansion

23.	Light Compression

24.	Light Noise Reduction

25.	Limiter

26.	Noise Gate – Fast

27.	Noise Gate – Slow

28.	The Politicians Friend

29.	ALC with Short Time Constants

30.	Background Sound Enhancement

31.	Basic 2,500 Hz De-Esser

32.	Basic 2,500 Hz De-Esser

33.	De-Ess #1

34.	De-Ess #2

35.	De-Ess #3

36.	Downward Expander

37.	General Purpose Compressor

38.	Moderate Speed Ballad Compressor

39.	Noise Gate

40.	Noise Gate with 100 mSec Time Constants

41.	Rock Compressor

42.	Slow Ballad Compressor

43.	Telephone "near-far" party compensation

44.	The Sportscaster

45.	The Wedding DJ

Ez-Impulse Noise Filter

1.	Aggressive Scratch and Crackle Remover

2.	Crackle Only Eliminator

3.	Gentle Scratch and Crackle Remover

4.	Scratch and Crackle Eliminator

5.	Scratch Only Eliminator

6.	Very Large 78 RPM Clicks

7.	Very Large LP Clicks

8.	Heavy Scratch, Light Crackle

9.	Aggressive Scratch & Crackle Remover

10.	Light Scratch, Moderate Crackle Remover

File Conversions

1.	Monophonic to Stereo

2.	Monophonic Wave file Clone

3.	Monophonic Wave file Clone w/ 3 dB Attenuation

4.	Monophonic Wave file Clone with 3 dB Gain

5.	Monophonic Wave file Clone with 6 dB Attenuation

6.	Monophonic Wave file Clone with 6 dB Gain

7.	Stereo Lateral Cut to Monophonic

8.	Stereo Solo Vocal Reduction

9.	Stereo to Monophonic

10.	Stereo to Single Track Mono (left)

11.	Stereo to Single Track Mono (right)

12.	Stereo to Stereo Reverse

13.	Stereo Vertical Cut to Monophonic

14.	Stereo Wave file Clone

15.	Stereo Wave file Clone w/ 3 dB Attenuation

16.	Stereo Wave file Clone w/ 3 dB Gain

17.	Stereo Wave file Clone w/ 6 dB Attenuation

18.	Stereo Wave file Clone w/ 3 dB Gain

19.	Strange Brew

20.	Mono to Stereo Simulator

21.	Mono to Stereo Simulator 2

22.	Mono to Stereo Simulator 3

23.	Stereo Phase Inversion

24.	Stereo Wave file Clone with 9.9 dB Gain

25.	Stereo Wave file Clone with 6 dB Gain

Filter Sweeper

1.	33.3 RPM Low Pass Masked Fade-In

2.	33.3 RPM Low Pass Masked Fade-Out

3.	78 RPM Low Pass Masked Fade-In

4.	78 RPM Low Pass Masked Fade-Out

5.	Exponential High Pass Masked Fade-In

6.	Exponential High Pass Masked Fade-Out

7.	Exponential Low Pass Masked Fade-In

8.	Exponential Low Pass Masked Fade-Out

9.	Exponentially Swept Notch Filter

10.	Linear High Pass Masked Fade-In

11.	Linear High Pass Masked Fade-Out

12.	Linear Low Pass Masked Fade-In

13.	Linear Low Pass Masked Fade-Out

14.	Linearly Swept Notch Filter

15.	Log Low Pass Masked Fade-In

16.	Log Low Pass Masked Fade –Out

17.	Logarithmically Swept Notch Filter

Gain Change

1.	1.0 dB Gain Decrease

2.	1.5 dB Gain Decrease

3.	3 dB Gain Decrease

4.	6 dB Gain Decrease

5.	9 dB Gain Decrease

6.	10 dB Gain Decrease

7.	20 dB Gain Decrease

8.	1.0 dB Gain Increase

9.	1.5 dB Gain Increase

10.	3 dB Gain Increase

11.	6 dB Gain Increase

12.	9 dB Gain Increase

13.	10 dB Gain Increase

14.	20 dB Gain Increase

15.	78 RPM Laterals Gain Correction

16.	Curvilinear Fade-In

17.	Curvilinear Fade-Out

Graphic Equalizer

1.	Acoustical 78 EQ

2.	Bass Boost

3.	Bass Boost with Slight High End Cut

4.	Bass Cut

5.	Brilliance

6.	Deep Bass

7.	Flat

8.	Front Row Center

9.	Loudness Contour

10.	Mids Boost

11.	Mids Cut

12.	Noble

13.	Presence

14.	Sad Face

15.	Smiley Face

16.	Treble Boost

17.	Treble Cut

18.	Ultra Bass

19.	Acoustical 78 EQ

20.	High Definition

21.	Sharp Edge Remover

22.	Sub-Woofer Accentuation

23.	Alternating Ripple

24.	Alternating Ripple Reverse

Harmonic Reject Filter

1.	American 120 Hz Buzz Filter

2.	American 120 Hz Buzz Filter - Gentle touch

3.	American 60 Hz Buzz Filter

4.	American 60 Hz Buzz Filter - Gentle touch

5.	European 100 Hz Buzz Filter

6.	European 100 Hz Buzz Filter - Gentle touch

7.	European 50 Hz Buzz Filter

8.	European 50 Hz Buzz Filter - Gentle touch

9.	Flange-1

10.	Flange-2

11.	Flange-3

12.	Flange-4

13.	35 mm Cine Frame Audio Flicker Filter

14.	NTSC TV Vertical Bleed-through Filter

15.	PAL TV Vertical Bleed-through Filter

16.	SECAM TV Vertical Bleed-through Filter

High-pass Filter

1.	Demo Audio Wave file De-Rumble

2.	Rumble Filter

3.	Steep Rumble Filter

