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Copyright © 2007 GoldWave Inc.
Do not upload or include this document on a website.
October 2007
GoldWave is a professional digital audio editor that plays, records, edits, processes, and converts audio on your computer. This section lists some of the features of GoldWave and outlines the notation and organization of the manual.
GoldWave includes a complete set of audio processing features.
Familiarity with the Windows interface, such as property windows, tool bars, scroll bars, etc., is recommended before reading this manual.
For those who are unfamiliar with digital audio, Appendix A briefly introduces some of the fundamentals of computer audio. Appendix D gives a tutorial for recording audio from a turntable, removing noise, and splitting the file into tracks for CD-R burning. Appendix E contains troubleshooting information and answers to common questions.
Section II: Getting Started, covers system requirements and installation. Section III: Using GoldWave explains the interface and menu structure in detail. Topics are covered in the order that they appear in GoldWave's menu. Section IV: General Information, provides support, copyright, and warranty information.
Bold or link coloured text and a vertical bar are used to denote menu commands. File | New, for example, means to select the New command from the File menu. This notation is used to refer to other sections within this manual as well. In the above example, you can find information by looking for New under the File Menu Commands section. If the first word is Start, then select the command from the main Windows task bar menu structure instead.
Options and settings are given in a fixed width font or in quotes.
A information icon emphasizes helpful information and techniques.
An exclamation mark emphasizes warnings and other important information.
The following sections give instructions for installing and configuring GoldWave on your computer.
The minimum system requirements for GoldWave are:
If you need to edit large files, you will need a large amount of hard drive space. One minute of CD-quality sound requires 20 megabytes of storage. Editing a full CD requires at least 1GB of storage.
For editing audio in movie files and editing mp3 files, you must have a recent version of Windows Media Player installed.
The following section gives instructions for installing GoldWave on your system. Before running GoldWave make sure that you have an appropriate Windows sound driver installed. If you need to add one, use the "Add New Hardware" item under Start | Settings | Control Panel. The driver and installation instructions should be included with your sound card. The current settings for your sound card are listed under the "Sound, video, and game controllers" item of the Device Manager. The Device Manager is found under "My Computer" Properties or the System icon in the Control Panel.
If you downloaded the self-installing version of GoldWave, simply run the download. It prompts you to provide a destination folder where GoldWave will be installed. A desktop shortcut and Windows Start menu items are created automatically, if desired.
Check the readme.txt file for any additional information not available at the time this manual was created. New versions of GoldWave will be available from the web site:
To choose which audio devices to use for playback
and recording, use the properties
button on GoldWave's Control window, then choose
the Device tab. Drop down lists of installed playback and
recording devices are presented. Select
appropriate devices from the lists. Use the "PCM 16 bit"
quality setting unless your card supports the higher settings.
To associate file types with GoldWave, such as wav or mp3 files:
See Windows help for more information about file type associations.
The following sections give information about GoldWave's user interface, features, and menu structure. The first few sections provide general overviews, while subsequent sections provide details on menu commands.
GoldWave is composed of three windows: the Main window, Sound windows, and the Control window.
The Main window contains the main menu, two rows of tool bar buttons, and status bars (see Main Window figure below). It groups together and manages all the Sound windows.
The tool bar buttons provide quick access to many of the frequently used commands. The upper bar holds File, Edit, View, and Tool commands, while the lower bar contains Effect commands. The function of each button is displayed in the lower status bar when the mouse pointer is positioned directly over it. Use the Options | Tool Bar command to configure the tool bar.
The status bars show attributes of the Sound window, including the sampling rate, length, selected region, channels, and general file format information. By right-clicking the mouse pointer over any status item that has a menu icon on its right side, the unit or format for that status item can be changed. If you right-click the mouse pointer over the length item, for example, you are presented with a menu showing length in terms of storage size, time, and samples. If you right-click on the channels item, then you can select a single channel or both channels of a stereo file.
Sound windows are created when you open a file. These windows contain a waveform graph of the sound with a time axis near the bottom. For stereo sounds, two separate graphs are shown. The top green graph is the left channel and the bottom red graph is the right channel. The selected part of the sound is highlighted with a blue background between two cyan markers. A white, vertical line shows the current playback position within the sound. This line is the playback cursor.
A cue point slot is located just below the graph. Cue points are shown as inverted yellow and blue triangles. Overlapping cue points are shown in slightly different colours.
Near the bottom of the Sound window, a small Overview area shows the entire sound with the selected part in highlighted green and/or red with a blue background and the rest with a black background. A beveled frame indicates what part of the sound is currently displayed and zoomed. Initially, the entire sound is selected.
You can change the selection by using the mouse buttons or the keyboard. See Editing Overview for details. You can configure the window size and axes format of Sound windows using the Options | Window command. The Options | Colours command sets the colour scheme.
The Control window interacts directly with your sound card. It contains buttons to play and record sounds as well as controls for volume, balance, and playback speed. Real-time visuals display audio data whenever a sound is played or recorded. See Control Overview for more details.
When performing time consuming processing, such as decoding a compressed file when opening it, or encoding a file when saving it, or using most effects, a progress window appears showing the amount of processing done and the estimated time remaining to complete it. Use the Cancel button to abort processing at any time. Use the Priority setting to reduce the load on the computer's processor to give more time to other programs.
Some poorly designed (or overclocked) computers overheat when
performing complex processing, such as Noise
Reduction and saving in MP3 or other compressed formats. Reducing the
Priority setting helps to avoid thermal related errors.
In GoldWave the mouse wheel supports zooming, scrolling and selection, or playback speed adjustments. Click the middle mouse button or the wheel button to display a menu to configure the behaviour of the mouse wheel. The mouse wheel works only when the Main window is active and only on the currently active Sound window.
Zooms in and out of the waveform when the wheel is rotated up or down. The location of the mouse pointer is used as the focal point. Position the mouse over the area of interest when using the wheel. See View Menu Commands for information about viewing parts of the waveform in more detail.
When zoomed in, rotating the wheel up or down scrolls the waveform left or right. Holding the shift key moves the start marker. Holding both the shift and control keys moves the finish marker. Holding just the control key scrolls vertically, when zoomed in vertically. See Editing Overview for more information about selecting part of a file.
Increases or decreases the playback speed by changing the Speed fader on the Control window.
GoldWave displays and accepts several different time formats. The time is separated into hours (H), minutes (M), seconds (S), and fractions of a second, like thousandths (T). Two digits are given for hours, minutes, and seconds. Zero or more digits are given for the fractional part. The basic format looks like this: HH:MM:SS.TTTTT. When using this format, minutes and seconds must be numbers from 0 to 59. Only five digits can be given after the decimal point. Other supported formats are given in the following table.
| Format | Description |
| HH:MM:SS.TTTTT | Hours, minutes, seconds, and fractions of a second. MM and SS must be between 0 to 59, inclusive. A decimal must be used to separate fractions of a second from seconds. Colons must be used to separate hours, minutes, and seconds. All values are optional. A time can be entered as HH:: to specify hours only, or MM: to specify minutes only. |
| MMMMM:SS.TTTTT | Minutes, seconds, and fractions of a second. SS must be between 0 to 59, inclusive. Minutes can be larger than 59. |
| SSSSS.TTTTT | Seconds, and fractions of a second. Seconds can be larger than 59. |
| HH:MM:SS.XX/YY | Hours, minutes, seconds, and frames. This is the same as the first format, but instead of providing fractions of a second as a decimal, frames are used. The numerator, XX, specifies the frame number and the denominator, YY, specifies the frame rate, such as 30 for a 30fps animation, or 75 for CD frames. If HH, MM, and SS are not given, then XX may be greater than YY to specify any frame. Otherwise XX must be smaller than YY. Refer to the examples below. |
| Example | Meaning |
| 5 | Five seconds |
| 3:00 | Three minutes |
| 9: | Nine minutes |
| 2:: | Two hours |
| 7.1/2 | Seven and a half seconds |
| 5123/60 | Frame number five thousand one hundred twenty-three in a sixty frames per second file |
| 34:25.15/75 | The fifteenth CD aligned frame beyond thirty-four minutes and twenty-five seconds. |
| 1::.3/4 | One hour and three-quarters of a second. |
| 12:34:56.789 | Twelve hours, thirty-four minutes, fifty-six seconds, and seven hundred eighty-nine thousandths of a second. |
| .67 | Sixty-seven hundredths of a second. |
The Control window (see Control Window figure) is the interface to your audio hardware and drivers. On the bottom half of the window are visuals which display sound during playback and recording. On the top left area of the window is a standard set of audio controls, including play, stop, record, rewind, pause, and fast forward. A status visual is located just below these controls. In the top right area of the window are controls to set the playback device's volume, balance, and speed. A level visual is located just below these.
The Control window can be resized to change
the size of the visuals or to hide them. Use the
Window Menu Commands
to rearrange the controls horizontally or vertically.
The Properties
button presents the Control Properties window containing several tabs to
configure playback, recording, volumes, visuals, and devices. These options
are described in the following sections. After installing GoldWave, you
should take a moment to see if the correct playback and recording devices
are selected under the Device tab and familiarize yourself with the settings
under the Play (see figure) and Record tabs.
The Play Properties page contains options to configure the
green 
and yellow 
play buttons and set the speed for rewind and fast forward. The green and yellow
play options are given in the table.
| Option | Button function |
| All | Plays entire sound. |
| Selection | Plays region between start and finish markers (the selected part of the file). |
| Unselected | Plays regions outside the start and finish markers. This lets you quickly test how a cut or delete will sound without actually changing the sound. When possible, playback is confined to the region shown in the Sound window view so that the entire sound does not have to be played. |
| Cursor | Continues playback from the current cursor's position and stops at the finish marker if it is located after the cursor. Otherwise playback continues to the end of the file. |
| Cursor to end | Continues playback from the current cursor's position and stops at the end of the file. |
| View | Plays all of the sound currently shown in the Sound window view. This is useful if you zoomed in on part of the sound. |
| View to end | Starts playback at the left side of the sound currently shown in the Sound window view and continues playback to the end of the sound. |
| Finish | Plays three seconds just before the finish marker, so you can determine if that marker is in the right place without listening to the entire selection. |
| Intro/loop/end | This is a special playback feature that plays the sound in three sections. The beginning of the sound, outside the selection, is played first. Then the selection is played and looped. Finally the end of the sound, outside the selection, is played. This is useful for musical accompaniment or looped instrument samples. |
| Loop | If checked, it specifies the number of times playback should be repeated. A value of 1 loops playback once, so the selection is played twice. A zero value loops forever. |
Fast/Rewind Speed
The playback speed of the fast forward and rewind buttons is
controlled by these values. A value of 1.00 is normal speed.
Entering a value of 3.00 for Rewind speed, for example, means the
rewind button will play the sound backwards three times faster than
normal. By entering small numbers (such as 0.1) the rewind and
fast forward buttons will play very slowly. This is useful for
finding pops or clicks, since the visuals will move slowly
through the data.
Marker preview
Marker preview (scrubbing) plays a very short section of audio
whenever an edit marker is
moved with the keyboard
arrow keys. The
value specifies the amount of time to play just after
the start marker or just before the finish marker. Set the time to 0
to disable previewing.
The Record Properties page includes features to monitor the recording sources, start recording automatically when a sound is detected, or delay recording for a certain length of time. The basic options are given in the table.
| Option | Purpose |
| Monitor input | Connects the recording source to the visuals so you can adjust volume levels before recording. See Recording Sounds for information about selecting a different recording source and setting volumes. |
| Ctrl key safety | Prevents you from accidentally recording over a sound. To record, you must hold down the Ctrl key, otherwise a safety message appears. |
| Set finish marker at stop | Automatically moves the finish marker to the place where recording stopped. This makes it easier to trim the file after recording. |
| Show settings window | Displays an information window whenever recording is started. The window gives the current recording setup, including the recording device, the selected inputs, and other settings. Click on a label link to change the setup. Recording may be stopped when the setup is changed. |
| Filter dc offset | Automatically filters a dc offset from the recorded audio. Use this option if you see a lot of activity on the low frequency bars and VU meters even when recording silence. |
| Auto save |
Automatically saves the file when recording ends.
Bounded recording mode must be selected.
If recording is manually
stopped, the file is not automatically saved. Use this option with
the Timer option to save
a recording after a scheduled event.
If you start recording in a new, untitled file, you will be prompted to provide a filename so that it can be saved under that name automatically. The default save format is used for the file type and attributes.
|
| Power down system | Automatically turns off the computer after saving the recording. Use this option with the Auto save and Timer options to shutdown the computer after a scheduled recording. This option is available on Windows 2000/XP or later. |
| Bounded to selection | Records within the selection only. Recording stops automatically at the end of the selection. If you stop recording before the end is reached, the rest of the selection is replaced with silence. Use this option to record for a fixed length of time. |
| Bounded and looped | The is similar to the above option, but recording restarts automatically when the end is reached and continues to record over and over until the stop button is pressed. This is useful if you are trying to capture a sound but do not know when it might occur. By loop recording a 1 minute sound, you will always have the last minute of audio stored for recall. |
| Unbounded | Recording starts at the start marker's position and continues recording until all storage is exhausted or until you press the record stop button. The file size is increased automatically to hold the new audio. This is useful if you do not know how long the recording will be. |
Delayed Recording
The Timer option delays recording until the specified time and day of
the week. Use this feature to automatically record something at
a later time. The time is given in 24 hour time. A time of 06:00:00 is
6:00 AM and a time of 18:00:00 is 6:00 PM. 00:30:00 is 12:30 AM or
30 minutes past midnight. When entering the time, remember
to include the seconds. Entering 18:00 means 00:18:00.
You must press the record button to activate the timer.
Example
Record for 35 minutes on Tuesday at 7:00 PM:
Remember to press the record button to activate delayed recording
(timer or level activated).
Turn off any power management settings that may power down the computer.
Level activated recording is useful for automatically synchronizing recording to a sound source or efficiently capturing airport or police radio communications containing mostly silence that does not need to be recorded. It automatically starts recording when the sound source is above a given level and pauses recording when the sound is below the level. The Threshold specifies how loud a sound should be before recording begins. A value less than 0.2 is typical. The Duration time specifies how long to record after the sound becomes quiet again. Using a value of 3 allows recording to continue for three seconds after the sound goes below the specified threshold. To minimize silence, use a value of 1 second or less, but not zero. A zero value causes recording to continue without stopping once triggered. The Prebuffer time specifies the amount of audio to store prior to activation. When activation occurs, the prebuffer audio is inserted before the currently recorded audio, allowing you to hear the sound slightly before activation. The Time stamp cues option provides a way of marking the date, time, and position of each activation. Cue points with the current date and time are created and can be view under the Cue Points tool. Use the edit box to specify the format for the date (this is done using the C strftime function). Some format specifiers are given below.
| Specifier | Purpose |
| %a | Short weekday (Sun, Mon, ... ) |
| %A | Weekday (Sunday, Monday, ... ) |
| %b | Short month (Jan, Feb, ... ) |
| %B | Month (January, Febuary, ... ) |
| %d | Day of the month (01 to 31) |
| %H | Hour in 24-hour clock (00 to 23) |
| %I | Hour in 12-hour clock (01 to 12) |
| %m | Numerical month (01 to 12) |
| %M | Minutes (00 to 59) |
| %p | "AM" or "PM" |
| %S | Seconds (00 to 59) |
| %y | 2 digit year (00 to 99) |
| %Y | Year, all digits |
| %Z | Time zone name |
| Example | Cue name generated |
| %d %b %y, %H:%M:%S | 12 Jan 05, 14:23:56 |
| Date: %A, %B %d, %Y. Time: %I:%M:%S%p | Date: Wednesday, January 12, 2005. Time: 02:23:56PM |
| Today at %I:%M%p | Today at 02:23PM |
| %Y-%m-%d at %H=%M=%S (Easy to sort and safe for filenames) |
2005-01-12 at 14=23=56 |
The Volume Properties page lets you adjust recording volumes and select or unselect recording sources. Be sure to select the correct volume device that corresponds with the recording device selected under the Device tab.
A volume fader, edit box, and checkbox is shown for each source. To select a source, check the appropriate checkbox. If your sound card supports a master control, make sure that the Mute all option is turned off and that the master volume is not zero.
You can use the Monitor input option under the Record tab to activate the visuals without recording.
Note that volumes are changed regardless of whether you choose OK or Cancel to close the Properties window.
To select a different recording device, use the Device tab.