4.	Sub-Sonic Filter

5.	De-Thumper (for selective filtering)

6.	De-Thumper 2 (for selective filtering)

Impulse Noise Filter

1.	1940's Acetate Starting Point

2.	1940's Shellac 78 Starting Point

3.	45 RPM Starting Point

4.	Cylinder (2 minute) Record Starting Point

5.	Cylinder (4 minute) Record Starting Point

6.	Demo Audio Wave file De-Click

7.	Diamond Disc 80 RPM Starting Point

8.	Early Shellac 78 RPM Lateral Starting Point

9.	LP De-Click Starting Point

10.	Pathe´ 80 RPM Starting Point

11.	Vertical Cut Starting Point

12.	Vinyl 78 RPM Lateral Starting Point

13.	Vinyl Large Click Only Filter

14.	Vinyl Very Large Click Only filter

15.	Vinyl Extremely Small Click Only filter

16.	Vinyl Small Click Only Filter

17.	Vinyl first pass using HQ mode

18.	Vinyl second pass using HQ mode

19.	Vinyl third pass using HQ mode

20.	LP Static Discharge Noise Suppressor

21.	Sharp Rise-time AM Radio Static Suppressor

22.	AM Radio Static Filter

Low-pass Filter

1.	FM Multiplex Stereo Multi-path Filter

2.	Light LP Surface Noise Filter

3.	Passive Analog Scratch Filter

4.	Steep LP Surface Noise Attenuator

5.	Steep 78 RPM Surface Noise Attenuator

6.	Ultrasonic Filter

Median Filter

1.	78 RPM Record De-Crackler

2.	Extremely Muffled Recording Enhancer

3.	Muffled Communications Enhancer

4.	Timbre

5.	Weighted Median Filter

MultiFilter

1.	Dual Impulse Filter

2.	Forensics Demo Cleanup Filter

3.	Record Clean-up Filter

4.	Triple Impulse Filter

5.	Universal Pop and Click Filter

6.	Live 60 Hz Buzz Filter

7.	Live Channel Blender

8.	Live Dynamic Noise Filter

9.	Live Dynamics Processor

10.	Live File Converter

11.	Live Graphic Equalizer

12.	Live Impulse Noise Filter

13.	Live Paragraphic Equalizer

14.	Live Reverb

15.	Live Tube Amplifier

16.	Live Band-pass Filter

17.	Low Cost Mono Cassette Tape Enhancer

18.	Multiple Notch Filter

19.	Three Stage 50 Hz Buzz Filter

20.	Three Stage 60 Hz Buzz Filter

21.	White (Random) to Seismic Noise Converter

22.	General Purpose Vinyl De-Clicker

23.	A Room within a Room within a Room

24.	Brick Wall Speech Enhancer with Gain Leveler

25.	Brick Wall Speech Filter with Gain Leveler

26.	Bright Tubes and Round Bass

27.	Bright Tubes and Round Bass with Ambience

28.	Butterworth Speech Filter with Gain Leveler

29.	Live Band-pass Filter

30.	MP-3 Enhancer / Full Tilt Boogie

31.	MP-3 Enhancer One

32.	MP-3 Enhancer Two

33.	Poor Reception FM Stereo Enhancer

34.	Speech Enhancer (Basic)

35.	Speech Filter with Noise Gate

36.	Speech Filter with Noise Gate and Enhancer

37.	Speech Filter with Noise Gate and Ultra-Enhancer

38.	Very Bright Tubes

39.	Audio Mixer

40.	Dual Impulse Filter 2

41.	Dual Impulse Filter 3

42.	Live 60 Hz Buzz Filter (Odds & Evens)

43.	Live Adaptive Filter

44.	Live Average Filter

45.	Live Brick Wall Filter

46.	Live Channel Blender

47.	Live Channel Delay Line

48.	Live Continuous Noise Filter

49.	Live De-Clipper

50.	Live Ez-Impulse Noise Filter

51.	Live Hi-Pass Filter

52.	Live Low-Pass Filter

53.	Live Notch Filter

54.	Live Median Filter

55.	Live Polynomial Filter

56.	Live Spectral Filter

57.	Olde Overloaded Radio Playing Far Away

58.	Record Clean-Up Filter

59.	Universal Pop & Click Filter

60.	Very Scratch Vinyl De-Clicker

61.	White (Random) to Brown Noise Converter

62.	MP3 Enhancer Three

63.	MP3 Enhancer Four

64.	Butterworth Speech Filter w/Noise Gate

65.	Forensics Filter Lineup-Nominal

Notch Filter

1.	American 120 Hertz Hum Filter

2.	American 120 Hertz Hum Filter (sharp)

3.	American 60 Hertz Hum Filter

4.	American 60 Hertz Hum Filter (sharp)

5.	American AM Heterodyne Filter

6.	American AM Heterodyne Filter (sharp)

7.	European 100 Hertz Hum Filter

8.	European 100 Hertz Hum Filter (sharp)

9.	European 50 Hertz Hum Filter

10.	European 50 Hertz Hum Filter (sharp)

11.	European AM Heterodyne Filter

12.	European AM Heterodyne Filter (sharp)

13.	Simple 78 RPM Record Hiss Filter

14.	Simple 78 RPM de-crackle filter

15.	50 Hz Slot Filter

16.	60 Hertz Slot Filter

17.	100 Hertz Slot Filter

18.	120 Hertz Slot Filter

19.	400 Hertz Notch Filter (Aircraft Electrical Noise Filter)

20.	400 Hertz Slot Filter

21.	FM Stereo Pilot Frequency Attenuator

22.	NTSC Horizontal Scan Signal Attenuator (15,750 Hz)

23.	Middle C Finder (262 Hertz on the Musical Scale)

24.	PAL Horizontal Scan Signal Attenuator

25.	SECAM Horizontal Scan Signal Attenuator

26.	1000 Hz Slot Filter - One Tenth Octave

27.	1000 Hz Slot Filter- Quarter Octave

28.	1000 Hz Slot Filter- Half Octave

29.	1000 Hz Slot Filter- One Octave

Paragraphic Equalizer

1.	6 KHz Brick Wall Low-pass Filter

2.	25 Hz Brick Wall High-pass Filter

3.	25 mSec De-Emphasis

4.	25 mSec Pre-Emphasis

5.	75 mSec De-Emphasis

6.	100 Hz Boost with 12 dB Shelf

7.	100 Hz Boost with 16 dB Shelf

8.	100 Hz Cut with -12 dB Shelf

9.	100 Hz Cut with -16 dB Shelf

10.	2 KHz Boost with 12 dB Shelf

11.	2 KHz Boost with 16 dB Shelf

12.	2 KHz Cut with -12 dB Shelf

13.	2 KHz Cut with -16 dB Shelf

14.	3 KHz Boost with 12 dB Shelf

15.	3 KHz Cut with -12 dB Shelf

16.	78 RPM, 125 Hz Turnover Curve

17.	78 RPM, 200 Hz Turnover Curve

18.	78 RPM, 250 Hz Turnover Curve

19.	78 RPM, 500 Hz Turnover Curve

20.	78 RPM, 800 Hz Turnover Curve

21.	American 120 Hz Hum Filter

22.	American 120 Hz Hum Filter (sharp)

23.	American 60 Hz Hum Filter

24.	American 60 Hz Hum Filter (sharp)

25.	European 100 Hz Hum Filter

26.	European 100 Hz Hum Filter (sharp)

27.	European 50 Hz Hum Filter

28.	European 50 Hz Hum Filter (sharp)

29.	Flat Response

30.	NAB Tape Playback Curve

31.	Pink to White Noise Converter

32.	Reverse NAB Tape Playback Curve

33.	Reverse RIAA Phono Equalization Curve

34.	Reverse RIAA w/ 125 Hz 78 Turnover

35.	Reverse RIAA w/ 200 Hz 78 Turnover

36.	Reverse RIAA w/ 250 Hz 78 Turnover

37.	Reverse RIAA w/ 500 Hz 78 Turnover

38.	Reverse RIAA w/ 800 Hz 78 Turnover

39.	RIAA Phone Equalization Curve

40.	Rumble Filter

41.	White to Pink Noise Converter

42.	75 mSec Pre-Emphasis

43.	Soft Slope (negative 1 dB / Octave)

44.	Soft Slope (positive 1 dB / Octave)

45.	Sound Level A-Weighting Curve

46.	Sound Level C Weighting Curve

47.	DTMF Comb Filter (Narrow)

48.	DTMF Comb Filter (Normal)

49.	DTMF Comb Filter (Ultra-Wide)

50.	Fletcher-Munson Aural Hot Spot Accentuator

51.	Fletcher-Munson Aural Hot Spot Attenuator

52.	Fletcher-Munson Contour @ 20 dB SPL

53.	Fletcher-Munson Contour @ 40 dB SPL

54.	Fletcher-Munson Contour @ 60 dB SPL

55.	Fletcher-Munson Contour @ 80 dB SPL

56.	Fletcher-Munson Contour @ 100 dB SPL

57.	Fletcher-Munson Contour @ 120 dB SPL

58.	Stereo Simulator Left Channel Comb Filter

59.	Stereo Simulator Left Channel Comb Filter (Wide)

60.	Stereo Simulator Right Channel Comb Filter

61.	Stereo Simulator Right Channel Comb Filter (Wide)

62.	Random IIR-Based Phase Shifter 1

63.	Random IIR-Based Phase Shifter 2

Reverb

1.	Large Auditorium

2.	Large Cathedral

3.	Large Cavern

4.	Large Submarine

5.	Large Train Station

6.	Light Reverb

7.	Lighthouse

8.	Lots of Reverb

9.	Medium Size Auditorium

10.	Medium Size Wood Room 1

11.	Medium Size Wood Room 2

12.	Phasing Effect 1

13.	Phasing Effect 2

14.	Slap-back 1

15.	Slap-back 2

16.	Slap-back 3

17.	Slap-back 4

18.	Small Intimate Night Club 1

19.	Small Intimate Night Club 2

20.	Small Intimate Night Club 3

21.	Small Masonry Room 1

22.	Small Masonry Room 2

23.	Small Stone Church

24.	Small Theater 1

25.	Small Theater 2

26.	Small Wood Church

27.	Small Wood Room

28.	Ultimate Reverb

29.	Big Brass Boomy Bass

30.	Bright Boomy Bass

31.	Grande Canyon

32.	Saint Peters Basilica

33.	Boomy Olde Wood Room

34.	Delay Line, Very Short

35.	Delay Line, Short

36.	Delay Line, Medium

37.	Delay Line, Long

38.	Echo 1

39.	Echo 2

40.	Echo 3

41.	Echo 4

42.	Ambience 1

43.	Ambience 2

44.	Ambience 3

45.	Ambience 4

46.	Gentle Touch 1

47.	Gentle Touch 2

48.	Metal Plate Reverb

49.	Classical Concert Hall

50.	Smokey Club

Polynomial Filter

1.	5th Order Asymmetrical Expander Non-Linearity

2.	5th Order Asymmetrical Compressing Non-Linearity

3.	Asymmetrical Expanding Non-Linearity with Phase Inversion

4.	Basic Asymmetrical Non-Linearity

5.	Class A Amplifier

6.	Comb Generator

7.	Double Inflection Point Non-Linearity

8.	Double Inflection Point Non-Linearity #2

9.	Even + Odd Order Harmonic Generator

10.	Even Order Frequency Multiplier

11.	Four Inflection Points #1

12.	Four Inflection Points #2

13.	Full Wave Rectifier

14.	Gross Non-Linearity

15.	Inverse Class A Amplifier

16.	Instantaneous Compressor