The Visual Properties page controls what visuals are used in what location. There are four visuals: status, level, left, and right. The status visual is located at the upper left side. It displays elapsed time and playback and recording status. The level visual is located at the upper right side. It graphically displays the current output or input level. The left and right visuals display audio in a variety of ways, as listed in the table.
| Visual | Description |
| 3D Bars | A 3 dimensional logarithmic frequency 10 band bar graph. |
| Analog Meter | A scaled needle meter. |
| Bars | A logarithmic frequency 10 band bar graph, commonly found on stereo systems. |
| Blank | Disables the visual and may improve performance on slower systems. |
| Blowing Inferno | Fire coloured, double-sided spectrum graph. |
| Spectrogram | Coloured frequency spectrum, with time on the x-axis, frequency on the y-axis and colour as the magnitude. The colours, in increasing magnitude, are black, purple, blue, cyan, green, yellow, red, and white. A cyan point, for example, is higher magnitude than a blue point. |
| Spectrum | Frequency analysis of the sound. |
| Spinning Logo | A rotating GoldWave logo that tilts in response to the audio. |
| VU Meter | Peak and current amplitude level meter. |
| Waveform | Standard amplitude waveform, much like the 1:1 zoom level in a Sound window. |
| X-Y Graph | The sound is plotted with the left channel against the right channel to generate Lissajous patterns. This is often used to see the phase difference between two equal frequency signals. If the left and right channels are in phase, the pattern is a diagonal line running from the lower left to the upper right. If the channels are 90 degrees out of phase, the pattern is a circle. For general stereo sounds, it looks like a crazy scribble. The larger the scribble, the larger the difference between the channels. Monaural sounds always show a diagonal line since the left and right data are the same. |
Some visuals have properties you can set, such as axes ranges, colours, display modes, etc. Right-click on a visual and select Properties from the popup menu to see these properties. You can resize the Control window to make the visuals larger or smaller.
Quick Select Menu
Use the quick select menu list to select your favourite visuals. The selected
visuals appear in the popup menu when you right-click on a visual in the Control
window.
Frame Rate
The frame rate sets the number of times per second that
visuals are updated and drawn. A value of 60 or less
gives good results, but you may want to use higher values to get an extra
detailed spectrogram. The actual frame rate is limited by your system's
speed. Use a lower frame rate for older systems.
FFT Window
When calculating any frequency related graphs, a window function
must be used to smooth out analysis at the endpoints. The Kaiser 7 or Hamming
windows are usually the best, but you can try the other windows
just to compare the results.
The Device Properties page contains drop down lists for playback and recording devices. Normally you do not need to change these settings unless you have more than one sound card or you encounter a problem.
The Quality settings control the bit depth to use. Most cards support 16 bit quality, but more recent cards support 24 bit quality. Sound card drivers on recent versions of Windows should support the IEEE 32 bit floating point setting, but often that is not the case, so that setting should not be used until properly supported by the industry (probably never).
The playback Prebuffer/latency fader sets the amount of audio to store before sending it to the device. Using a higher value may eliminate gaps and stutters on a slow system, but it increases the delay between changing effect settings and hearing those changes during previewing. Using lower values makes effect previewing more responsive, but may cause gaps and stutters if the system is too slow to process all the audio or emulated DirectX drivers are selected. This setting does not apply to recording. Values between 0.4 and 0.8 are recommended.
The Alternative playback initialization option helps to avoid some of the many problems with sound drivers and can eliminate deadlock (freezes) when previewing certain DirectX Audio Plug-ins. It eliminates a loud click that occurs at the start of playback on some drivers. This option causes the DirectX sound buffer to be recreated each time playback is started rather than reusing the same buffer. It begins playback on a separate thread without pausing or holding up the main program thread. This makes the interface seem more responsive when starting playback.
The Mono source options specify the channel to use when recording a mono file: Left channel captures audio on the left channel only; Right channel captures audio on the right channel only; Average captures audio on both channels and averages it into a single channel.
Joystick control allows playback and recording to be controlled using a game controller or a foot pedal. The first controller detected is used. The following table lists the modes of operation.
| Mode | Description |
| None | Joystick control is disabled. |
| Game controller | The main directional pad controls playback. Left is rewind, right is fast forward, down is pause, and up unpauses. The first button (button 1 or A) starts or stops playback. The second button (button 2 or B) starts or stops recording. |
| Foot pedal, hold | A three switch pedal controls playback, rewind, and fast forward. The pedal must be held down. Playback pauses when the pedal is released. |
| Foot pedal, toggle | A three switch pedal controls playback, rewind, and fast forward. Tapping the pedal starts or pauses playback. |
Use the Configure button to configure foot pedal controls. Different brands of foot pedals use different switch combinations. After choosing the Configure button, hold down each pedal for two seconds starting with the play pedal, then the rewind pedal, and finally the fast forward pedal. If a rewind or fast forward pedal is not available, wait for five seconds and that pedal will be skipped. The OK button becomes enabled when all pedals are assigned or skipped.
Use the Troubleshoot button to test audio devices and list status information. The information may be helpful in tracking down recording or playback problems.
After opening a sound (see File | Open), you use the
green play
button
or the yellow play
button
to play it. Both buttons play the selection by default, but you can configure these
buttons under the
Play tab of the Control Properties window to
play different parts of the file.
Right-click on the button to quickly access some of those settings.
To start playback at any point in the sound, click on the time line under the waveform in the Sound window or right-click on the waveform and choose the Play From Here command from the popup menu. You can right-click-and-drag to select a part of the sound to play as well.
While a sound is playing, it is displayed on the visuals. The current position is displayed in the Sound window as a white, vertical line on the waveform (cursor). You can move the start and finish selection markers to the playback position by using the bracket keys, [ and ] or Edit | Marker | Drop... commands. See Editing Overview for more information about changing the selection. You can set cue points by using the Ctrl+Q key or the Edit | Cue Point | Drop Cue command.
To play the entire sound, configure the Yellow play button to All.
To play the selection, configure the Yellow play button to Selection.
While a sound is playing, you can pause it with the pause
button.
Remember to use either play or stop later. Pause freezes the visuals
and the current position marker so you can see the shape of the sound in
the visuals or move the selection markers.
Playback can be stopped immediately with the stop
button. The current
position is reset to the beginning. Note that recording is stopped using a
different button.
Use the rewind
button or fast forward
button to quickly move back and forward through the sound. The current
position is displayed in the
Sound window
as a white, vertical
line on the waveform. You can adjust the speed of rewind and fast forward with the
Play tab of the Control Properties window, as described
previously.
Use the record
button to record your own sounds. Before you start, you
need to create a new file using the File | New
command. You can make room for recording in an existing sound
by using the Edit | Insert Silence
command. Audio is recorded into the selection of the
Sound window
replacing any audio that was
previously there. Recording stops automatically when the end of the
selection is reached (bounded mode) or
when no more storage is available (unbounded
mode). You can stop recording at any time with the
recording stop
button. A recording pause
buttons appears in place of the record button so
that you can pause recording.
Many recording options are available in the
Record tab of the Control Properties window.
Right-click on the record button to quickly access some of these options.
Sound cards usually have several recording sources, such as microphone, line-in, CD, and MIDI. To select and adjust a recording source, use the Volume tab of the Control Properties window or the Windows Volume Control accessory. To use the Volume Control accessory, use Start | Programs | Accessories | Entertainment | Volume Control (it may be under Multimedia). Next, use the Options | Properties command in the Volume Control window. In the Properties window, select the Recording option and make sure all the items in the list are checked. Finally, choose OK to see the Recording Control. You can select a recording source and adjust the volume. To adjust the volume before recording, use the Monitor input option under the Record tab of the Control Properties window.
Remember to press the playback button on the cassette player, record player, or CD player when recording from an external device. See the Appendix D for a tutorial.
If you want to record vocals over existing music, you'll need to use two files in GoldWave. You can record in one file while playing the other. Some sound cards have a "Stereo Mix" (or "What U Hear" or similar) recording source you can select that will allow you to record both the microphone input and the playing sound at the same time. If that source is not available, you'll need to mix the two files after recording instead.
For long recordings, turn off any power management settings that may
power down the computer.
Use the top volume fader to change the playback volume. Move the fader right or click the plus button to increase the volume. Move it left to decrease the volume. The current volume is shown numerically in a popup tip window to the left of the fader. A value of 100% is full volume.
Use the middle balance fader to change the left/right balance. Move the fader in the direction you want to shift the balance. Right-click on the fader to display a popup menu to quickly set the balance left, right, or center.
Note that these faders do not change the recording volume. See Recording Sounds for more information.
The playback speed of the audio device can be changed with the bottom speed fader. Move the fader right to increase the speed, and left to decrease it. The relative speed is shown numerically to the left of the fader in a popup tip window. Right-click on the fader to display a popup menu to quickly set the speed to commonly used ratios. Note that changing the speed also changes the pitch like spinning a vinyl record faster or slower.
Almost all commands in GoldWave operate on the currently selected part of a sound. The selected part, or selection, is the highlighted part of the sound graph between two vertical markers (see Main Window figure). The vertical markers are cyan lines located to the left side (start marker) and right side (finish marker) of the view.
GoldWave provides several ways of setting the selection. You can:
If you just click the left mouse button without dragging, the start marker is moved. If you just click the right mouse button, a context menu appears, which can be used to start playback at any position. If you click-and-drag with the right mouse button, you can play or zoom in on that area without altering the current selection.
Additional notes and techniques:
You can redraw the waveform with the mouse to remove pops/clicks or other small defects. To do this, you must first zoom in so that individual samples are visible (see View | Zoom 1:1 or View | Zoom 10:1).
Cutting and pasting audio in GoldWave works much the same way as cutting and pasting text in a word processor. Mixing and cross-fading involves combining two or more sound together so that they play at the same time.
The Edit | Cut command removes sections of audio. The Edit | Paste command inserts sections of audio from the clipboard. Before you can paste, you need to use Edit | Cut or Edit | Copy to place some audio into the clipboard.
To join several files together:
The Edit | Mix command mixes one sound with another so they both play at the same time.
To add vocals to music:
When mixing more than a couple of sounds, you should reduce the
mixing volume and the destination volume to prevent clipping
distortion. The volume of the destination sound can be reduced
before mixing by using the
Effect | Volume | Change Volume
command.
A crossfade occurs when one sound fades out while another sound fades in. Radio stations often use crossfades to fade out the end of one song while fading in the next song so there is no break in the music. GoldWave's Edit | Crossfade command does the same thing by using the clipboard audio as the second song. The entire song to fade in must be copied to the clipboard before using the command.
To do a crossfade in GoldWave, follow these steps:
In some cases more control is needed. To do a crossfade manually in GoldWave, follow these steps:
GoldWave supports both hard drive based editing and RAM based editing. These features are described below. Hard drive storage is enabled by default. Use the Options | Storage to configure the storage mode. For uncompressed files, GoldWave will read the audio directly from the original file. It does not copy a file to temporary storage until it is edited or modified. The original file is not changed until it is saved. For most compressed files, the data has to be decompressed to temporary storage when the file is opened.
Working with compressed files may take much more storage than expected. MP3 files, for example, have to be decompressed into temporary storage before GoldWave can edit them. Such files may require over 20 times the amount of compressed storage when opened. A 10MB MP3 file could require over 200MB of storage space.
In hard drive based editing, the entire sound is stored in a temporary file on your hard drive where it can be modified. This allows you to edit huge files provided the required drive space is available. Only a small amount of RAM is required for each opened sound. The drawback is that editing and effects processing take more time since audio data must be transferred to and from the drive.
In RAM based editing, the entire sound is stored in your computer's memory. This allows you to edit and process files very quickly. It saves time and reduces the load on your hard drive. The drawback is that the size of the files must be small enough to fit in the available RAM. If you edit or record large files, Windows will start swapping memory to the hard drive, which significantly degrades performance and may cause defects when recording. Also note that in the event of a system crash, it will not be possible to recover a file stored in RAM. Editing files larger than 1GB in RAM is not recommended due to limitations in Windows.
This section explains file formats and gives general information about how files are handled by GoldWave. Several features for storing and handling files can be configured using Options | Storage and Options | File Formats.
Sound files come in a variety of forms. Usually, the form or type of sound can be determined from its filename extension, such as .wav or .mp3. GoldWave supports all the sound types listed in the Supported File Types table, and more depending on what file format plug-ins are installed. Each file type can have several sub-formats or attributes. The .wav type for example, can hold audio encoded or compressed in dozens of different ways, including PCM, ADPCM, companded, or MPEG1 Layer 3.
| Extension | Comments | Cues | Info | >4GB |
|---|---|---|---|---|
| .aif .afc |
Apple / Macintosh sound files. Compressed files are not supported. Cue points are supported. Limited support for file text information. The blocks NAME, COPY, ANNO, and AUTH are preserved. | |||
| .ape | Monkey's Audio compressed files. Requires the APEFile plug-in. | |||
| .asf .avi |
Microsoft audio and/or video files. GoldWave can extract the audio portions of these files, but cannot save or create them. | |||
| .au | Sun or NeXT files, commonly used on web pages and in Java. Supports 8 & 16 bit linear, mu-law and A-law encoded files. No support for file text information. | |||
| .dwd | DiamondWare sound files. GoldWave supports 8 & 16 bit PCM attributes through a plug-in. No support for file text information. An external plug-in for this format is included in the plug-in SDK. | |||
| .flac | FLAC sound files. This is a lossless compressed format. GoldWave supports 8, 16, and 24 bit attributes with low (fast), medium, and high (slow) modes of compression. | |||
| .iff | Amiga 8SVX files. The blocks NAME, COPY, ANNO, AUTH, and CHAN are all preserved. Limited support for file text information. | |||
| .mat | Matlab files. The data must be normalized (i.e. -1.0 to 1.0) for double precision data. If the "wavedata" variable is two dimensional, the data is assumed to be stereo. GoldWave saves audio data in the "wavedata" variable and the rate in the "samplingrate" variable. A 11025Hz sampling rate is assumed if none is present. No support for file text information. | |||
| .mov | QuickTime movie files. If Apple's QuickTime player or iTunes software is installed, GoldWave extracts the audio portion from the file (if present). See the .mp4 and .m4a types for more information. | |||
| .mp3 | MPEG1 Layer 3 compressed sound files. To read these files, you must have an MPEG decoder installed (usually included with Windows). To save a file in this format, you must have the LAME encoder installed. See the GoldWave website for details. File text information is supported in ID3v2 tags. The ID3v1 tag is read, but not written. | |||
| .mp4 .m4a |
Unencrypted iTune/MPEG4 sound files. If Apple's QuickTime player or iTunes software is installed, GoldWave will open these files through the QTFile plug-in, which is included with GoldWave. Files cannot be saved or created. File text information is read from the files. | |||
| .ogg | Ogg Vorbis compressed sound files. Gives better quality compression than MP3. Refer to the Vorbis website for more information. File text information is supported. | |||
| .raw | Headerless files containing binary data in 8 bit, 12 bit, 16 bit, 24 bit, 32 bit, single or double precision IEEE, mu-law, or A-law format. | |||
| .sds | MIDI instrument sample dump standard format. Loop points are not supported. No support for file text information. | |||
| .smp | Sample Vision 16 bit PCM sound files. Markers/Loops are not supported. No support for file text information. | |||
| .snd | Raw or NeXT sound files. NeXT files are automatically detected. Opening Raw files displays the File Format window for attributes. No support for file text information. | |||
| .txt | An ASCII text file containing a series of Y values (amplitudes) in human readable form. Values range from -1.0 to +1.0 for floating point data and -32768 to +32767 for integer data. No support for file text information. | |||
| .wav | RIFF WAVE 8 to 32 bit PCM mono or stereo, A-law encoded, mu-law
encoded, and Microsoft ACM compressed files. MPEG compressed
audio is support only if the MPEG decoder is installed.