17.	Instantaneous Expander

18.	Inverse Push-Pull Power Amplifier

19.	Inverse Push-Pull Power Amplifier with Phase Inversion

20.	Inverting Real Time Expander

21.	Negative DC Offset

22.	Phase Inverter

23.	Positive DC Offset

24.	Saturated Amplifier with Local Feedback

25.	Sensitive Full Wave Rectifier

26.	Slightly Expansive Non-Linearity

27.	Sensitive Half Wave Rectifier

28.	Very Expansive Non-Linearity

29.	Very Kinky

30.	Weak Bulbs in the Olde Radio Set

31.	Even Order Frequency Multiplier 2

32.	Four Inflection Points #2

33.	Compress +, Expand –

34.	X 0.25 Gain Increase

35.	X 4 Gain Increase

36.	X 0.25 Gain Increase With Expansion

37.	Level Sensitive Fuzz Box

Punch and Crunch

1.	1950's Jukebox

2.	AM Radio De-Compressor

3.	Background Music Compressor

4.	Big Bad Bass

5.	Big Brash Bass

6.	Big Brass Bass

7.	Classical Music Compressor

8.	Exciter

9.	Fat Round Bottom

10.	Female Lead Vocal Enhancer

11.	Male Lead Vocal Enhancer

12.	Fission Reaction

13.	FM Radio Dial Presence

14.	Front Row, Center

15.	Fusion Reaction

16.	Heavy Compression / Balanced Spectrum

17.	Heavy Expansion / Balanced Spectrum

18.	Heavy Expansion Big Bass

19.	Heavy Expansion / Bright

20.	Heavy Expansion / Brilliant

21.	Heavy Expansion / Round Bottom

22.	Hip Hop

23.	Light Compression / Balanced Spectrum

24.	Light Expansion / Balanced Spectrum

25.	Light Expansion / Bright

26.	Light Expansion / Brilliant

27.	Light Expansion / Round Bottom

28.	Moderate Compression / Balanced Spectrum

29.	Moderate Compression / Muted

30.	Moderate Expansion / Balanced Spectrum

31.	Moderate Expansion / Bright

32.	Moderate Expansion / Brilliant

33.	Moderate Expansion / Round Bottom

34.	Very Heavy Compression / Balanced Spectrum

35.	Very Heavy Compression / Muted

36.	Very Heavy Expansion / Bright

37.	Very Heavy Expansion / Brilliant

38.	Very Light Compression / Balanced Spectrum

39.	Very Light Compression / Muted

40.	Very Light Expansion / Balanced Spectrum

41.	Very Light Expansion / Brilliant

42.	Forensics (Noise Reduction)

Spectral Filter

1.	100 Hz Hum Filter (for Defective European Filter Capacitors)

2.	120 Hz Hum Filter (for Defective US Filter Capacitors)

3.	400 Hz Aircraft Electrical Noise Filter

4.	50 Hz Hum Filter (European)

5.	60 Hz Hum Filter (US)

6.	60 Hz Buzz Filter

7.	5 KHz Brick Wall Low Pass Filter

8.	Speech Filter

9.	Speech and 400 Hz Aircraft Electrical Buzz Filter

10.	Speech and 50 Hz Buzz Filter

11.	Speech and 60 Hz Buzz Filter

12.	Phase Shifter- Randomized

13.	Phase Shifter

14.	400 Hz Aircraft Electrical Buzz Filter

Stretch & Squish Filter

1.	5 % Faster

2.	5 % Slower

3.	10 % Faster

4.	10 % Slower

5.	20 % Faster

6.	20 % Slower

7.	30 % Faster

8.	30 % Slower

9.	Spoken Word Transcription Rate

10.	Spoken Word Transcription Rate 1

11.	Spoken Word Transcription Rate 2

12.	Higher Pitch and Speed

13.	Much Higher Pitch and Speed

14.	Disguised Voice Effect

15.	Lower Pitch and Speed

16.	Disguised Voice Effect #2

17.	Constant Pitch

18.	Spoken Word Transcription Rate #3

19.	2% Slower

20.	2% Faster

21.	15% Slower

22.	15% Faster

Virtual Valve Amplifier

1.	25 Watt / Channel Power-Amp 1 (6L6-GC)

2.	25 Watt / Channel Power-Amp 2 (6L6-GC)

3.	25 Watt / Channel Power-Amp 3 (6L6-GC)

4.	25 Watt / Channel Power-Amp 4 (6L6-GC)

5.	8 Watt, 6L6 Single-Ended Amplifier 1

6.	8 Watt, 6L6 Single-Ended Amplifier 2

7.	8 Watt, 6L6 Single-Ended Amplifier 3

8.	Grunge (Odds Only)

9.	Harmonic Enhancer 1 (Evens + Odds)

10.	Harmonic Enhancer 2 (Evens + Odds)

11.	Harmonic Enhancer 3 (Evens Only)

12.	Harmonic Enhancer 4 (Evens Only)

13.	Harmonic Enhancer 5 (Evens Only)

14.	Harmonic Enhancer 6 (Odds Only)

15.	Harmonic Enhancer Sweet Spot

16.	High-End (Triode) Audio Pre-Amplifier

17.	Overloaded Guitar Power Amplifier

18.	Overloaded Guitar Pre-Amplifier

19.	Pentode Microphone Pre-Amplifier 1

20.	Pentode Microphone Pre-Amplifier 2

21.	Purist 1

22.	Purist 2

23.	Purist 3

24.	Triode Tube Pre-Amplifier 1

25.	Triode Tube Pre-Amplifier 2

26.	Triode Tube Pre-Amplifier 3

27.	Triode Tube Pre-Amplifier 4

28.	Triode Tube Pre-Amplifier 5

29.	Triode Tube Pre-Amplifier 6

30.	Vacuum Tube Fuzz Box 1

31.	Vacuum Tube Fuzz Box 2

32.	Vacuum Tube Fuzz Box 3

33.	Vacuum Tube Fuzz Box 4

34.	1935 Retro 15 Watt Push-Pull Amp

35.	1938 Retro 4 Watt Class A Power Amp

36.	Bright & Brashy Brass

37.	Red Hot Jazz

38.	12AX7 Based Class A Amplifier (Sweet)