Only files with one 'data' chunk are supported. The chunks LIST INFO, LIST adtl, and 'cue' are detected. All others are ignored. Cue points are supported. File text information is supported. |
|||
| .wma | Windows Media Audio files. Supports several different bitrates from low bandwidth to high quality lossless. File text information and cue points are supported. | |||
| .voc | Sound Blaster files. Supports: 8 bit mono/stereo, 16 bit mono/stereo, mu-law encoded mono/stereo. ADPCM compressed files are not supported since the compression algorithm must be licensed from Creative Labs. No support for file text information. | |||
| .vox | Dialogic ADPCM encoded raw files. The File Format window is presented where you can specify the Telephony type and 4 bit VOX ADPCM format. You can use the Options | File Formats command to assign a default format for .vox files. No support for file text information. |
| - Supported | - Limited support | - Unsupported |
Normally, GoldWave detects and automatically opens all the supported file types. However, there are several cases where GoldWave may not be able to open a file:
If any of these conditions occur, GoldWave displays the File Format window (shown below) so that you can specify the type and attributes manually. GoldWave lists all the file format plug-ins that support reading raw audio data. If you are working with PCM or uncompressed binary data (like CD audio), select the Raw type. If you are working with Telephony files, select the Dialogic type. Other types may be listed depending on what plug-ins you have installed.
| Format | Description and Attributes |
| PCM | Audio is uncompressed 8, 12, 16, or 32 bit data. A Windows system usually creates 8 bit, unsigned or 16 bit, signed, little endian data. A Macintosh system usually creates 8 bit, signed and 16 bit, signed, big endian data. The signed attribute tells GoldWave how the bits should be interpreted. The endian attribute tells GoldWave the byte ordering of the data. Big endian has the most significant byte first. Little endian has the least significant byte first. |
| Telephony | Audio is in a compressed format used in telephone applications. This includes mu-law, A-law, ISDN A-law (inverted A-law), and 4 bit ADPCM VOX Dialogic files. |
| Floating point | Audio is binary IEEE floating point single precision (32 bit) or double precision (64 bit) data. The byte ordering is usually little endian. |
| Text | Audio is a plain text (ASCII) file containing numbers. The float attribute tells GoldWave that the numbers range from -1.0 to 1.0. The integer attribute tells GoldWave that the numbers range from -32768 to 32767. |
| Encoded | Audio is compressed using an encoding algorithm. Such files cannot be opened properly through the File Format window and require a separate plug-in for decoding. |
If you do not know the format, experiment with trial-and-error. Appendix A has more information about sound attributes. Start with an 8 bit or 16 bit PCM attributes, then try the mu-law or A-law formats. Generally, sounds will be noisy if the format or number of bits is incorrect, in which case you will have to close and reopen the sound using a different format. You can leave the sampling rate unchanged since it affects only the playback speed and can be changed later using Effects | Playback Rate.
GoldWave supports external file format plug-ins for opening and saving files. These plug-ins are created by other developers by using the GoldWave Plug-in Development Kit to handle file types that GoldWave does not support directly.
You can use Options | File Formats to enable and disable plug-ins or to change the order in which they are used.
When you open a file in GoldWave, these steps are followed:
Effects modify, enhance, and change sounds in a variety of ways. These commands are similar to font menu commands in word processors. For example, using font commands, you can change the size of the letters. In GoldWave, using the Volume | Change Volume effect changes the "size" of a sound. Changing the colour of a font would be similar to changing the pitch of a sound.
For an introduction to some of the terms used in this section, refer to the Editing Overview section and Appendix A. A variety of volume scales may be used by effects.
Most effects in GoldWave are cumulative. This means that if you use the same effect with the same settings, then the sound is changed each time. For example, if you use the Volume | Change Volume effect with a value of -6.02dB, then the volume of the sound decreases to half its current level. If you use that effect again, the volume decrease again, giving one quarter the original volume.
Another example is the Time Warp effect. If you specify a change of 50%, then time is slowed to half and the sound is twice as long. Using the effect again at 50% makes the sound four times as long.
There are a few exceptions. The Maximize Volume effect has an absolute setting. Maximizing the volume to 0dB sets the sound's peak volume to 0dB. Using the effect again at 0dB has no affect.
Many effects have similar controls such as presets and shape boxes. These are explained below.
Presets store settings, parameters, and
shapes (described below) for quick retrieval
the next time
the effect or command is used. Controls for presets consist of a drop down list
box, an add 
button, and a remove 
button,
as shown in the
figure above.
To add a new preset:
button.To delete a preset:
button.To change a preset:
button.Several effects in GoldWave use Shape Controls to set graphical parameters or dynamically alter the effect across the selection. Shape Controls usually consist of a graph window and a set of controls, including a point number box, an add point button, a remove point button, an X value box, and a Y value box as shown in the figure above.
Graph Window
The graph window initially contains a single line with two
endpoints, shown as large dots. By clicking the left mouse button
anywhere inside this window, you can add new points to bend the
line into a variety of zigzag shapes. To move a point, click on it
and drag it to a new location. To remove a point, click the right
mouse button over the point. Note that endpoints cannot be
removed.
Controls
Points can be added, moved, and removed by using these controls. Use the
Point box to select the current point. Change the X and Y
values to move the point. Use the add point button to insert a
new point between
the current point and the next point. Use the remove point button
to remove the current point, except if it is an endpoint.
Some dynamic effects, such as Doppler, Pan, and Shape Volume start previewing audio based on the current point's time value. If the X value of the current point is 1:00, for example, then preview playback starts at that time rather than at the beginning of the selection. This lets you preview the point's settings without playing the entire selection.
To save a shape, use the Presets controls, explained above.
Previewing is a way of listening to how the current effect settings
will sound without having to process the entire
selection first.
Preview Controls consist of a play
button and a stop
button. When
you press the play button previewing usually starts at the beginning
of the selection.
In some cases, such as effects that have time
based shapes, previewing will start at
the current point's time rather than at the beginning of the
selection.
In most cases, you can change the effect settings while the preview plays and hear the changes as they are made. For more complex effects and most shape based effects, you will need to use the Apply button to apply the changes or shape while previewing.
The Apply button is used only for previewing.
It does not apply the effect to the sound file. Use the OK button.
If any errors or exceptions occur while using
a plug-in, you'll need to contact the plug-in creator
for assistance. GoldWave Inc. does not provide support for
plug-ins created by separate developers.
This section outlines the accessibility features in GoldWave. Almost all of GoldWave's fuctionality is accessible through the keyboard. See Keyboard Commands and Options | Keyboard.
The status visual in the Control window becomes a text box that gives the elapsed time, the current status, and the peak levels (in parentheses). The peak levels are updated only while holding the Ctrl key. When the key is release the highest level is shown. The level is reset whenever the Ctrl key is press again. The levels are given in percentages, but tabbing to the text box and pressing D changes it to decibels. Pressing D again changes it back to percentages.
All images are removed from the menus so that standard Windows text menu items are used instead of custom drawn ones, ensuring that menus are easily readable. If menus cannot be read, be sure to uncheck the "Tool bar images in menu" box under Options | Tool Bar.
Almost all windows in GoldWave provide edit boxes where values and settings can be entered rather than relying on visual controls only. Use the Tab key to move between controls.
Use Alt+F6 to switch between GoldWave's Main window and Control window. Use the Tab key to move through the buttons and faders. The arrow keys change the fader settings. Unless you need to access the volume, balance, or speed faders, it is best to dock the Control window. Use the Tool | Control command to dock or undock it.
Finding certains parts of a sound file is done through keyboard navigation, which involves playing the file and moving the playback cursor until the area if interest is located. GoldWave includes many keys for playing different parts of the sound. They are listed in the following table. The amount that the cursor is moved depends on the zoom level, explained in the View section below.
The View | Auto Scroll Lock menu item must be checked for keyboard navigation to work.
| Keystroke | Action |
| Space | Starts playback or stop it (toggles playback). The region that is played depends on the Green play button settings. |
| J, K, L | Rewinds, plays, and fast forwards respectively. Playback always starts at the cursor's position. |
| Shift+J, Shift+K, Shift+L | Makes the playback speed slower, normal, and faster respectively. |
| F4, F5, F6, F7, F8 | Plays, rewinds, fast forwards, pauses, and stops respectively. The Green play region is played. |
| H | Starts playback relative to the mouse's horizontal position in the waveform. |
| Shift+[ | Plays three seconds of audio up to the start marker. |
| Shift+] | Plays three seconds of audio up to the finish marker. |
| Ctrl+[ | Plays from the start marker to the finish marker. |
| Ctrl+] | Plays from the finish marker to the end. |
| Left | Move the cursor backward. |
| Right | Move the cursor forward. |
| Page Up | Moves the cursor one screen backward. |
| Page Down | Moves the cursor one screen forward. |
| Home | Moves the cursor to the start marker's position. |
| End | Moves the cursor to the finish marker's position. |
| Ctrl+Home | Moves the cursor to the beginning of the sound. |
| Ctrl+End | Moves the cursor to the end of the sound (playback stops). |
In addition to using the Edit | Marker | Set and the Edit | Marker | Drop... commands, there are a number of ways of changing the selection with the keyboard. They are listed in the following table. The amount that the selection markers are moved depends on the zoom level, explained in the View section below.
| Keystroke | Action |
| Shift+Left, Shift+Right | Moves the start marker left or right (backward or forward). See the Marker preview playback setting. |
| Ctrl+Shift+Left, Ctrl+Shift+Right |
Moves the finish marker left or right. |
| Ctrl+A | Selects the entire sound. |
| Shift+Home | Moves the start marker to the beginning of the sound. |
| Shift+End | Moves the start marker to the finish marker's position. |
| Ctrl+Shift+Home | Moves the finish marker to the start marker's position. |
| Ctrl+Shift+End | Moves the finish marker to the end of the sound. |
| [ (left bracket) | Drops the start marker at the playback cursor. |
| ] (right bracket) | Drops the finish marker at the playback cursor. |
After deleting or cutting the selection, the selection is empty, so if the playback buttons are set to play the selection, nothing will play. Use Ctrl+A to select the entire file.
Keep in mind that most of the playback keys stop playback at the finish marker and there may be more audio beyond it. Use Edit | Trim to remove any audio outside the selection, if you only want to save what is in the selection.
If recording is stopped, the finish marker is moved to the stop time. To continue recording from that point, use Shift+End to move the start marker to the finish marker, then use Ctrl+Shift+End to move the finish marker to the end. Starting recording will continue from where it was stopped.
The view and zoom level control how much audio is displayed on the screen (or page). By default, GoldWave displays the entire file. Use the View Menu Commands to change the amount of audio displayed. This also changes the amount the playback cursor and selection markers are moved with each keystroke. The cursor is moved one tenth of the screen size. Selection markers are moved one hundredth of the screen size. Using View | 10 Seconds displays 10 seconds of audio in the view. The Page Up or Page Down key moves the cursor 10 seconds backward or forward. The Left or Right key moves the cursor 1 second backward or forward. The selection keys move the start or finish marker a tenth of second (0.1). Using View | 1 Second displays 1 second of audio. At that level, the page, arrow, and selection keys move 1 second, one tenth of a second, and one hundredth of a second respectively.
The Shift+Up and Shift+Down keys zoom in and out by 75%. The status bars display the selection range and the current zoom level.
Use Window options to change the default initial zoom level when a file is opened.
This section explains commands under the GoldWave's File menu. The File Overview section provides general information about how GoldWave handles files.
Use New to create a new sound with attributes you specify. These attributes are discussed in Appendix A. Note that GoldWave allows you to create and edit sounds that may not be playable with your audio hardware. For CD quality, use stereo, with a sampling rate of 44100Hz. Several commonly used settings are provided in the Presets list. Use the Initial file length box to specify the time length of the file. You can alter the length later with Edit | Trim or Edit | Inserting Silence.
To change the default save format shown in the status bar, see File Formats. To change the recording bit depth, see Device Properties
The Open command presents a list of files in your sound folder. The sound folder can be set using the Options | Storage command. All recognized file types are listed. After you select a file, a Sound window is opened and details about the sound are displayed in the status bar. See the File Format section above if GoldWave could not open the file.
The Storage Overview section explains how the files are stored for editing. Depending on the size of the file, you may want to change the storage setting under Options | Storage.
Use Close to close the current sound. If any changes were made, you are asked to save them.
Use Close All to close all sound windows. If any changes were made to any of the sounds, you are asked to save each one.
This command assigns or changes text information stored in the file, such as artist, title, copyright, and date. Information is stored in certain file types only, such as .wav, .wma, .aiff, .ogg and .mp3. Some file types only store a subset of all the items given in the File Information window. There is no verification of the information entered. It is up to you how to use these items and to follow any guidelines required for a particular file type.
Any additional information or metadata within a file not shown in this window
is not preserved by GoldWave.
Use the Copy and Paste buttons to copy information from one file and paste it into another.
The sound is saved in a file using its original name and type. If memory or disk space is low, the file may not be saved successfully. GoldWave will inform you if this happens. If Save fails, try deleting some unneeded files or close other applications. Make sure that the file is saved successfully before closing GoldWave, otherwise the changes will be lost. Note that audio from video and movie files cannot be saved. You must save those files in an "audio only" format. You will see the Save As window if you need to save the file in a different type.
Cue points are saved only in certain file types. If you added
cue points to a file that does not support them, you can use
File | Save As to save it in
a different type or save them in a separate
Cue File.
An option to confirm saving is given under the Options | Window command.
Save As saves a sound using a different filename or file type. To save the sound using a different name, simply type in the new name in the File name box. To save the sound using a different type, select the type from the "Save as type" list box, then select attributes from the Attributes list box. If a file type supports customized attributes, then the Attributes label will become a link you can choose to display a configuration window.
Since each file type supports different attributes, always select the type before selecting attributes. Java and Web sounds, for example, should be saved using the "Sun (*.au)" type and the "Java/Web" attributes.
You can use Save As to compress a sound as well. Select the "Ogg (*.ogg)" type and one of the many Vorbis attributes. The smaller the bitrate (kbps number), the smaller the file will be. Note that quality may be reduced as well.
Use Options | File Formats to assign a default format if you always prefer to use a specific file type and attributes.
Unless the number of channels or sampling rate has changed, GoldWave does not
use the compressed audio data after saving. It continues
to use the original audio stored in temporary storage. You must close
and reopen the file in GoldWave (or play it in a separate program)
to hear how the compressed audio sounds. It is strongly recommended that
you listen to the compressed file before discarding the original to
ensure its quality is acceptable.
The correct type must be selected from the type box.
Typing in a different extension by hand for the filename does
not convert the sound to the type associated with that extension.
Saves all modified sounds. Since saving all sound can take some time, you are asked for confirmation before all sounds are saved.
Save Selection As saves the selected part of the sound to a file. Use this command to save parts of a large file. The Save As window appears where you can specify the new filename, type, and attributes for the file.
Batch Processing converts a set of files from one format to another and applies effects. Use this feature to compress all your .wav files to .mp3 to save disk space or convert from .mp3 to .wav to prepare for writing to a CD-R audio disc. You can remove pops/clicks, equalize, and maximize volume levels of all your vinyl album or tape recordings. Or you can add copyright and other text information to all your own files.
Add files with the Add Files button or drag-and-drop a group of files from Windows Explorer. Add an entire folder (including subfolders) or all the files in a folder of a specific type by using the Add Folder button. Remove items from the list by selecting one or more of them and choosing the Remove button. The Remove All button removes all files and folders from the list.
The tabbed pages along the bottom let you configure conversions, effects, destination folder, and file information. These are explained below.
When everything is configured, you can use the preset controls to save all the settings, then choose the Begin button to start processing all the files. A status window will appear showing the progress and listing any errors that occur.
Use the Apply button to start processing directly on the current sound already opened in GoldWave. Only the items listed on the Process tab will be applied to the sound. The sound is not converted and file information is not changed. This button is enabled only when a sound is opened, there are items in the process list, and no files have been added to the file list.
If you are trying to convert raw files, such as .vox,
you'll need to use
Options | File Formats
to assign a format for that type first.
Use the "Save as type" drop down list to select the destination format for the conversion, then use the Attributes drop down list to select the specific attributes to use for the destination type. If a save type supports customized attributes, then the Attributes label will become a link you can choose to display a configuration window.
If the attributes allow any sampling rate to be used, you can specify the destination rate to use by checking the Rate box and entering the rate in the box. Some attributes have a fixed rate, so a separate rate cannot be specified for those. If no rate conversion is needed, make sure the Rate box is not checked. In that case, a processed file will have the same rate as the original file.
Process Tab
If you want to apply effects or edits to a group of files, use this tab
to add a set of effects, edits, or chains to the list. If no effect
processing is required, remove all effects by using the Remove All
button on this tab. To remove a single effect from the list, select
the effect and use the Remove button. To change the order of processing,
drag-and-drop items within the list.
To add an effect, use the Add Effect button. It presents a tree list of all effects available and their presets. Select the preset you want to use. Chains can be added by using the Add Chain button. Effects and chains are performed on the entire file by default. To perform them on part of the file only, add an edit to set the selection first (see the fade in/out example below).
To add an edit, use the Add Edit button. The "Set Marker/Selection" edit command specifies what part of the file (the selection) will be use for all subsequent effects and edits during processing.