39.	12AX7 Based Class A Amplifier (Warm)

40.	12AX7 Based Class A Amplifier (Full Range)

41.	2A3 Based Class A Amplifier (Sweet)

42.	2A3 Based Class A Amplifier (Warm)

43.	2A3 Based Class A Amplifier (Full Range)

44.	12AU7 Based Class A Amplifier (Sweet)

45.	12AU7 Based Class A Amplifier (Warm)

46.	12AU7 Based Class A Amplifier (Full Range)

47.	6EJ7 Based Class A Amplifier (Sweet)

48.	6EJ7 Based Class A Amplifier (Warm)
6EJ7 Based Class A Amplifier (Full Range)


A Brief History of Diamond Cut Productions

In the spring of 1986, an R&D engineer/scientist by the name of Craig Maier read an article in The Star Ledger, a local newspaper, entitled "Budget Cuts Cast Shadow on Edison National Historic Site." The article, written by science editor Kitta McPherson, described the deteriorating condition of the Edison National Historic Site and its archives located in West Orange, New Jersey. Among the many artifacts which were not receiving the proper curatorial attention due to poor funding was a collection of test-press recordings which were made by the Edison company between the years of 1927 through 1929, which was their last few years in the record business. Craig told a friend and fellow engineer named Rick Carlson about the article in hopes that it might stir up in him some interest in the Edison site as well. Craig and Rick, after some considerable discussion, decided to offer to volunteer some of their spare time and technical expertise in the area of audio hardware and software engineering in order that the Edison Lateral collection of test pressing recordings could be transferred to digital tape so that the "sound artifacts" would be eternally preserved and archived in the digital domain at the site.

Contact was made with then Supervisor Museum Curator, Dr. Edward Pershey, Ph.D. During their first meeting at the site, Dr. Pershey showed the two engineers thousands of one-of-a-kind test pressing recordings that were piled in stacks on a long row of tables on the second floor of the Edison main laboratory building. This initial introduction to the collection was an earnest attempt to sober up these two individuals as to the magnitude of the undertaking for which they were volunteering. The total number of songs which were recorded numbered over 1200 in anywhere from two to five takes each. This only further increased their interest in the project since the possibility of finding some truly important music that had previously been unheard since the late 1920's would be quite high in such a large collection of test pressings. After several additional meetings with Dr. Pershey, an informal agreement was made such that the two engineers could proceed to seek out funding from private sources to set up an audio restoration laboratory in one of their own homes for the project. They contacted around 30 companies in the New Jersey area seeking funds to help build their laboratories. After about seven months of effort, they succeeded in raising enough money to fund their project. In addition to fund raising, they also designed and constructed several pieces of custom equipment that was needed for the project (equipment which was not readily available on the market at the time).

The next step was to become educated in the proper technique of archival audio transferring. To that end, they hired Mr. Tom Owens of the Rogers and Hammerstein musical library in New York City as an engineering consultant. Tom spent time with the two engineers at the New York City Public Library sound lab (Rogers and Hammerstein) teaching them some of the "tricks of the trade." Tom also visited the first sound lab which the two engineers set up for the restoration project located at Craig's home in Verona, NJ. He provided constructive criticism regarding the sound lab which the two engineers had set up, allowing them to improve upon their initial system. One significant problem which Tom highlighted for the two engineers was that of establishing the correct turnover frequency for the transfer of these lateral test pressings. Documentation could not be found at the Edison site regarding the specifics of this important parameter. So Rick and Craig devised some experiments which were conducted on a "high-end" vacuum tube based Edison phonograph designed around the same time period as the test pressings in order to deduce the correct turnover frequency. After their experiments, modifications were made to their magnetic phonograph pre-amplifier to provide the most likely proper turnover frequency for the transfers.

A seven year pro-bono contract was drawn up between the Edison National Historic Site / U.S. Department of the Interior, and Rick Carlson and Craig Maier for the purposes of executing the project outlined above.
Finally, the two engineers were ready to begin the project. Nearly one full year had lapsed before the first record was transferred to digital tape at Craig's home in Verona, N.J. Shortly thereafter, the sound lab was rebuilt in the Maier's new home in Rockaway Township, NJ. That is the location in which the lion’s share of the transfer project took place over the next seven years.

After transferring around 900 of the songs (times 2 - 5 takes per song, about 2,200 transfers in total) Craig and Rick decided that the music was not doing much good sitting in the underground vault of a museum. Since they were the only two people alive who had heard almost the entire collection, they decided that it would be a good idea to try to release some of this previously unreleased material (only around 200 of the songs had ever been released in the Edison lateral format). So they approached the Edison site in order to try to accomplish this. After about one year of frustration in dealing with the bureaucracy, they decided it would be a lot easier to form their own company and release these songs under their own record label. Thus was formed Diamond Cut Productions in 1992 with Craig and Rick providing their own seed capital for the venture. Their first release entitled "Unreleased Edison Laterals 1 - - - an anthology of Edison Needle type records" was such a success in the market that they were able to start another project in 1994 entitled "The California Ramblers, Edison Laterals 2." For this project, they decided to improve on the audio restoration process, which they had used on their previous release. Instead of analog signal processing, they migrated to digital signal processing utilizing their own algorithms to remove crackle, ticks, pops and hiss from the original material. They named their process (which ran on an inexpensive pc) "Diamond Cut Audio restoration tools" or DC-Art for short. Their technique proved successful to the extent that the Smithsonian Institution Press employed Diamond Cut Productions to perform audio restoration for some of their American Songwriter Series of CD releases using this process. Diamond Cut's third CD release entitled "Hot Dance of the Roaring 20's, Edison Laterals 3" was processed utilizing exclusively their own audio restoration program; all analog processing equipment had been abandoned by this point in time. In the meantime and in parallel with the efforts to bring "Hot Dance . . . " to the market, Craig worked with County records to produce and release an Edison olde tyme group on CD called "Ernest Stoneman and his Dixie Mountaineers" using their audio restoration process. In the spring of 1996, their program was first formally introduced into the commercial marketplace at a meeting of "Record Research" which was held at the Maier residence in Rockaway Township, NJ. Since then it has been sold throughout the world for not only musical audio restoration applications, but for others such as 911 call restoration, clarification of police surveillance recordings, cleanup of radio broadcasts for release on CD, restoration of historic spoken word recordings, cockpit voice recording restoration, plus many others.