To change settings for an edit or effect, remove it, add a new one with the correct settings, then drag-and-drop it to the correct position in the process list (if necessary). If you require effect settings that are not available in any of the current presets, open a file and use the Effect menu or the Effect Chain Editor tool to create a new preset or chain with the settings you require prior to using the Batch Processing command.
Process Example
You have hundreds of songs and want to create a 10 second sample
file for each song, with the beginning faded in, the end faded out,
and the volume maximized. To set up that processing requires adding
eight edits and effects, as explained below.
To use clipboard related edit commands, it may be necessary to open a file and copy it prior to using Batch Processing. For example, to paste or mix an announcement at the beginning of many files, first open the file containing the announcement, use the Edit | Copy command, then use the Batch Processing command and add a "Paste" or "Mix" edit command to the Process list.
Folder Tab
If you want all processed files to be stored in the same folder
where they currently reside, select "Store all files in their
original folder".
If you want all processed files to be stored in a specific folder, select "Store all files in this folder" and specify a folder in the box provided. You can click on the folder button to browse for a folder.
When adding an entire folder with subfolders by using the Add Folder button, use the "Preserve subfolder structure" box to ensure that the relative subfolders are maintained when storing processed files in a different destination folder. If this box is not checked, all files will be stored in the destination folder and no subfolders will be created. Folders listed in the file list will have double backslashes. The part before the double backslashes will be replaced by the destination folder. For example, if the destination folder is C:\Folder1\ and the added folder (with subfolders included) is C:\Source1\Source2\, then the file list may look something like this:
C:\Source1\Source2\*.* C:\Source1\Source2\\Source3\*.* C:\Source1\Source2\\Source4\*.* C:\Source1\Source2\\Source4\Source5\*.*
Note the double backslashes. The destination folders will be:
C:\Folder1\*.* C:\Folder1\Source3\*.* C:\Folder1\Source4\*.* C:\Folder1\Source4\Source5\*.*
Taking the last item, the folder is divided into three parts with the following colour coding: the
original source root folders, the subfolders, and the pattern.
C:\Source1\Source2\\Source4\Source5\*.*
The destination folder is:
C:\Folder1\
The original source folders are replaced by the destination folder to give:
C:\Folder1\Source4\Source5\*.*
Using the Add Folder button more than once when
preserving subfolders is not recommended unless you fully understand how
the source folder, subfolders, and destination folder are manipulated.
To overwrite any files having the same name and folder as the processed file, check the "Overwrite existing files" box. GoldWave fully processes original files before overwriting them.
Information Tab
Use this tab to control how information is processed. See
File | Information. Select
"Retain information currently
in each file" if you do not want the file's information to be
changed. Select "Replace information in all files" and use the
Set Info button to provide all the information.
Any blank entries are removed from the file.
Select "Replace specified information only in all files" to
replace some of the information. Use the Set Info button
to enter the information
to replace. Any blank entries are not removed or changed in the file.
If the Track number
is set to ##, GoldWave replaces it with a sequential number based
on the order in which the files are processed. The first processed file
will have a track number of 01, then the next file will have 02, etc.
Depeding on the settings used, all processed files may have exactly
the same information (except for ## track numbering mentioned above),
so care must be taken when specifying file specific information such
as Title. Also note that not all file types can store information.
When retaining information and converting to a different file type, some information
from the old file type may not be valid for the new file type. GoldWave does not verify
any information, so care must be taken when retaining information while converting.
"C:\Program File\GoldWave\GoldWave.exe" /process[:preset] <filespec> [<filespec> ...]If no preset is specified, then the default (previous) settings are used. The filespec can specify a single file or contain wildcard characters (* and ?). Quotes are required if the filespec contains any spaces.
To convert all Wave files to MP3 on the command line, for example, you'd first use the Batch Processing command to create a preset with conversion set to the "MPEG Audio" file type and save that preset as "MP3". On the command line, enter the following parameters:
/process:MP3 "C:\My Music\*.wav"
If the preset name contains spaces, use quotes around the entire parameter:
"/process:Trim and convert" "C:\My Music\*.wav"During processing the progress windows is displayed in minimized form, usually just above the Windows Start button. Double-click on it to open it to monitor processing. If no errors occur, the window will disappear and GoldWave will close automatically. Otherwise the progress window opens so that errors can be viewed. To use a log file instead, so that GoldWave always closes regardless of errors, use the /proclog command line parameter to specify the name of the log file:
/proclog:<filename>Quotes are needed around the entire parameter if the filename contains spaces. An example would be:
"/proclog:C:\Log Files\Batch.log"All messages and errors are written to the log file. If you do not require a log file, but still need GoldWave to always close, use nul as the filename:
/proclog:nul
Exit closes all Sound windows then closes GoldWave. Any playback or recording is stopped. You are asked to save any changed files.
A list of several recently used files is appended to the File menu. You can quickly reopen one of these files by selecting it from the menu.
Edit commands remove, insert, copy, and move sections of sound. For an introduction to the concepts and terms used in this section, refer to the Editing Overview section.
The Undo command reverses the most recent change made to a sound. The undo feature keeps a copy of the original data in a temporary file. The temporary folder and the number of undo levels are determined by the Options | Storage settings. To disable Undo, set the number of undo levels to zero.
The Redo command reverses the most recent undo. It restores the the last undone command without any processing. The is helpful if you accidentally undo a change (such as a recording) and want to recover the change or if you want to do a quick "before and after" comparison. Redo is possible only immediately after undo. If you make any other changes to the sound, Redo cannot be used again until you Undo one or more of those changes. Many levels of Undo and Redo are possible. In other words, you can undo several changes and redo them all.
Use Cut to remove the selection from the sound and put it in the clipboard. The contents of the clipboard can then be superimposed or inserted into a Sound window using Mix or Paste. If you just want to remove the selection and do not need to paste or mix it, you should use the Delete command instead.
Note that if only one channel is selected in a stereo sound, then only that channel is removed. Since it is not possible for one channel to be longer than the other, the end of the cut channel is padded with silence (this is also true for Delete).
To cut:
The Copy command copies the selection into the clipboard. The selection is not removed from the sound. The contents of the clipboard can then be mixed or placed into another sound using Mix, Paste, Paste New or Replace.
To copy:
You can copy individual channels of a stereo sound by using the
Edit | Channel menu to select a single channel.
The Copy To command copies the selection to a new file. Use this command to divide a large file into smaller sections or save a piece of a file. The selection is not removed from the sound. The Save As window appears where you can specify the filename, type, and attributes for the file. This is the same as the File | Save Selection As command. To automatically split a large file into several smaller pieces, use the Cue Points tool.
To copy the selection to the file "section.wav":
You can save individual channels of a stereo sound by using the
Edit | Channel menu.
The Paste New command creates a new Sound window containing the sound copied into the clipboard. The new sound will have the attributes and length of the clipboard sound. This command is useful when you need to alter or save the clipboard audio.
To paste part of a sound into a new sound:
After copying a sound into the clipboard, you can use these commands to insert it into another sound. The Paste command inserts the clipboard at the start marker's position. Paste At inserts the clipboard at the location you specify: at the beginning of the file, at the finish marker, or at the end of the file. The length of the sound is increased so that the clipboard will fit. The clipboard is automatically converted to match the attributes of the sound.
To insert the clipboard into the sound:
To append the clipboard to the end of the sound:
Copying a small selection and pasting it several times creates a stutter
effect.
Use Mix to blend (combine) the clipboard with the sound. Note that before you can use Mix, you need to use the Copy command to copy audio into the clipboard. Mixing essentially allows two sounds to be played at the same time, such as vocals and music or voiceovers. You are prompted for the mix volume applied to the clipboard and the start time. A volume of 0dB is full volume. Lower values make the clipboard sound quieter. Adjust the start time to synchronize the clipboard audio with the Sound window audio you are mixing with.
To mix the clipboard with the sound:
Use Crossfade to fade and mix the ends of two sounds, such as fading out one song while fading in another. Note that before you can use Crossfade, you need to use the Edit | Copy command to copy the entire song into the clipboard. See the Crossfading overview for step-by-step instructions.
The Duration specifies how long the transition between the clipboard audio and file audio lasts. The Clipboard position setting specifies which end of the file to fade the clipboard audio. If End of file is selected, then the clipboard is pasted and crossfaded at the end of the file. If Beginning of file is selected, then the clipboard is pasted and crossfaded at the beginning of the file. The Fade curve settings control how the audio is faded. If the song being faded out already has a fade out at the end, use the None fade out curve setting. The fade is shown graphically and can be previewed. Note that any changes made while previewing will not take effect until preview is restarted.
The selection is used when
performing a crossfade. Use the
Edit | Select All command before
using Crossfade unless you intend to crossfade within a certain
part of the sound.
Use Replace to replace the selection with the clipboard. The selection is deleted and the clipboard is inserted in its place. If the clipboard is longer or shorter than the selection, the length of the file is adjusted as required. To avoid changing the length of the file or altering the timing of the sound following the selection, use Overwrite instead.
To replace part of a sound with the clipboard:
Use Overwrite to overwrite part of the sound with the clipboard beginning at the start marker's position. The amount of sound overwritten depends on the length of the clipboard. The length of the file is not changed (unless the clipboard would go beyond the length of the file) and no shifting is performed. Note that if the clipboard is longer than the current selection, then some sound outside the selection will be overwritten as well. The finish marker will be placed at the end of the overwritten sound.
Use Overwrite instead of Replace when you need to preserve tempo, timing, or alignment of the sound following the selection.
To overwrite part of a sound with the clipboard:
The Delete command removes the selection from the sound. The selection is not copied to the clipboard. You should always use Delete instead of Cut when the selection is not needed. The Delete command is faster because it does not copy the selection to the clipboard.
Note that if only one channel is selected in a stereo sound, then only that channel is removed. Since it is not possible for one channel to be longer than the other, the end of the deleted channel is padded with silence (this is also true for Cut).
To delete:
The Trim command removes everything outside the selection. The selection is not affected. Use this command to keep part of a sound and discard everything else. This command is frequently used after recording to trim any leading or trailing silence. Note that if only one channel of a stereo sound is trimmed, the end of that channel is padded with silence. As an alternative, you can use the Copy To command to save the selection to a separate file or use the AutoTrim command to automatically trim silences.
To trim:
The AutoTrim command removes leading and trailing silences from the ends of the selection. Unlike the Trim command, this command works only on the selection and does not remove silences outside the selection.
Silence to keep controls how much silence to leave on the ends. Use zero to remove all silence. Extra silence is not added if the existing silence is less than this amount. The Threshold specifies the minimum level of audio that should not be considered silence. If the audio has a lot of background noise or hiss, use values above -30dB. If the audio is clean, values below -40dB.
Leading silence, trailing silence, or both will be removed depending on the Trim option you select.
Note that if only the left or right channel is selected in a stereo sound, then only that channel is trimmed. Since it is not possible for one channel to be longer than the other, the end of the trimmed channel is padded with silence, which results in trailing silence.
To trim silences from the ends of the entire file:
The Mute command replaces the selection with silence. Unlike Delete or Cut, the length of the sound is not changed. Use this to remove offensive language from music without interrupting the overall beat.
The Insert Silence command inserts some blank space in the sound at the start marker's position. Use this command to make room for recording or to insert a delay. You are prompted to specify the duration of the silence.
To insert 1 minute of silence:
Use Select View to select all of the sound currently shown in the Sound window's graph. The start and finish markers are moved to the far left and far right of the view.
Use Select All to select the entire sound. The start and finish markers are moved to the beginning and end of the sound.
The Channel submenu sets which channel of a stereo sound to use or modify for editing or effects. Use this feature to copy a single channel from a stereo sound or apply an effect to only one channel. The currently selected channel is shown in the status bar. When recording or using effects such as Resample, Playback Rate, or stereo effects, the channel setting does not apply and both the left and right channels are modified.
You can right-click on the channel item in the status bar to quickly select a channel or use Options | Tool Bar to add a "Channel" toggle button to the tool bar.
The Marker submenu lists commands for changing the positions of the start and finish markers for adjusting the selection.
Sets the start and finish markers to an exact time or sample position. To specify a time, choose the Time option and enter the time in hours, minutes, seconds, and thousandths of a second. For example, you could enter 1:04:27.873. To specify a sample position, choose the Sample option and enter the position.
If you want the length of the selection to be aligned to a CD sector or 1 kilobyte, select the appropriate option. When the OK button is pressed, the finish marker is adjusted to align the selection length.
Moves the start and finish markers to their previous positions. The last five positions are stored automatically whenever the markers are moved. Using this command repeatedly sets the selection back to each of those five positions. Note that absolute positions are stored, so any modifications, such as deleting part of the sound will not be factored into the previous positions. In such situations the previous positions may not select the same part of the sound that was previously selected (that part may be deleted).
When playing or recording a file, you can drop the start or finish marker at the current position. You can use the bracket keys, [ and ] to drop the start and finish markers respectively. Note that the start marker cannot be dropped after the finish marker.
Moves the start and finish markers to previously stored positions. These positions are set using the Store Selection Positions command. Hold the Ctrl key to recall the positions stored within a different Sound window.
Saves the current positions of the start and finish markers. Use the Recall Selection Positions command to move the markers back to these positions. Hold the Ctrl key to store the positions so they can be recalled in a different Sound window.
When editing, it is important that the waveform not change suddenly from one amplitude to the next, otherwise a click will occur. This can happen when deleting the selection. The amplitude of the waveform at the start marker may be completely different from the amplitude at finish marker. After deleting the selection, these two different amplitude will be right next to each other, causing a click.
The "Snap To Zero-Crossing" feature helps to minimize the problem by making sure that the markers are always near zero amplitude samples. When you drag and release a marker, it is automatically moved to a position where the amplitude approaches zero. This means that when you delete the selection, the amplitudes at both the start and finish markers will be more closely matched (near zero).
Since stereo sounds can have very different left and right channels, it is not always possible to find an ideal zero-crossing position. However, you can use the Edit | Channel menu to limit the snap feature to a single channel.
Note that if you zoom in close enough so that the true shape of the waveform is shown (such as View | Zoom 1:1), the snap feature is automatically turned off so that markers can be placed at any position.
This submenu lists commands for working with cue points.
Displays the Cue Points tool.
Drops a cue point at the current playback, recording, or start marker position. The name of the cue point is set to the time position by default, but that can be changed by the Auto Cue settings under the Cue Points tool.
Moves the start marker to the next cue point or to the end of the file.
Moves the start marker to the previous cue point or to the beginning of the file.
Divides the file into smaller sections using cue points at split points. See Split File Button under the Cue Points tool for more information.
Displays vertical cue lines on the Sound window graph. This setting applies to each Sound window individually, so lines can be shown in one window and not in another. When a new window is opened, the last setting selected is used.
This section explains commands under the GoldWave's Effect menu. Please see the Effects Overview section for general information about how GoldWave employs effects.
A Doppler effect is defined as a change in frequency of a wave caused by motion. You often hear it when police or ambulance sirens drop in pitch as they pass near by.
In GoldWave the Doppler effect dynamically alters or bends the pitch of the selection. It does this by altering the speed at which the waveform is played. Note that both pitch and tempo are changed. Shape Controls are presented where the pitch/speed can be varied over the selection from 0.25 to 2.0 times normal. You can use Effects | Volume | Shape Volume to dynamically alter the volume as well.
The "Slow down" preset gives you a good idea of what it sounds like when the batteries start to fail in a portable tape player. Other presets can change your voice to a smurf ("Faster") or a giant ("Slower").
Dynamics alters the amplitude mapping of the selection. It can limit, compress, or expand a range of amplitudes. The amplitude mapping is set using Shape Controls, where x-axis and y-axis both have a range of -1 to 1. When the line stretches diagonally from the lower left corner to the upper right corner, the input amplitude (x) and output amplitude (y) are the same for every point on the line. By changing the line, the output will differ from the input.
The figure shows an example of amplitude mapping for clipping distortion. Point P1 has an input value of -0.4 and an output value of -0.4. Therefore no change occurs to the amplitude. Point P2 on the other hand, has an input value of 0.8 and an output value of 0.5. In this example, all input amplitudes in the range of -0.5 to 0.5 remain unchanged. Any values outside this range will be limited to ±0.5, so that the final sound will have no amplitude magnitudes greater than 0.5. Essentially, any values that are too high are "clipped" to fit within the range.
In practical terms, dynamics can increase the volume of quiet sections of a sound without greatly increasing the loud sections as well. It can introduce mild or heavy distortion effects (such as the "Blare" or "Hiss noise" preset).
See Also
Compressor/Expander
Echo produces an echo or reverb effect in the selection. The parameters include number of echoes, echo delay, echo volume, feedback, stereo, and tail generation.