Enhanced Audio has now become one of the predominant players in the international audio restoration software market. In the software domain, they have also introduced a product specifically designed for the enhancement of compressed MP3 files called Enhance/MP3.   Also, new features and improved performance will be added into their legacy audio restoration software products on an on-going basis.

In the future, Diamond Cut Productions expects to continue releasing more CDs in their Edison Lateral Cut series. However, they have also branched out into other musical venues from the 1920's and 1930's.


Diamond Cut Audio Restoration Tools Development Timeline

Mid 1993 -

While frustratingly removing clicks from an Edison record using just a computer and a mouse whilst drinking too much wine and beer, Rick Carlson and Craig Maier wrote the first few lines of code in an attempt to perform this process automatically.  Although very crude at the beginning, this particular algorithm (the Impulse Filter) eventually developed into its present level of sophistication occupying about 500 K of compiled code.  Several of those original lines and thinking are still in the present day code.  Rick had stored those original lines of code under the filename of dcart.exe ultimately giving rise to the name "Diamond Cut audio restoration tools."

Late 1993 -

Having heard about Diamond Cut through the grapevine, Bruce Talbot, executive producer for the Smithsonian Collection of Recordings contacted us and inquired if they would be willing to apply their "de-clicking" process to a particularly bad recording of "Darn That Dream." This restoration was released on the American Songbook Series on a CD entitled  "James Van Heusen." which was copyrighted in 1994 as release number RD 048-18 / A 23955.

Early 1994 -

Having been satisfied with their work on the "James Van Heusen" CD, Bruce Talbot Again contacted Diamond Cut to do some work on a release in the American Songbook Series, which was to be called "Richard Whiting."  On it, they restored a "basket case" copy of a Rudy Vallee rendition of "Honey."  In the liner notes for the CD it says "The only source available for "Honey" was a 78-rpm disc in very poor condition.  The sound quality has been greatly improved by Craig Maier and Rick Carlson of Diamond Cut Productions, using their DC-Art system of sound restoration."  This CD is also copyrighted in 1994 but under the release number of RD048-22 / A 24571.

Mid 1994 -

While recuperating from surgery and being extremely bored, Craig decided to restore an entire artist's output of Edison Lateral Cut records which Rick and Craig had earlier transferred to digital audiotape as part of the Edison Lateral Cut Disc archival project.  The artist was Ernest V. Stoneman (and his Dixie Mountaineers).  He used their DC-Art program to remove all extraneous clicks, pops, and hiss from the originals.  Ultimately, Diamond Cut Productions sold the Master digital tape to County Records which released and copyrighted the CD in 1996 on County release number CD-3510.

Late 1994 -

Diamond Cut Productions decided that it was time to release another in its series of mostly Unreleased Edison Lateral Cut recordings.  They choose "The California Ramblers" as the subject of the release.  They also decided to use no additional analog processing equipment other than their DC-Art program to restore the recording.  De-clicking, de-popping, de-hissing, de-rumbling and minor equalization were all performed via their computer algorithms.  Thus was their first release which included only two analog steps, that of the mastering back in the late 1920's and that of the transfer to digital tape in the late 1980's and early 1990's.  This CD is entitled "The California Ramblers - - - Edison Laterals 2" and was released under number DCP-301D and copyrighted in late 1994.  This CD is still available from Diamond Cut Productions, Inc. and other venues throughout the world.

Early 1995 -

Diamond Cut Productions decided to expand their business from the production of CD's to the manufacture of audio restoration software products.  They decided that since the program had been quite useful to in their CD business, it might also be useful to many other people with audio collections in need of restoration.  The first public release took about 7 months to smooth out the bumps in the program that we had learned to work around. Making a commercial software product is much different than writing one for yourself as they were soon to find out!

April 21st 1995-

Diamond Cut Productions sent out the initial Beta version of DC-Art to some potential customers identified as QA 1.1.  They worked on de-bugging this software for the next few months.  If you have an original copy of QA 1.1, it may be an antique!  (Antique software - - - what's this world coming to??)

July 1996

Version 1.0 was officially released at a meeting of "Record Research" held at the Maier Rockaway Township residence.  The first copies were sold at the end of the meeting during which the program had been demonstrated to a small group of about 15 people.

September 4th 1997

A distributorship agreement is signed between Diamond Cut Productions, Inc. and Tracer Technologies, Inc. in order to facilitate the worldwide marketing and distribution of the DC-Art product line.

December 1997

Version 2.0 of DC-Art featuring, among other things, real time Preview was introduced into the market.

September 1998

Version 3.0 of DC-Art, otherwise known as Diamond Cut 32 was introduced into the market with novel features such as the Virtual Valve Amplifier.

August 1999

Version 4.0 otherwise known as Live and Millennium was introduced.  This brought with it a new level of performance and features in the audio restoration and enhancement software market. Feature like Live feed through mode, and 24-bit/96KHz support have changed the landscape in this area of endeavor.

December 1999

Established German distributorship of the Live and Millennium program through Digital Broadcast Systems GmbH (dBS).  The products can be seen in the German language at www.diamondcut.de

August 2001

Updated Live and Millennium to version 4.8 with bug fixes and the addition of a high resolution VU Meter.  Also, the frequency resolution of the Spectrum Analyzer was greatly enhanced.