Echoes specifies the number of times the sound is repeated and mixed at a diminished volume with the given delay.
The Delay is the amount of time it takes for the first echo to be heard. If more than one echo is used, then each subsequent echo delay is compounded by this amount again. In other words, the second echo is delayed twice as long, the third echo is delayed three times as long, and so on.
The Volume controls how loud the first echo is. The volume of subsequent echoes is compounded in the same way the delay is.
Feedback makes the echo sound deeper and richer. It regenerates each echo, and the echo of those echoes, and so on, creating many more echoes than before.
The Stereo option makes the echo bounce between the left and right channel of a stereo sound.
The Generate tail option adds some silence to the end of the selection so that the trailing, fading echoes can be stored. This increases the length of the sound. Turn this setting off if you do not want to change the length of the selection or have any silence inserted. If the option is off, echoes will end abruptly rather than trailing off gradually.
The Compressor/Expander effect dynamically alters the audio volume level. It is commonly used as a compressor, limiter, expander, or gate.
This effect does not change the size of the file. For
file compression,
use File | Save As and
change the type and attributes.
Compressors and limiters are used to decrease or limit the dynamic range of audio. In plain English, they reduce the volume of loud sounds while leaving the rest of the sound unchanged. Expanders and gates are used to increase the dynamic range of audio. They reduce or eliminate quiet sections while leaving louder sections unchanged, which can help to reduce background noise.
Compressors always work on loud sections and expanders always work on quiet sections. Normally, both compressors and expanders only reduce the volume. However, GoldWave also allows you to boost the volume.
The Multiplier specifies the amount of volume change. It is the scale factor applied to the sound when the threshold is met. For compressors, it is the amount to scale the loud sections. For expanders, it is the amount to scale the quiet sections. Normally a value less than 0dB should be used to reduce the volume.
The Threshold specifies the audio level to activate the expander or compressor. Compressors scale the volume level of all sounds above that level. Expanders scale the volume level of all sounds below that level.
The Attack and Release times specify how quickly the expander/compressor is activated. An attack value of 0.100 means that the audio level will have to cross the threshold for at least one tenth of a second before the multiplier is used at full strength. A release value of 0.100 means the multiplier will continue to be used for one tenth of a second after the audio level no longer crosses the threshold.
Use the Expander or Compressor mode to specify what processing is required.
The Anticipate attack setting tells the effect to scan ahead for audio that crosses the threshold. If the Attack time is set to 0.100, then the effect scans ahead by 0.100 seconds. This means that the multiplier will be at full force the instant the threshold is crossed rather than building to full force 0.100 seconds later. The Use smoother setting smooths out any sudden volume changes which may occur during processing with small attack/release times.
Compressor Example
You have recorded some music that has a few loud moments, but you want
to raise the overall volume without distorting the loud
parts. Select the "Reduce loud parts" preset.
After compression, use the
Maximize Volume
effect to stretch the volume to the full dynamic range.
Alternatively you could use the "Boost quiet parts" preset.
Expander Example
You have recorded someone talking and notice background noise
during the quiet parts. To reduce the noise, select the
"Noise gate" preset or the "Reduce quite parts" preset.
See Also
Dynamics
Filters are used to remove a range of frequencies from a sound and can produce a variety of effects. The submenu contains several filter related effects.
Bandpass filters block all frequencies outside a specified range, keeping only frequencies within the range. Bandstop filters block all frequencies within a specified range, keeping all other frequencies outside the range.
Frequency Range
The From and To boxes
specify the initial frequency range of the
filter. If the Dynamic option is selected, then a final
frequency range can be given in the other From and To
boxes, allowing you to fade from on frequency range to another.
Otherwise the initial range remains constant for all processing.
Settings
Select Bandpass if you want to keep only the frequencies
within the range. Select Bandstop if you want to block
all the frequencies in the range.
The remaining Static, Dynamic, and Steepness options are explained under
Low/Highpass filter.
Example
To remove hum and hiss from a voice recording:
Equalizers are commonly found on stereo systems. They boost or reduces certain ranges of frequencies. Simple equalizers control only treble and bass. GoldWave's equalizer controls 7 bands, as shown in the figure.
Center frequencies for each of the 7-bands are given at the top of each fader. Adjust the faders to boost or reduce a band by +12dB to -24dB.
To change bass, adjust the two or three left-most bands. To change treble, adjust the two or three right-most bands. Several presets are included to demonstrate bass, mid, and treble changes.
More detailed equalization is possible using the Parametric EQ, described below.
Lowpass filters block high pitched, frequencies (treble) but allow low pitched frequencies (bass) to pass. They can be used to reduce high end hiss noise or remove unwanted sounds above the given cutoff frequency. If you were to apply a lowpass filter with a cutoff frequency of 1000Hz on speech, it would make it sound mumbled and deep. Lowpass filters can also be used to eliminate aliasing noise when used before downsampling.
Highpass filters block low pitch frequencies, but allow high pitched frequencies to pass. They can remove deep rumbling hum or remove unwanted sounds below the given cutoff frequency. If you were to apply a highpass filter with a cutoff frequency of 1000Hz on speech, it would make it sound thin and hollow.
Cutoff Frequency
The Initial cutoff specifies the constant cutoff frequency for
static filtering. If the Dynamic option is selected (see
below), then a final cutoff frequency can be given in the
Final cutoff box, allowing you to fade from one cutoff to another.
Filter Options
Select Lowpass if you want to keep only the frequencies
below the cutoff frequency. Select Highpass if you want
to keep only the frequencies above the cutoff frequency.
If you want the cutoff frequency to remain constant throughout the selection during processing, select the Static option. If you want the cutoff frequency to change from the initial value to the final value, select the Dynamic option. Note that dynamic filtering will take more processing time.
The Steepness value specifies how sharply the filter cuts off frequencies outside the cutoff frequency. A higher steepness makes the filter sharper, but it also increases processing time. In technical terms, the steepness specifies the number of second order cascade filters used.
Examples
To make speech gradually become more hollow and thin:
Remove all high end hiss noise from a voice recording.
Noise Reduction uses frequency analysis techniques to remove unwanted noise, such as a background hiss, a power hum, or any continuous, consistent sound. It cannot be used to separate or remove complex or brief sounds, such as coughs, laughter and applause. It cannot remove instruments or vocals from music.
The interface, shown in the figure below, includes a frequency analysis window, with Shape and other controls. The x coordinate is the frequency in Hertz and the y coordinate is the magnitude in decibels.
The frequency analysis provides graphical information about all frequencies within the sound at the given time. For stereo sounds, the left channel is shown in green and the right channel is shown in red. The time of the frequency analysis within the sound is given in the Time box. Move the scroll bar located below the analysis window to change the analysis time to show the frequency analysis of a different part of the sound. The detail of the analysis depends on the FFT size setting, explained below.
Reduction Envelope
Noise is removed using a reduction envelope. The shape of the
envelope should closely match the shape of the noise you want to
remove. The frequency analysis graph can help determine that
shape. Change the analysis time so that it coincides with a time in
the sound where only the noise is heard (use the
preview button
to play the file to find such a place and time). Once you have
isolated the noise in the analysis
graph, you can then create the envelope. The envelope can be created
in four different ways, depending on the reduction envelope
setting. The "Use clipboard" option generally gives the best results.
Also try the presets.
Use shape
Lets you manually create an envelope shape or select a
preset shape. See Shape Controls
for information about creating shapes. By creating a horizontal
line at about 80dB, you can remove a hiss from a sound. In
some cases, you'll need to trace the outline of the graph
or draw completely different shapes to reduce the unwanted
frequencies levels.
Use current spectrum
Creates an envelope based on the shape of the graph shown
in the frequency analysis window. This is particularly
useful for removing a complex buzz or hum. It is important
that the analysis Time is set to a place where the noise is
heard by itself initially.
Use average
Applies an averaging envelope throughout noise reduction
processing. The envelope is continuously updated, based
on the average frequency analysis of the sound. Use this setting if
the noise changes frequently throughout the sound.
Use clipboard
Creates an envelope based on an analysis of the waveform in the
clipboard. This is the most flexible option and usually gives the
best results. Before you can use this
option, you must use Edit | Copy
to copy a piece of noise
into the clipboard. For best results, the piece should contain only
the noise you want to remove from the rest of the file. The noise
can even be copied from a different file. After you copy
the noise, remember to change the selection
to the part of the file you want to apply the noise reduction.
Settings
The FFT size determines the detail of the frequency analysis and the
noise reduction envelope. Usually values of 11 or higher give the
best results. The Overlap value specifies the percentage of the FFT
size to overlap from one calculation to the next. A value of 88 works
well. The Scale value lets you alter the reduction envelope scale.
A value of 100 uses the envelope as it is. A value of 200 doubles
the envelope, which doubles the amount audio removed from the
sound. A value of 50 halves the envelope, which halves the amount
removed. Normally it should be set to 100.
A parametric equalizer (shown below) is a flexible tool for reducing or enhancing ranges of frequencies. GoldWave presents an easy to use interface where all the parameters for up to 40 bands can be configured quickly. The presets contain some commonly used templates.
Graph window
The graph shows frequency on the x-axis in Hertz and the
gain on the y-axis in decibels. Each enabled band is
displayed in the graph as a diamond shaped box located at
its center frequency and gain. The width of the box shows
the bandwidth. The currently selected band is shown in
blue and its exact settings are given in edit box controls.
A short time frequency analysis graph is drawn with the left channel in green and the right channel in red. The time of the analysis can be changed using the scroll bar located at the bottom of the graph. The analysis helps determine what frequencies to boost or reduce. A high pitched squeal, for example, would appear as a spike near the right side of the graph. Whereas a low pitched hum would be a spike or bump near the left side.
Controls
A band is configured by selecting its number from the
Band box and adjusting the Gain, Width, and Center faders. A quicker
method is to drag-and-drop the band to a new location on the
graph. Note that because of the logarithmic frequency
scale, the width of a diamond changes as you move it left
or right. The bandwidth, however, remains constant.
Use the diamond plus button to add more bands. Use the diamond minus button remove existing bands. The current (blue) band given in the Band box is the one removed.
The "Notch" preset is effective for removing a simple tone from a sound, such as a 60Hz hum or telephone dial tones. The "Bass boost" and "Treble boost" presets work the same way as the bass and treble controls on a stereo system. Adjust the gain up or down to control them.
A pop/click filter is a specially designed filter that searches for abrupt changes in the sound and eliminates them. Such a filter is often used to remove pops and clicks caused by dust and scratches when recording old vinyl records.
The Tolerance defines how abrupt a change can be before it is considered a click. It is best to start with a value near 1000%. Using a lower value will detect more clicks, but may eliminate natural clicks such as drum sticks tapping together or a conductor tapping the baton. Values less than 500% should be used on short selections only.
When a click is detected, the filter attempts to reconstruct the damaged waveform based on the surrounding waveform shape making the repair almost imperceptible. However with excessive pops and clicks or at low tolerance levels, reconstructed waveforms may overlap and sound distorted. The tolerance setting should be kept as high as possible. Using a very low setting may introduce more distortion than existed in the original. This is most noticeable in voice recordings or instrument solos, particularly trumpet solos.
The filter requires a minimum selection of 4000 samples (about one tenth of a second at CD quality) to establish a base line. Using the filter on a shorter selection has no effect.
This filter automatically removes silences from a sound. Use it to save storage space or to remove long pauses in speeches or police/airport radio recordings.
The Silence threshold specifies the volume level for the silence. Any audio below this level is considered silence and is subject to removal, provided it has a long enough duration. The Duration specifies how long the silence must be before it is reduced. Any silences short than this remain unchanged.
Reduce to is a percentage that specifies the length of the reduced silence relative to its original length. A value of 75, for example, reduces a 10 second silence to 7.5 seconds.
Maximum length specifies the maximum length for reduced silences. This setting overrides the Reduce to setting if the reduced length still exceeds the maximum. If this is set to 5 seconds in the above example, then the 10 second silence is reduced to 5 seconds.
Use this filter to reduce hiss and crackle. Length sets the length of the smoother filter. The larger the value, the more averaging is applied to the audio and the duller it will sound. Use the Volume setting to help offset the loss in volume for larger Length values.
A spectrum filter is a general purpose audio filter similar to a parametric equalizer, but with much greater control. Instead of using individual bands, the entire frequency spectrum is controlled using a shape line that controls the gain. This allows many kinds of filters to be designed, such as lowpass, highpass, bandpass, bandstop, notch, peak, comb, and more. Filtering is performed in the frequency domain using Fast Fourier Transforms (FFTs).
A spectral analysis window is displayed with a shape line and several other controls. The X and Y coordinates are updated when you click-and-drag a shape point. The X coordinate is the frequency in Hertz and the Y coordinate is the magnitude in decibels. The time of the spectral analysis shown is given in the Time box. If you move the time scroll bar, located below the analysis window, the graph changes to show the spectral analysis of a different part of the sound. The Master gain controls the overall gain of the filter, which is equivalent to shifting the entire shape up or down.
Initially the shape line is horizantal at 0dB, which means that no changes in gain are made at any frequency. Alter the shape line up or down to increase or decrease the gain at a particular frequency. In technical terms, the shape line represents the frequency response function.
Settings
The FFT size
determines the detail of the spectral analysis and the resolution of the filter.
Higher settings provide a higher resolution, allowing the filter to follow
the shape more accurately, with sharper cutoffs. When processing high sampling
rate files, such as 88kHz or 192kHz, the FFT size must be set higher
for the filter to follow the shape. Using too high a value may cause overshoot
and oscillations in the gain.
The Overlap value specifies
the percentage of the FFT size to overlap from one
calculation to the next. The lowest value gives the fastest processing and
generally works well.
A flanger effect is similar to an echo effect in that the original sound is mixed with a delayed copy of itself. Unlike an echo, where the delay is constant, a flanger varies the delay over a given range. The speed, or frequency, at which the delay varies is controlled as well. The Flanger effect presents a window where you can set the depth, frequency, and fixed delay parameters and control how the sound is mixed. Many presets are included to demonstrate the kinds of unusual audio effects that are possible.
Volumes
The Source volume
specifies the volume of the original sound to mix
with the final sound. A value of 0 means the original sound will
not be mixed at all with the final sound. If this value is set to 100 and all
other volumes are 0, no change is made to the sound. A value
of -100 inverts the source, which is equivalent to subtracting
the original instead of adding it to the final sound.
The Flanger volume specifies the volume of the delayed sound to mix with the final sound. Usually, this value should be in the range of 25 to 100, or -25 to -100 for an inverted mix.
Feedback specifies the level of feedback (previously generated output) to mix with the final sound. This makes the effect more full and pronounced. Set this value to 0 if you do not want any feedback. In general the feedback should be set to between -75 to 75.
Flanger Settings
Variable delay specifies the maximum variable delay in milliseconds. A
value of 40 will allow the delay to vary from 0 to 40 milliseconds.
Frequency specifies how fast to vary the delay. A value of 2 will vary the delay over from 0 to its maximum twice a second. For a value of 0.2, the maximum delay is reached every five seconds.
The Fixed delay is added to the depth to change the minimum delay. If the variable delay is 40 and the fixed delay is 10, the delay will vary from 10 to 50 milliseconds.
The Sine modulator and Triangular modulator settings control how the delay is varied. If sine is selected, then the delay varies on a sinusoidal pattern. If triangular is selected, then the delay varies in a simple linear pattern.
The Stereo option allows the flanger and feedback audio to be bounced between the left and right channels of a stereo sound, giving a more pronounced stereo effect.
Interpolate (see figure below) uses linear interpolation to smooth out the waveform within the selection. Use this command on a tiny selection to remove a pop or click. This command should not be used on a large selection.
Invert reflects the selection about the time (horizontal) axis. The selection is essentially turned upside-down. This produces no noticeable effect in mono sounds and has a slight effect in stereo sounds. Inverting a single channel of a stereo sound produces an "in" or "out" effect.
Inverting can be used before mixing so that the two sounds are subtracted instead of added.
Mechanize gives a robotic or mechanical characteristic to a sound through a method known as amplitude modulation. This effect was widely used in old science fiction movies. The rate of modulation is controlled with the Frequency value. The sound is modulated with one of several waveforms given in the table.
| Waveform | Purpose |
| Sine | The sound is modulated with a sinusoid, which tends to shift the pitch and cause distortion. Using a very small frequency value (less than 2.0) causes the sound to fade in and out. |
| Triangle | The sound is modulated with a triangular wave. This is similar to the Sine modulator but causes more distortion at higher frequencies. |
| Square | The sound is modulated with a raised square wave, with amplitudes ranging from 0.0 to 1.0. This causes very heavy distortion at high frequencies and intermittent (on/off) audio at low frequencies. |
| Clipboard | The sound is modulated with audio stored in the clipboard. Audio must be copied to the clipboard before using this option. No attempt is made to convert the clipboard audio to a compatible sampling rate. |
The Offset effect adjusts or removes a dc offset in the selection by shifting it up or down so that the wave is centered on the horizontal axis.