February 2001

Released the code for Enhance/MP3, a low cost product used for improving the sound quality of MP3 audio files.  Diamond Cut Productions Inc. and Tracer Technologies form a new Partnership called Enhanced Audio Inc. (www.enhancedaudio.com), whose sole purpose is to market, distribute and create the world’s best audio enhancement, restoration, and analysis tools.

June 2002

Released Live Five to Beta


Diamond Cut Productions Edison Lateral Series CD and Cassette Releases

Diamond Cut Productions is proud to offer releases of Edison Lateral Cut Test Pressing recordings available in the CD format, with one available in both CD and Cassette.  Also, we have an assortment of non-Edison recordings available.  The following is our current product offering:

1.	Unreleased Edison Laterals 1 CD Version (DCP-201D) - - - $17.95 each

2.	Unreleased Edison Laterals 1 Cassette Version (DCP-201S)  - - - $9.95 each

3.	The California Ramblers, Edison Laterals 2  (DCP-301D)  - - -  $17.95 each

4.	Hot Dance of the Roaring 20's, Edison Laterals 3 (DCP-202D)  - - - $17.95 each

5.	Ernest V. Stoneman and his Dixie Mountaineers (Edison) - - -  $17.95 each

6.	Eva Taylor with Clarence Williams, Edison Laterals 4  (DCP-303D) - - - $17.95 each

7.	Vaughn De Leath - The Original Radio Girl, Edison Laterals 5 (DCP-304D) - - - $17.95 each

8.	Hot and Rare - Hot Dance tunes from Rare Jazz Recordings (DCP-203D) - - - $17.95 each

9.	B.A. Rolfe and his Lucky Strike Orchestra  (DCP-305D) - - - $17.95 each

10.	The Marvelous Melodies of Peter Mendoza  (DCP-306D) - - - $17.95 each

11.	Edison Diamond Disc Fox Trots: 1920 - 1923 (DCP-307D) - - - $17.95 each

12.	Rudy Vallee’ and His Connecticut Yankees: 1928 - 1930 (DCP-308D) - - - $17.95 each

13.	Eddie Duchin and his Central Park Casino Orchestra 1932 - 1937 (DCP-309D) - - - $17.95 each

14.	Jazz: It’s a Wonderful Sound! (DCP-500D) - - - $17.95 each

For all other items, please include $2.00 per item to cover Shipping and Handling in the U.S.  Europe, please include $4.00 per item.

N.J residents add 6% Sales Tax.

To order, please visit our on-line store located at www.diamondcut.com, or

please send your order including your check or money order payable in U.S funds to:

Diamond Cut Productions, Inc.

P.O. Box 305

Hibernia, NJ  07842-0305

Fax your order to 973-316-5098; we take Visa or MasterCard

Call in your order at 973-316-9111

Visit our website located at www.diamondcut.com

Note: Dealers, Retailers, and Record Distributors are invited to request a copy of our Discount Schedule.


Tracer Technologies. . . For the Supplies You Need

Tracer Technologies has been in the business of supplying audio products for the last 8 years.  If you need a few little tweaks to finish your system, pay us a visit at www.tracertek.com

•	Sound Cards:  The most important part of this equation.  You now own the world’s best audio restoration software…don’t funnel it through anything that has the word "Blaster" in it…or any bundled sound card.

•	Stereo Preamp: You cannot plug a turntable directly into your sound card.  If you don’t want to drag your whole stereo to your computer, Tracer offers a full assortment of preamps that will make this job cake.

•	Speakers:  Make your ears happy.  Stop depending on those cheesy little bundled speakers that came with your computer.

•	Headphones: All part of the ear happiness theory.  Work on your audio files while the rest of the house sleeps.  Also hear greater detail while you work.

•	Mixers: Any easy way to plug anything into your sound card.  Mics, instruments, etc. quickly and easily plug in.

•	Microphones: If you’ve got to record your voice…stop looking, Tracer has a line of mics that are inexpensive, but rivals to the world’s best studio mics.

•	Multi-track Software:  The one thing that FIVE/LIVE doesn’t do.  Now you can quickly and easily record multiple tracks just like a professional recording studio.

•	Record Cleaning Kits: Before you restore; clean.  These kits can help save needle and record wear by giving you a clean surface to start.

•	Turntables: If you’re in the market for a good turntable, we have a line that is very high quality, but won’t cost an arm and a leg.

•	MP3 Players:  Now that you can make your own MP3s, why not try our cool little portable MP3 players.

•	Home MP3 Players:  MP3 comes home.  We have nifty little units that look great in your stereo rack, but control your computer’s hard drive and provide access to your entire library of MP3s within your home stereo.

Tracer Technologies

3600 Board Rd.

York, PA  17402

717 764 9240

fax 764 9254

www.tracertek.com

sales@tracertek.com


License Agreement

This is a legal agreement between you, the end user, and Diamond Cut Productions, Inc.  (Diamond Cut Productions).  The enclosed Diamond Cut Productions software program (the SOFTWARE) is licensed by Diamond Cut Productions for use only on the terms set forth herein.  Please read this license agreement.  Opening the diskette or CD ROM package or the use of this software indicates that you accept these terms.  If you do not agree to these terms, return the full product with media unopened and proof of purchase to Diamond Cut Productions or its representative within 30 days for a full refund, or destroy the software and its component parts.

GRANT OF LICENSE.  Diamond Cut Productions grants to you the right to use one copy of the enclosed  FIVE/LIVE SOFTWARE on a single terminal connected to a single computer (i.e. single CPU) or to a network server.  If you install the SOFTWARE on a network server, you must purchase a separate copy of the SOFTWARE for each computer terminal that will be used to operate the SOFTWARE.

GRANT OF NETWORK LICENSE. If you are acquiring a version of the SOFTWARE specifically intended for network use, Diamond Cut Productions grants you the right to use the SOFTWARE on a LICENSED COMPUTER NETWORK as provided below.  A computer network is any combination of two or more terminals that are electronically linked and capable of sharing the use of a single software program.  A LICENSED COMPUTER NETWORK is a computer network for which you have acquired and dedicated at least one (1) Diamond Cut Productions standard version of the SOFTWARE (which can run stand-alone or as a network server).  For additional users to use the SOFTWARE on the network you must acquire a WRITTEN LICENSING AGREEMENT from Diamond Cut Productions indicating the number of users expected to be able to use the SOFTWARE.  You may have as many copies of the SOFTWARE in simultaneous use on the network as is specifically authorized in the WRITTEN LICENSING AGREEMENT.

COPYRIGHT. The SOFTWARE is owned by Diamond Cut Productions and is protected by United States copyright laws and international treaty provisions. You may either (a) make one copy of the SOFTWARE solely for backup or archival purposes provided that you reproduce all copyright and other proprietary notices that are on the original copy of the SOFTWARE provided to you, or (b) transfer the SOFTWARE to a single hard disk provided you keep the original solely for backup or archival purposes.

OTHER RESTRICTIONS.  You may not rent or lease the SOFTWARE.  You may not reverse engineer, decompile, disassemble, or create derivative works from the SOFTWARE.

GOVERNMENT LICENSEE.  If you are acquiring the SOFTWARE on behalf of any unit or agency of the United States Government, the following provisions apply:

The Government acknowledges Diamond Cut Productions representation that the SOFTWARE and its documentation were developed at private expense and no part of them is in the public domain.

The Government acknowledges Diamond Cut Production's representation that the SOFTWARE is "Restricted Computer Software" as that term is defined in Clause 52.227-19 of the Federal Acquisition Regulations (FAR) and is "Commercial Computer Software" as that term is defined in Subpart 227.471 of the Department of Defense Federal Acquisition Regulation Supplement (DFARS).  The Government agrees that:

(i) If the SOFTWARE is supplied to the Department of Defense (DoD), the SOFTWARE is classified as "Commercial Computer Software" and the Government is acquiring only "restricted rights" in the SOFTWARE and its documentation as that term is defined in Clause 252.227-7013 (c) (1) of the DFARS, and

             (ii) If the SOFTWARE is supplied to any unit or agency of the United States Government other than DoD, the Government's rights in the SOFTWARE and its documentation will be as defined in Clause 52.227-19 (c) (2) of the FAR.

RESTRICTED RIGHTS LEGEND.  Use, duplication or disclosure by the Government is subject to restrictions as set forth in subparagraph (c) (1) (ii) of the Rights in Technical Data and Computer Software clause at DFARS 252.227-7013.  Diamond Cut Productions, P.O. Box 305, Hibernia, NJ 07842-0305.

EXPORT LAW ASSURANCES.  You acknowledge and agree that the SOFTWARE is subject to restrictions and controls imposed by the United States Export Administration (the "Act") and the regulations thereunder.  You agree and certify that neither the SOFTWARE nor any direct product thereof is being or will be acquired, shipped, transferred or re-exported, directly or indirectly, into any country prohibited by the Act and the regulations thereunder or will be used for any purposes prohibited by the same.

GENERAL. This Agreement will be governed by the laws of the State of New Jersey, except for that body of law dealing with conflicts of law.  Should you have any questions concerning this Agreement, or if you desire to contact Diamond Cut Productions for any reason, please write: Diamond Cut Productions Customer Sales and Service, P.O. Box 305, Hibernia, NJ 07842-0305.

NUMBER OF COPIES LICENSED. If you have not purchased a license that authorizes use of the Software on multiple computers or by multiple individuals, then you are authorized to use ONLY a single copy of the Software on a single computer.  Only ONE copy of the Software may be created for archival or backup purposes.  All copies of the Software MUST include the Diamond Cut copyright notice and other legal notices.

TERM. This license is effective from your date of purchase and shall remain in force until terminated.  You may terminate the license and this License Agreement at any time by destroying the Software and the accompanying documentation, together with all copies in any form.

LIMITED WARRANTY.  Diamond Cut Productions, Inc. warrants that the SOFTWARE will perform substantially in accordance with the accompanying written materials for a period of ninety (90) days from the date of receipt.  Any implied warranties on the SOFTWARE are limited to ninety (90) days.  Some states do not allow limitations on duration of an implied warranty, so the above limitation may not apply to you.

CUSTOMER REMEDIES.  Diamond Cut Productions, Inc. entire liability and your exclusive remedy shall be, at Diamond Cut Productions option, either (a) return of the price paid or (b) repair or replacement of the SOFTWARE that does not meet Diamond Cut Productions Limited Warranty and that is returned to Diamond Cut Productions with a copy of your receipt.  This Limited Warranty is void if failure of the SOFTWARE has resulted from accident, abuse, or misapplication.  Any replacement SOFTWARE will be warranted for the remainder of the original warranty period or thirty (30) days, whichever is longer.  These remedies are not available outside of the United States of America.

NO OTHER WARRANTIES.  Diamond Cut Productions, Inc. disclaims all other warranties, either express or implied, including but not limited to implied warranties of merchantability and fitness for a particular purpose, with respect to the SOFTWARE and any accompanying written materials.  This limited warranty gives you specific legal rights.  You may have others, which vary from sate to state.

NO LIABILITY FOR CONSEQUENTIAL DAMAGES.  In no event shall Diamond Cut Productions, Inc. or its suppliers be liable for any damages whatsoever (including, without limitation, damages for loss of business profits, business interruption, loss of business information, or other pecuniary loss) arising out of the use of or inability to use this Diamond Cut Productions, Inc. product, even if Diamond Cut Productions, Inc. has been advised of the possibility of such damages.  Because some states do not allow the exclusion or limitation of liability for consequential or incidental damages the above limitation may not apply to you.


The 50 Hz Printable Stroboscope


The 60 Hz Printable Stroboscope

THIS SECTION IS NOT INCLUDED IN THE DEMO HELP