Positive values shift the graph up and a negative values shift it down. Use the Scan Offset button to automatically determine what offset to use to remove any existing offset. After scanning completes, the Left and Right values are set such that they will cancel out the offset in the given channel. If the values are zero, then no offset was detected.
Any offset should be removed to minimize pops/clicks during editing. Offsets may interfere with other effects as well.
You should check the offset from time to time after processing
effects. Otherwise, the offset may increase with each effect,
resulting in some distortion.
Pitch changes the pitch (frequency) of the selection. This is useful for converting instrument samples from one note to another. The new pitch is specified using a scale factor or using semitone and fine tune values.
Scale
This option scales the pitch by a percentage value. If you set the
scale to 50, that is equivalent to a downward shift by one
octave. A value of 200 is the same as an upward shift of one octave
and would make a voice sound like a chipmunk. A value of 75
would make a woman's voice sound like a man's.
Semitone
This option changes the pitch by semitones (notes on a piano). If
your sound is a note at middle C and the semitone value is 2, the
note is changed to D. A value of -1 changes the note to B. A
value of 12 make the note one octave above middle C. The Fine
tune value lets you make a slight pitch adjustment in hundredths of
a semitone. For example, a value of 50 changes a note from C to
halfway between C and C#.
Preserve length
If this option is checked, a complex algorithm is used to keep
the length of the original note the same as the new note. In other
words, the tempo will not be changed. In terms of a voice
recording, this changes the pitch of the voice without changing the
speed at which the words are spoken. This option requires a
substantial amount of processing time and will affect the quality of
the sound.
The FFT size determines how much of the sound to process at one time, based on a power of 2. A value of 10, for example means that 1024 (210) samples are processed at a time. Values from 11 and up give good results. The Overlap determines what percentage of the processed samples are recalculated. It should be at least 88. Values of 90 and 95 will give better results, but require more processing time.
This menu lists all plug-ins detected by GoldWave. If any compatible DirectX plug-ins are detected, then they are listed under a DirectX submenu.
This command reverses the selection so that it plays backward. Now you have an easy way to capture all those "satanic verses" or reverse speech messages. You can play a sound backwards by using the rewind button on the Control window as well.
Reverb adds body and richness to the sound by simulating the acoustic reverberation and echoes within a chamber or room.
The Reverb time sets the size of the reverb. A longer time simulates a larger chamber or room.
The Volume controls how loud the reverb is. Values less than -18dB give good results.
The Delay scale alters the delay of the reverb for fine tuning. Use 1.0 for a standard reverb.
The Stereo submenu contains commands that apply to stereo files, such as swapping channels and left/right panning.
This command swaps, mixes, inverts, or combines the left and right channels in a variety of ways. The left and right channels are replaced with a mixed combination of both channels, depending on the volume levels set. Use the controls in the Left channel box to set the mixing volume percentages for the left channel. Use the controls in the Right channel box to set the mixing volume percentages for the right channel.
To swap the left and right channels:
This command automatically balances the left and right channels and maximizes the volume levels. Essentially, this effect uses the Match Volume effect internally on the left and right channels, then uses the Maximize Volume effect.
After processing, the left and right channels will have the same average volume level and at least one channel will have full dynamic range (1.0 or 0dB). Note that it is rarely possible for channels to have the same average and both have full dynamic range at the same time. Typically one channel will have a dynamic range of slightly less than 1.0 or 0dB.
Pan presents the Shape Controls where left and right panning can be controlled. The graph is divided into green and red regions, representing the left and right channels respectively. The line, initially located between the regions, represents the center for panning. By bending and/or moving the line, you can dynamically alter the selection's left/right balance or pan to and from each channel. The figures below show several examples of panning shapes.
 Figure: Pan from left to right |
 Figure: Pan from right to left and back to right |
 Figure: Pan from left center to right center |
There are two end points on the pan shape line by default. Use the Point box to select the point to edit. The X and Y values control the location of the selected point. X is the time and Y is the amount of panning. A positive Y value pans to the left. A negative value pans to the right. For end points, only Y can be changed. X is fixed at the beginning and ending of the selection.
Example: Select point 1 and set Y to -1.0 to make panning start at full right. Select point 2 and set Y to 1.0 to make panning end at full left. Panning will go from right to left over the selection.
Use the Add Point button to insert a new point between the two end points. Note that the point numbers change, with point 3 as the end point and point 2 as the new one. Both the X and Y values can be changed for that point. To make panning go slightly left at 10 seconds into the file, for example, set X to 10.0 and Y to a positive value, such as 0.25. When adding points, be sure to select the point after which you want to insert the new point. The X value for each point is confined to adjacent points.
The Show balance option calculates and displays the current peak balance in yellow on the graph. For a typicaly stereo song, a spiked line roughly centered around zero would appear. For a 2 channel mono file, it would be a perfectly flat line at zero. For an unbalanced file, the line would be above or below zero.
The Change volume only setting limits the pan effect to volume changes only. Normally panning mixes the left and right channels to alter the balance. This setting prevents any mixing and only changes the relative volumes of the channels.
This effect reduces vocals from certain stereo recording by subtracting the left and right channels and by using a bandstop filter. This works best when vocals are recorded exactly the same on the left and right channels and no stereo effects have been applied to the vocals. Note that any instruments record the same in both channels are reduced as well.
Usually subtracting the left and right channels destroys the stereo sound, giving mono output. However, by integrating a bandstop filter, GoldWave is able to restore some of the stereo, enhancing the output.
Increase the bandstop filter Volume and bring the From and To values closer together for increased stereo (increases the vocals). Decrease the bandstop filter volume and move the From and To values farther apart for reduced stereo (decreases the vocals).
Try the presets to learn what the different settings do.
Time Warp changes the playback speed or alters the tempo of the selection. This effect has many uses: it can stretch or compress a sound to fit in a certain time, it can slow down instrumental music for easy transcription, or it can change the tempo of one musical passage to match rhythm and beats of another. The Change option lets you specify a relative change in percent. A value of 50 makes the selection play twice as slow. A value of 200 makes the selection play twice as fast. The Length option lets you specify a new length for the selection. Use this to make a sound fit a certain time, such as squeezing a 35 second commercial into a 30 second spot.
Three different time altering algorithms are provided, each with certain advantages and disadvantages.
Speed
The Speed algorithm simply changes the sampling rate of the entire
sound so that it plays
back at a different speed, similar to spinning a vinyl record faster
or slower. It works the same way as the speed fader in the
Control window, but in this case, the sound itself is
changed. This technique is very fast and produces excellent quality,
however, the pitch of the sound is changed as well. In other words,
if you were to speed up a voice, the pitch becomes higher, making
the voice sound like a chipmunk.
Similarity
The Similarity algorithm uses correlation to add and overlap small, similar
sections of the sound. This technique preserves the pitch. It
generally produces high quality voice and fair quality music when
using small speed or time changes. A fair amount of time may be
require for processing, depending on the Search range value. For
voice, the Window size should be set between 20 and 30 and the
Search range set to between 5 and 10. For music, a larger Window
size and Search range gives better results, such as 60 and 25.
FFT
The FFT algorithm uses Fourier transforms and interpolates or decimates the
frequency analysis to change the length. This technique preserves
the pitch, but can introduce some artifacts into the sound. Best
quality is obtained by using the Oscillator synthesis option, but that
requires significant processing time. The FFT size should be set
from 11 to 12 and the Overlap should be at least 75, but can be set
to 88, 90, or 95 for better quality. The FFT size and Overlap
settings are explained under the Noise Reduction
filter section.
If you changed the speed fader in the Control window, remember to
set it back to 1.00 so that the device plays at the correct speed.
The Volume submenu contains several volume related commands. Volumes are usually specified in decibels (dB) or by a percentage (%) of the sound's original amplitude. For more information about volumes, see Volume Scales in Appendix A. Unlike the volume fader in the Control window, which changes the device playback volume, these commands scale the sound's data to change the volume.
This command modifies the selection so that it sounds louder or quieter. You need to enter the new relative volume. Values less than 0dB make the selection quieter. Values greater than 0dB make it louder. A value of 0dB leaves the volume unchanged.
If you are trying to make the volumes of several different songs sound the same, use Effect | Volume | Match Volume instead.
Fade In gradually increases the volume throughout the selection. You need to specify the initial volume. Normally to fade in from complete silence, you'd use the lowest possible value of −160dB.
The Logarithmic and Linear options control the shape of the fade. A Logarithmic fade will fade in more rapidly than a Linear fade.
To fade in the first 5 seconds of a sound from silence:
Fade Out gradually decreases the volume throughout the selection. You need to specify the final volume. A value of −160dB fades to complete silence.
The Logarithmic and Linear options control the shape of the fade. A Logarithmic fade will fade out more rapidly than a Linear fade.
To completely fade out the last 5 seconds of a sound:
The Match Volume effect makes the volumes of separate files seem similar. When creating a CD, you often find that each songs is recorded at a different volume level. This means you have to adjust the CD Player volume from one song to the next. By using the Match Volume effect, you can adjust the volume levels of each song so they all have the same average level, eliminating the need to adjust the volume for each song later.
Volume changes are based on a root-mean-square (rms) average. The rms average is calculated with silent regions (below -44dB) excluded. Files with similar average levels will seem to have similar overall volume levels.
Use this effect on each file to set the Average to the same value. Use the File | Batch Processing command to apply this effect to a group of files before burning them to a CD. The Clipping control options are explained in the table below.
If you want to
match the left and right channel levels of a single file, use the
Effect | Stereo | MaxMatch
effect instead.
The average value to use depends on the files. You should open each file and display the Match Volume effect to see what average value it has, then apply an overall average value to all the files. To avoid clipping distortion, it is best to use the minimum average across all files. For example, if one file has an average of -20dB and all the other files have a higher average, such as -18dB, then use -20dB for all files.
Unlike the Maximize Volume volume effect, the Match Volume effect may result in clipping distortion if the average level is set too high. The Final peak area displays the resulting final peak level after processing. If the peak exceeds 0dB, the value is shown in red as a warning that clipping may occur. Use a lower average to avoid clipping.
This effect should not be used with Maximize Volume. Use one or the other, but not both (one cancels the other).
| Option | Purpose |
| Allow clipping | Uses the given average level, regardless if clipping is required. |
| Reduce average level to avoid clipping | Reduces the average level just before processing begins to ensure that the audio will not be clipped. A lower level is used for the entire file, so the file may not sound as loud as other files processed with the same level where clipping was not required. |
| Abort processing if clipping is required | Processing is aborted immediately when clipping is required. A range error is displayed. Use this setting for batch processing to prevent writing clipped audio. |
Maximize Volume searches the selection for the current peak volume level. It then displays the level and the position of the level within the file. You can then specify a new absolute maximum volume level. The volume of the entire selection is changed so that the maximum will match that value. A value of 0dB gives you full dynamic range. This is often referred to as "normalizing" the volume.
Some effects in GoldWave may cause the volume level to go above 0dB. After using many effects, you should use the Maximize Volume effect before saving a file to ensure that the full dynamic range is not exceeded. Otherwise clipping may result in the saved file.
Always use Maximize Volume before saving a file to avoid
clipping and make full use of the dynamic
range.
Shape Volume presents Shape Controls where the volume envelope of the selection can be defined. The shape line is initially horizontal at 1.0, representing normal volume. By bending or moving the line, you can dynamically change the volume over the selection. Adding a point below 1.0 decreases the volume. Adding a point above 1.0, in the red section, increases the volume. Note that increasing the volume may cause clipping distortion. Several preset shapes are included.
The Show envelope option calculates and displays the current volume envelope of the sound. The left channel envelope is shown in green and the right channel envelope is shown in red.
Example
To use this effect to reduce the volume of background music before
mixing in a voice recording or narration over the music:
The Playback Rate effect changes the playback rate of the entire sound. The sound plays faster (or slower) and its pitch is higher (or lower), similar to playing a vinyl record faster or slower. Essentially, this just changes the sampling rate value shown in the status bar.
While the playback rate of the audio device is controlled with the
speed fader in the Control window, this has no effect on the
file's data. You must use the Playback Rate effect for the change
to be savable.
The Resample effect changes the sampling rate of the entire sound. Unlike Playback Rate, this command re-calculates and interpolates all the data so that the pitch and playback time are not affected. Use this command to convert any sampling rate to the standard CD rate of 44100Hz or the standard telephony rate of 8000Hz.
Up-sampling and down-sampling filters are automatically applied as required.
If you have a sound recorded at 44100Hz and do not need CD
quality, you can save large amounts of disk space by resampling the
sound to 22050Hz or 11025Hz. This reduces the size by 2 to 1 or 4 to 1.
Before reading this section, review the terms introduced in the Interface Overview and Editing Overview sections.
View commands allow you to see a more detailed graph of part of the sound. They are similar to zoom commands in the Windows Paint accessory. When you zoom in (or magnify) the sound, you see a smaller section, but with greater detail. When zoomed out, you see the entire sound, but with less detail. The Overview box near the bottom of each Sound window gives you some information about what section of the sound is currently shown in the view (see the Main Window figure).
When zoomed in to a part of the sound, a scroll bar appears at the bottom of the Sound window so you can move to different parts of the sound while still keeping the same level of magnification. The current level of magnification is displayed in the Main window's status bar next to the modified status. The magnification is given as two numbers separated by a colon. The first number is the number of pixels and the second number is the number of samples mapped to those pixels. For example, 1:1 means one sample is mapped to one pixel and for 51:1003, 1003 samples are mapped to 51 pixels.
Most view commands use the start marker's position as the starting location for magnification, so you should move the start marker to the position of interest first.
Use the Options | Window
command to set the initial zoom level when a file is opened.
The entire sound is graphed in the view. In other words, it zooms all the way out so that the entire sound is visible. You can move the start and finish markers to select any part of the sound.
This magnifies the graph to any level you specify. The level can be given as a time length or as a ratio. If the Length option is selected, then the length you specify is shown in the graph. For example, use 1:00 to show one minute of audio. If the Ratio option is selected, then the given number of samples are mapped to a single pixel on the screen. Values greater than 1 display an approximation of the waveform. Values less than 1 (such as 0.1) reveal individual samples and allow direct editing of the waveform.
The Start time specifies what position in the file to begin drawing the zoomed waveform. If the given level is not possible, the closest valid level is used.
The selection is magnified, increasing the detail of the graph. You can zoom in many times by changing the selection and magnifying it again until only a single sample is shown in the view.
Click-and-drag the right mouse button over the waveform to zoom in
without changing the selection.
A Preset button is provided in the tool bar so that you can quickly zoom to your favourite level. The sound is magnified to the level of detail specified under Options | Window. The level can be set to any value you find convenient.
This returns the view to the previous zoom level. Use this to switch back and forward between two different zoom levels.
When this item is checked, the view scrolls automatically to follow playback and recording. Scrolling occurs only when zoomed in and when the current playback or recording position goes outside the view. This feature also enables keyboard navigation, which allows the playback cursor to be moved via the keyboard.
Magnifies the sound by a factor of 1.33x. This gives 33% more detail, but shows 33% less sound. The middle of the view is used as the zoom center. This command complements Zoom Out.
Click-and-drag the right mouse button over the waveform to zoom in
on any part of the sound.
Reduces magnification by a factor of 1.33x. This gives 33% less detail, but shows 33% more sound. The middle of the view is used as the zoom center. This command complements Zoom in.
When the number to the left of the colon is greater than the number to the right, a very small section of sound is magnified at a high level of detail. At these levels, individual samples are easily visible and direct waveform editing with the mouse is possible.
At a level of 1:1, each audio sample is represented as a single pixel on the screen. This reveals a true representation of the shape of the sound.
These show the given amount of time of audio beginning at the start marker's position. You can use View | Finish to see the audio at the end.
Vertically zooms all the way out so that the entire vertical amplitude range of the sound is shown.
Magnifies the graph vertically to show 1.33 times as much amplitude detail. Zooming is centered on the horizontal center of the view.
Reduces vertical magnification to show 1.33 times less amplitude detail. This show a larger range of the amplitude. Zooming is centered on the horizontal center of the view.
These commands scroll the view to either the start or finish marker's position. The view is centered over the marker's position provided its position and the level of magnification permit it to be centered. These commands are especially useful when you need to move a marker to a precise position. For example, you can zoom in 1:1 and move the start marker to an exact position and then use View | Finish to set the finish marker's position.
GoldWave includes several tools for working with audio files. These are described in the following sections.
The CD Reader tool digitally copies audio directly from an audio CD to a file on your hard drive, without using your sound card. This features has several advantages over normal recording:
The CD-ROM drive must be MMC compliant (Multimedia Command Standard). For Windows 95, 98, and ME you must have an ASPI driver installed. For Windows 2000, XP, or later, support is built-in. Due to the wide variety of interfaces and inconsistent device standards, incompatibilities may arise that will require a system reset. It is recommended that you close all other programs before proceeding.
If your system is configured correctly, the window allows you to select a CD device and to specify tracks to copy. If you have only one CD-ROM drive, only one device is listed in the drop down list. If you already have an audio CD in the drive, the track times are shown in the lists. Otherwise you will need to insert an audio CD and re-select the device from the list.
Read Tracks Tab
Use this tab to copy several tracks to separate files.
The Get Titles button downloads information (album, titles, etc.)
about the CD from an Internet database. An active connection to the
Internet is required. You can select a single track from the list and
use the Rename button (or Alt+R) to manually rename it. Use the
Save Tracks button (the one with the diskette icon) to save all titles
and disc information to the hard drive.
Use the Select All button to select all tracks or check the box for each track you want to save. You can preview tracks by selecting the Read Time Range tab, described below.
The Save button saves all selected tracks to separate files using the track title as the filename. You can select the file format and other options on the Save CD Tracks window that appears.
Use the Options tab to configure settings before saving.
Read Time Range Tab
Use this tab to copy any time range from the CD or to preview parts
of the CD. A fast CD-ROM or DVD drive is required for previewing.
You may have to increase the "Read speed" or the
"Number of sectors per read" setting under the Options tab
to get smooth playback. See the Options tab settings below.
The Save button saves the given time range to a single file. You can select the file format and provide a filename in the standard Save As window that appears.
Options Tab
Use the Options tab to changes settings for reading audio from the
CD and the Internet database server.
| CD Reading Options | |
| Option | Purpose |
| Number of sectors per read | Sets the number of sectors to read from the CD at one time. This value should be as large as possible, provided your CD-ROM drive supports it. Reading may fail if the value is too large for a given drive. Higher values increase overall reading speed. |
| Number of sectors to overlap | Sets the number of sectors to overlap. A value of 3 is recommended. Using a higher value more forcefully corrects jitter (positioning defects), but slows down reading because more sectors have to be re-read. Use a value of 0 if your CD-ROM drive automatically corrects jitter to speed up reading. |
| Read speed | Many CD-ROM drives allow the spindle speed to be controlled. For fast reading, this value should be set as high as possible. |
| Swap bytes | Changes the order of the bytes read from the CD. Some drives incorrectly return audio data in the reverse byte order. This gives loud, badly distorted audio. Check this option to correct it. |
| CD Database Options | |
| Server |
Sets the server address of the database to use for downloading
CD information. By default, it is
freedb.freedb.org, but you
can use a mirror site that is closer to your physical location
for faster access. The server must contain the
"/~cddb/cddb.cgi" path.
Examples: |
| Proxy |
Sets the HTTP proxy server for your network, if required. Leave this blank
if no proxy server is used. Otherwise set the server and port in
the servername:port format. Contact your system administrator
or ISP for more information if you are unable to connect to the database.
Example: |
| Automatically download titles | Automatically downloads titles when the CD Reader tool is started or when a CD device is selected. This eliminates the need to use the Get Title buttons. |
| Prefix track number in title | Prefixes the title with the track number when downloading the titles. |
| Specify CD ID manually | Allows a different CD ID to be given for the database search. When the titles are about to be downloaded, a window appears where the calculated ID can be changed. A category must be specified since it is required by the database query. This option is for advanced users and should rarely be checked. |
| Device Interface Options | |
| ASPI | Uses the Advanced SCSI Programming Interface (ASPI) to access the CD-ROM drive. Usually an ASPI driver is included with Windows 95/98/ME, but not with Windows 2000/XP. |
| SPTI | Uses the SCSI Pass-through Interface (SPTI) to access the CD-ROM. This option is not available in Windows 95/98/ME and is for Windows 2000/XP. |
| Editing Options | |
| Open track files for editing | Automatically opens each track file in GoldWave immediately after it is successfully read from the CD. This makes it easier to edit the files later. If you do not need to open or edit the files, keep this option off to save processing time. |
Troubleshooting
Use this command to show or hide the Control window. See the Control Overview section for more information.
Cue points mark and describe specific positions within sounds. They have numerous uses. When recording speech, for example, you can use them to hold information about the speaker or a translation of what the speaker said. For music, you can store lyrics for each verse. If you design instrument samples, cue points can hold looping points. Some multimedia applications use them to play or loop specific sections of a sound. When transferring albums to CDs, cue points can mark track division points, which can be used later to divide a large file into individual songs or tracks.
Cue points are shown as inverted triangles in the cue points slot of a Sound window, just above the time axis. If two cue points overlap, the colour of the cue point will be brighter.
Cue points are saved only in certain
files types.
Cue point are adjusted automatically when a file is edited, but not
when effects are applied. Any effects that alters the length of the
selection, such as
Time Warp or
Silence Reduction, will cause the
cue points within the selection to be misplaced.
There are several ways to create a new cue point:
To edit an existing cue point, you can:
To delete a cue point, you can:
Additional menu commands are provided to move cue points to the start or finish marker's position or vice versa. Right-click on a cue point in the list to display the menu.
Click on a column header to sort cue points by number, position, or name.
Select a cue and use the F4 key start playback at that cue. Press F8 to stop playback.
Copy All Button
Use this button to copy all the cue point information into the clipboard. You
can paste this into a text editor, such as the Notepad accessory.
Split File Button
The Split File button divides a large file onto smaller files using the cue
points as split positions. If you've recorded one side of an album and
need to divide it into individual songs, for example, you would set a cue
point at the start of each song, then use this feature to automatically
create separate files for each song. Each file can then be written to a
CD-R disc as a separate audio track using CD Recorder software.
Use the
Auto Cue button (below) to automatically set cue points at silences
between songs.
The Split File window options are given in the following table.
| Option | Purpose |
| Destination folder | Sets the folder where all the split files are stored. |
| Overwrite existing files | If this is checked, then files with the same name that already exist in the destination folder are replaced (overwritten). If the option is not checked, then splitting is aborted if a file with the same name is found. |
| Use cue name for filename | If selected, cue point names are used for the filenames of the split files. All cue point names must be unique, otherwise an error will be reported and splitting will be aborted. If any cue point names contain invalid filename symbols (such as /, *, ?, :, etc.), then those symbols are replaced with spaces. |
| Use base filename and number | If selected, the base filename and number (explained below) are used to automatically generate numbered filenames for the split files. |
| Use cue name prefixed with base filename and number | If selected, the base filename and number (explained below) are added to the beginning of the cue name to generate numbered filenames. |
| Prefix original filename | If checked, the name of the file being split is added to the beginning of each of the split filenames. This option is not enabled if the file does not have a name (untitled). |
| Base filename | This is used together with the number symbol, #, to create a unique name of each file. If you entered "Track##", for example, the files would be named Track01, Track02, Track03, etc. The # symbol can be placed anywhere in the base filename, so names like "#### - CD1" or "#Track##" are valid. The least significant digit is placed in the right-most # slot, so the first names would be "0001 - CD1" and "0Track01" in those cases. |
| First number | Sets the starting number to use in the # slots for naming tracks. |
| Use CD compatible wave format and alignment |
Ensures that each file is stored in a CD compatible Wave format and that the
length of each file is exactly aligned to a CD sector boundary, eliminating
gaps between files/tracks. This helps you create seamless tracks on a
CD, provided you configure your CD-R software to not write silence between
tracks.
Note that if the end of the last track file does not contain enough audio to perfectly fill a CD sector, a tiny section of audio (usually silence) may be discarded for alignment. If the file does not end in silence you can use the Edit | Insert Silence command to add 0.0133 seconds of silence to pad the end of the file before splitting. |
| Use default save format and attributes | Uses the format given under the Default Save Format tab of the Options | File Formats window to create the files. No alignment is applied. |
| Use file's current format and attributes | Uses the format and attributes of the file being split, as shown in GoldWave's status bar, to create the files. No alignment is applied. |
Any information enter through the File | Information command is stored in each split file, if possible. If the Track number is set to ##, GoldWave replaces it with a sequential number based on the order in which the files are split. The first split file will have a track number of 1, then the next file will have 2, etc.
Auto Cue Button
Use the Auto Cue button to automatically create cue points. It operates
in two modes: Mark Silence and Spacing.
The Split File button on the Cue Point window can be used later to
split a file into individual pieces based on these cue points.
Mark Silence Mode
This mode sets cue points at silences, such as marking quiet sections between songs.
A single cue point is added in the center of each detected silent region.
The Threshold value sets the volume level for the silence. In most cases, like vinyl recordings, the value should be -40dB or higher so that any background hiss, pops, or clicks are treated as silence. Otherwise no silence would be marked at all. If you find that no cue points appear, try increasing this value to -30dB or higher. If you find that too many points appear, delete them, then decrease this value or change the values below. Using the Pop/Click and Noise Reduction filters effect first may improve silence detection.
Silence length specifies how much silence is required before it is marked. Some songs contain brief silences that you usually do not want marked. This value helps to avoid marking any brief pauses within a song. Try values between 1.0 to 1.5 seconds to ignore these brief silences and mark only the longer silences between songs.
Minimum separation between cues controls the minimum amount of time between one cue point and the next. If you know all the songs are longer than 2 minutes, for example, then you can set this value to 2:00 to ensure no silences within a song are marked. All cue points will be at least two minutes apart.
Cue placement within area specifies where to place the cue point within the detected silent area. A value of 0 means at the beginning of the silence, a value of 100 means at the end of the silence. The default value of 50 places the cue point in the center of the silence area.
Spacing Mode
This mode sets cue points at fixed intervals, such as having cue points every
5 minutes. Cue points are added at the specific
interval, starting at the given time.
Start time sets the time to begin marking the file. If you enter 1:00, then the first cue point is inserted at time 00:01:00 in the file. Normally this value would be zero.
Interval specifies the time interval to use between each cue point. A value of 5:00 would set cue points at five minute intervals (00:05:00, 00:10:00, 00:15:00, etc.).
Cue Naming
The Cue naming options specify how cue points will be named as they
are added. Select Time based to name
each cue point based on its time position within the file. A cue
point added at two
minutes into the file, for example, will have the name 2:00.00000.
Select Numbered to sequentially number each cue point as they are added. The names will be 001, 002, 003, etc. The starting number depends on how many cue points are already in the file.
Select Lettered to name each cue point alphabetically, with three letters, such as AAA, AAB, AAC, etc. The starting name depends on how many cue points are already in the file.
Import/Export Buttons
Use the Import button to read cue points from a CD cue file. Use the
Export button to save all cue points to a CD cue file.
The name of the cue file depends on the name of the current Sound
file. For example, if the file you are working on is
"music.wav", then the cue file is "music.cue" by default. See
Options | Storage for a setting
to use cue files automatically.
A CD cue file contains track information that some CD Recorder programs,
such as CDRWIN and Nero, use when creating
a table of contents for a CD. Creating a cue file may eliminate the need
to split a large file into separate track files. You can open the cue
file in Notepad to edit or view its contents.
Cue point positions in a
cue file are accurate only to 1/75 of a second, which is the size
of a single CD sector. They are stored more accurately within
some sound file types.
The left window is a tree list showing all the effect plug-in modules, with the GoldWave branch expanded initially. Only effects that can be chained are listed. Effects requiring special access to the audio data (like scans) or ones that are time based cannot be chained.
You can drag-and-drop effects to the right "chain list" window or select an effect and choose the Add button. Effects are always added to the end of the chain list. Expand the branch of other listed modules to use effects in those plug-ins.
Use the Remove Last button to remove an effect from the chain list. Note that only the last effect in the list can be removed. An effect in the middle of the list cannot be removed unless all effects below it are removed first. Use the Remove All button to remove all effects in the chain list.
When an effect is added to the chain list, it appears as a button. Use the button to change settings for that effect. Settings can be changed while previewing the audio.
When you have finished creating the chain, use the Presets controls near the bottom of the Effect Chain Editor window to save the entire chain as a single preset.
The Expression Evaluator is a general purpose tool for manipulating and generating audio data. For a detailed explanation, see Appendix C.
Use the File Merger tool to join together separate files into a single file. Add files to the list by using the Add Files button. More than one file can be added at a time by selecting multple files. Files can be added via drag-and-drop or copy-and-paste from Windows Explorer as well. Use Ctrl+V on the keyboard to paste files.
Remove a file by selecting it and choosing the Remove button. The Remove All button removes all files from the list.
Files are joined in the order they are listed. Drag-and-drop files within the list to change their order.
Set the "Preferred sampling rate" for the merged file. This rate is used only if a rate is not specified in the attributes selected after you choose the Merge button. Many attributes have a predefined rate. The preferred rate will be ignored for those attributes.
As each file is merged, a cue point is added at the junction point. Check the Export cue file box to export these cue points to a separate Cue File.
Choose the Merge button to specify a filename, a file type, and attributes for the merged file. Merge processing begins immediately after you choose the Save button.
To split a file into smaller sections, see the Split File section under the Cue Points tool.
These command configure and customize GoldWave.
Use this command to change the colour scheme of Sound windows. A preview window shows the Sound's current colour scheme. Use the Item drop down list and the Colour button to change the colours or select a different scheme from the preset Schemes drop down list. You can click the mouse in the preview window to select an item you want to customize.
Displays the Control Properties window.
Undetectable Types Tab
Use this feature to associate a filename extension with an audio type and
specific attributes. This is useful for automatically opening files
that do not contain any information describing their format (raw files).
For example, if you work with Dialogic telephony files, you can associate
the .vox extension with a specific plug-in type. In this case, you'd
use the Dialogic type, usually with "ADPCM 4 bit, 8000Hz, 32 kbps, mono"
attributes. Whenever you open a .vox file, GoldWave will assume that
format without asking you to specify it each time. This must be done
prior to using File | Batch Processing
when working with raw files.
The list shows all current associations, if any. Use the Add button to create a new association. Use the Edit button to modify an existing association. Use the Remove button to remove an old association.
Example
To associate .vox with Dialogic ADPCM decoding:
See Appendix A for additional information about file attributes.
File Plug-in Precedence Tab
Use this feature to change the order that audio files are passed to
plug-ins for opening, or to enable and disable plug-ins. By default all
plug-ins found in the File plug-in folder are listed and enabled, with
the built-in GoldWave plug-in listed first.
Select an item in the list and use the Lower and Higher buttons to change the order. The plug-in at the top of the list is the first one to be given the opportunity to open a file, if it recognizes the format. Otherwise, the file is passed to the next plug-in and so on until the file can be opened or no plug-ins are left. See File Format Plug-ins for more information. Check or uncheck an item to enable or disable that plug-in module.
Default Save Format Tab
Use this feature to set a default save format for new files when
using the File | Save
command. The same format can be used for all "save" related commands
(Save As,
Save Selection As, etc.) if
you check the "Use this format for..." box. Use the
Save as type drop down list to select the type first, then use the
Attributes drop down list to select specific attributes. Whenever you
create and save a file, this format will be selected by default.
Note that when creating a new file, you can select
attributes (sampling rate and number of channels) that differ from
the default save format. Be sure to select appropriate attributes
to avoid conversions later when saving.
The "Do not allow other file types..." option disables the file type and attributes boxes under "save" related commands to prevent them from being changed. It forces the use of the default format.
Use Keyboard Options to set keyboard assigments (shortcuts) for functions in GoldWave. The list of the current assignments is given in the window with the function in the left column and the assigned shortcut in the right column. Use the arrow keys to move between items. To change a shortcut, move to right column for the function shortcut you want to change and type out the full text of the new combination of keystrokes, such as "Ctrl+Shift+Up", "X", "Ctrl+P", "Shift+Enter", etc.
Note that the Left, Right, Page Up, and Page Down keys are used for navigation
and scrolling the view and should not be reassigned. Combinations of these
keys with Ctrl and Shift are also reserved.
Choose OK to save and use keyboard assigments. Use the Defaults button to erase all keyboard assignments to restore them to their original, installed settings when GoldWave is restarted.
Use the Load and Save buttons to load or save keyboard assignments from or to a text file. These buttons allow you to backup assignments or load assignments created by others.
The following table lists some of the names of keys that differ from their physical label.
| Key | Text To Enter |
| Control | Ctrl or ^ ^X = Ctrl+X |
| Backspace | BkSp |
| Return | Enter |
| Left Arrow | Left |
| Right Arrow | Right |
| Up Arrow | Up |
| Down Arrow | Down |
| Page Up | PgUp |
| Page Down | PgDn |
| Insert | Ins |
| Delete | Del |
| Numeric Keypad 5 | Num 5 |
| Numeric Keypad / | Num / |
| Numeric Keypad * | Num * |
| Numeric Keypad - | Num - |
| Numeric Keypad + | Num + |
| System Request | (not useable) |
| Scroll Lock | (not useable) |
| Pause | (not useable) |
| Escape | Esc |
| Space Bar | Space |
| + (plus) | Shift+= (US Keyboards) |
| @ (at) | Shift+2 (US Keyboards) |
See Also
Accessibility
Keyboard Commands
Shows a list of compatible DirectX Audio Plug-ins currently installed on your system. Use this configuration to control what plug-ins GoldWave lists under the Effect | Plug-in | DirectX menu and to associate a different icon with a plug-in. Check the box next to an item in the Plug-in list to add it to the menu. Uncheck the box to remove it from the menu. Select an item, then choose an icon from the Icon list to associate that icon with the item. Note that GoldWave must be restarted before changes are acknowledged.
GoldWave requires plug-ins that process
in IEEE 32 bit floating point audio (WAVE_FORMAT_IEEE_FLOAT) format.
It will not list plug-ins that process only in PCM 16 bit audio format.
If you are unable to get a particular
DirectX Audio Plug-in working, please contact the plug-in developer,
not GoldWave Inc.
Allows you to disable system decoders provided with Windows and Media Player to handle file formats that GoldWave cannot open directly. Disabling system decoders may solve problems that occur when opening new file formats handled by external file format plug-ins.
The Storage Options window configures folders, file storage, and undo levels. See the Storage Overview section for additional information.
Sound Folder
This specifies the folder to use when you start GoldWave. The
File | Open
command lists files in this folder whenever you start GoldWave.
You can make GoldWave remember the last folder used or always
start in the given folder. Use the folder button to browse
for a folder.
Temporary Storage
This controls how GoldWave stores audio, as explained
in the Storage Overview section.
When using hard drive storage, you need to specify the folder to use when
creating temporary files. This folder should be located on a large,
local hard drive with plenty of free space. Changing this folder does
not affect opened files already in temporary storage.
Use RAM storage only when working with small files for faster, efficient
editing.
Undo Levels
This sets the number of edits/changes you can undo. A value of 5
means you can undo the five most recent changes performed on
the file. A value of 0 means that you cannot undo any
changes. Larger values use much more storage to store the
previous audio data. When using the RAM storage option, this
should be set to a small value. Otherwise RAM may be depleted
quickly and system performance will degrade.
Cue Points Storage
Use the "Automatically import and export separate cue file" setting
to automatically save or load cue points from a separate
cue file when
saving or opening a sound file. If you open a file named "music.mp3"
cue points are imported from a "music.cue" file, if it exists.
If you save a file named "music.mp3", cue points are saved
in a "music.cue" file (overwriting any existing file).
Automatic import and export is done only for file formats that cannot
store cue point information internally. Cue files are not saved
automatically for other file formats (such as .wav). Use the
Cue Points Tool to manually import
or export a cue file.
When automatically exporting cue files, any existing cue file with the
same name will be overwritten without notification!
The Tool Bar Options let you customize GoldWave's tool bars. Initially the top tool bar contains main menu commands, while the lower tool bar contains only effect commands. Select an item and use the Add and Remove buttons or drag-and-drop items between the Available and Current lists to control the layout of the tool bars. The order which items appear in the Current list is the same as the order they will appear on the tool bar. You can drag-and-drop items within the Current list to rearrange them.
Use the Visible checkbox to show or hide the entire tool bar. Use the Gray-scale checkbox to convert the coloured images to blank-and-white. Use the Captions checkbox to display text within the Main tool bar buttons. Use the "Tool bar images in menu" checkbox to display images in GoldWave's main menu. Do not check this option if screen reader accessibility is required.
You can drag-and-drop the tool bars themselves within GoldWave's Main window to change their locations. Click-and-hold on the vertical bumps near the left edge of the tool bar to drag it.
Use Window Options to configure the positions of the Main and Sound windows, set axis options, specify preset and initial zoom values, and set miscellaneous options.
Main Window Size
This controls the Main window's position and size when GoldWave
is started. Normal gives control to Windows. Maximize makes
the Main window occupy the entire screen. Save position saves the
Main window's position and size when GoldWave is closed so that
it will appear in the same location next time.
Sound Window Size
This controls the position and size of Sound windows. Normal
gives control to Windows, which usually results in cascaded
windows. Maximize makes a Sound window occupy the entire
Main window. Auto-tile resizes all Sound windows whenever a
new sound is opened or closed so that every one is visible.
Arrangement is the same as the
Window | Tile Horizontally
command.
Y Amplitude Axis Numbering
Displays or sets the units of the vertical axis in Sound windows.
Selecting Off hides the axis completely. Normalized shows an
axis with a range of -1.0 to 1.0. Signed 16 bit shows an axis
ranging from -32768 to 32767, which is the range of a 16 bit
sample. Unsigned 8 bit shows an axis with a range of 0 to 255.
X Time Axis Numbering
Sets the format for displaying the horizontal time axis near the
bottom of Sound windows. "Hours : minutes : seconds"
gives the time as three sets of numbers separated by colons, such as 12:23:56.
"Minutes : seconds" gives the time as two sets of numbers
separated by colons, such as 1234:56. Seconds gives the time as a
floating point number, such as 1234.56.
Zoom
The Preset zoom value is used to for the View | Preset
command. See View | Specify for more information about
zoom values. The Initial zoom specifies the zoom level to use when
a file is opened. If you choose All, then the entire waveform is
shown using whatever zoom level is required.
Miscellaneous
If the "Always confirm before saving" option is checked, you are
asked to confirm saving whenever you use the
File | Save command.
If the "Draw overview graph" option is checked, then the overview is graphed based on audio from the file. Otherwise it is drawn as simple lines, which is quicker since the entire file does not have to be scanned.
If the "Use left and right mouse button selection method" option is checked, then the start marker is set by using the left mouse button and the finish marker is set by using the right mouse button. No context menu appears. You will not be able to use the new right-click-and-drag features. This is the way older versions of GoldWave worked.
If the "Update default effect settings after each use" options is checked, then the last settings you use for an effect will be used the next time the effect window is shown. Essentially the "Default" preset is automatically updated when you choose the OK button on an effect window.
These commands organize Sound windows and the Control window. See Interface Overview for additional information.
Cascade layers Sound windows on top of each other so that their title bars are visible. Tile Horizontally arranges Sound windows above and below each other (or side by side if necessary) so that all windows are visible. Tile Vertically arranges Sound windows side by side (or above and below each other if necessary) so that all windows are visible. Minimize All minimizes all Sound windows so only a small title bar is shown. Arrange All arranges minimized Sound window in rows on the bottom of the Main window.
Classic Control places the Control window in the bottom right corner of the screen and arranges the controls and visuals in a square layout similar to previous versions of GoldWave. The Control window can be resized in any direction. Horizontal Control places the Control window along the bottom of the Main window and arranges the controls and visuals horizontally. Makes the visuals small. The Control window cannot be resized vertically in this mode. Vertical Control places the Control window along the right side of the Main window and arranges the controls and visuals vertically. Makes the visuals large. The Control window can be resized in any direction.
A list of all currently opened Sound windows is given at the bottom of the Window menu.
Starts Window's Help and gives a list of contents for GoldWave help.
Starts the default web browser to display the GoldWave manual (this file).
Displays GoldWave version and registration information.
Syntax:
"C:\Program Files\GoldWave\GoldWave.exe" -parameter "filename" ...
Quotes are required around the program path and the filename. Parameters are not case sensitive. Multiple filenames and parameters can be used. All files are opened.
| Parameter | Purpose |
| Close | Closes the program after playing the file on the
command line. This paramater must be used with Play. If
the user interacts with the program, it may not be closed.
Example |
| Config | Displays the GoldWave Setup Options window to configure
core functionality in GoldWave. This is the same as using Start | Programs
| GoldWave | GoldWave Setup.
Use this only when the program is not functioning correctly due to hardware or operating system limitations or problems. Buttons to backup and restore all GoldWave settings and presets to and from a separate file are given on this window. |
| Clipboard:filename |
Copies the specified file to the clipboard for batch processing.
This can be used only with Process. If the filename contains
spaces, double quotation marks must enclose the entire parameter.
Example |
| New | Creates a new file using the default settings given under the File | New command. Use this with Record to start recording in a new file from the command line. |
| Nosplash | Suppresses the splash window when starting the program. |
| Play | Plays the file given on the command line. Use Close to automatically close the program after the file has finished playing. Only the first file on the command line is played. |
| Process Proclog |
Starts GoldWave in batch processing mode. Most other parameters are ignored. See File | Batch Processing for details. |
| Record | Start recording in the file given on the command line.
Use this with New to create a new file for recording.
Example |
| Region:Start,Length |
Sets the initial selection for the first audio file given
on the command line.
Start is the starting point in samples. Length is the
number of samples to select. If either Start or Length
contain a decimal point, then both values are assumed to be
units of time rather samples.
Examples |
| Same | Opens a file in the current running instance of GoldWave rather than starting another copy of the program. Use this when associating a file type with the program. |
GoldWave ("the package") includes the following software and documentation:
GoldWave.exe GoldWave application file GoldWave.hlp GoldWave help GoldWave.htm GoldWave manual text Images (*.PNG) All images associated with this manual GWPreset.reg Presets and shapes ReadMe.txt Important information WhatsNew.txt Revision history DirectX.pig DirectX Audio plug-in wrapper FLACFile.pig FLAC file plug-in OggFile.pig Ogg Vorbis file plug-in WMAFile.pig Windows Media Audio file plug-in
The package is provided as is, without warranty of any kind. GoldWave Inc. shall not be liable for damages of any kind. Use of this software indicates you agree to this.
The package and this documentation are copyright © 2007 by GoldWave Inc. All rights reserved. This document and associated images may not be shown on a public website.
GoldWave is a trademark of GoldWave Inc.
Matlab is a trademark of The Math Works Incorporated.
Windows & Microsoft are registered trademarks of Microsoft Corporation.
Sound Blaster is a trademark of Creative Labs Incorporated.
All other trademarks/registered names acknowledged.
The latest information and updates can be found on the GoldWave website:
If you encounter any problems, please check the following information:
If a problem still cannot be resolved, please send a detailed description to the address below.
Questions, comments, and suggestions are welcome. You can find contact information here:
http://www.goldwave.com/contact.php
Postal Address
GoldWave Inc.
P.O. Box 51
St. John's, NF
CANADA A1C 5H5
Reducing images and audio to a digital state requires an analog-to-digital conversion. Instead of using just one bit of information, many bits are used to more accurately store the state. By using 2 bits, for example, four states are possible: 00, 01, 10, 11. For images, that could be black (00), dark gray (01), light gray (10), and white (11). For audio, that would give four different levels of loudness. Typically, many more bits are used. Most computer video cards use 16 to 32 bits to store a single dot. Sound cards typically use 16 bits for audio levels.
The number of bits to use depends on human perception and bit alignment within computers. Computer tend to bundle bits in groups of 8, called bytes, so using 8, 16, 24, or 32 bits would fit nicely in 1, 2, 3, or 4 bytes respectively. For images, 16 bits do not provide enough states to make the transition from one state to the next imperceptible, so 24 or more bits are used. For audio, 16 bits are adequate, which is what a CD contains, but audio systems using 24 bits will be common in the future.
Digital audio is composed of thousands of numbers, called samples. Each sample holds the state, or amplitude (loudness), of a sound at a given instant in time. For images, each point of light, or pixel, has a certain brightness and location and all pixels combine to make a picture (see figure below). For digital audio, all the samples combine to make a waveform of the sound.
When playing audio, each sample specifies the position of the speaker at a certain time. A small number moves the speaker in and a large number moves the speaker out. This movement occurs thousands of times per second, causing vibration, which we hear as sound.
There are several attributes that determine the quality and size of a digital audio file. They are the sampling rate, the bit depth, the number of channels, and the bitrate.
The sampling rate is the number of times, per second, that the amplitude level (or state) is captured. It is measured in Hertz (seconds-1, Hz). A high sampling rate results in high quality digital sound in the same way that high resolution video shows better picture quality. Compact disks, for example, use a sampling rate of 44100Hz, whereas telephone systems use a rate of only 8000Hz. If you've ever heard music on the telephone while on hold, you'll notice a big difference in quality when compared to the original music played on a CD player.
Higher sampling rates capture a wider range of frequencies and maintain a smoother waveform. The figure below shows a real world waveform in red and the digital waveform in black at different sampling rates. You can see that increasing the sampling rate makes each step of the digital waveform narrower. The shape more closely follows the real world. In general, the height of each step is reduced as well, but that depends on the number of bits. In simple terms, the sampling rate controls the width of each step.
The rate to use depends upon the type of sound and the amount of storage space available. Higher rates consume a lot of space. In the above example, the CD requires over 5 times the amount of storage as the telephone system for the same digital sound. Certain types of sounds can be recorded at lower rates without loss of quality. Some standard rates are listed in the table.
| Attributes | Quality and Usage | Storage (16 bit/mono, MB/minute) |
| 8000Hz | Low quality. Used for telephone systems. Good for speech. Not recommended for music. | 0.960 |
| 11025Hz | Fair quality. Good for speech and AM radio recordings. | 1.323 |
| 22050Hz | Medium quality. Good for TV and FM radio quality music. | 2.646 |
| 44100Hz | High quality. Used for audio CDs. | 5.292 |
| 48000Hz | High quality. Used for digital audio tapes (DAT). | 5.760 |
| 96000Hz | Very high quality. Used for DVD audio. | 11.520 |
As explained in the Digital Audio Basics section, the number of bits determines how accurately the amplitude of the waveform is captured. The figure below shows a real world waveform in red and the corresponding digital waveforms with 2 bit samples and 3 bit samples.
You can see that adding a single bit greatly improves the way the digital waveform conforms to the real world waveform. The 2 bit waveform looks like a rough approximation with large steps. Several amplitudes are rounded to the same state, such as samples 9 through 11. This is a source of quantization noise, explained later.
In the 3 bit waveform, no amplitudes are rounded to the same state. Each step is half the height of the 2 bit waveform, but it is still not perfect. From sample 1 to sample 2, there is a jump in the waveform, which also causes quantization noise to a much lesser extent. You'll notice that samples 0 and 1 are below the real waveform and samples 2 and 3 are above the waveform. This occurs because there are no in-between states to accurately store those amplitude levels, so the digital waveform ends up straddling the real one. Therefore more states, and bits, are needed.
8 bit and 16 bit samples are common. In an 8 bit sample, there are 256 different states or levels of amplitude. 16 bit samples have 65,536 levels. This makes a huge difference it terms of sound quality. Audio stored as 8 bit samples will often have much more quantization noise.
Samples can be stored as bits a couple of different ways. One way is to consider all the states as positive, with no values below zero. As shown in the figure above, the states 00, 01, 10, and 11 are the same as the positive numbers 0, 1, 2 and 3. This eliminates the need for a negative sign. Such samples are called unsigned. For 8 bit samples, the states would range from 0 to 255.
The other way is to use a form known as two's complement, which allows both positive and negative values. These samples are called signed. Since real world waveforms tend to fluctuate through a range of positive and negative values, signed samples are preferred. For 16 bit samples, the states would range from -32678 to 32767.
When a sample is stored using more than 8 bits, more than one byte is needed. The term endian is used to describe the way bytes are ordered in computer memory. It specifies the significance of the first byte in the group. A 16 bit sample, for example, requires exactly two bytes, byte A and byte B. They can be stored as A first, then B or as B first, then A. Generally a PC will store them one way and a Mac will store them the other way due to differences in the internal processor design of those systems.
Big endian order has the most significant byte stored first, making it similar to the way we read numbers. In the number 47, the 4 is first and is most significant and the 7 is last and is least significant. This ordering is used on Mac systems.
Little endian order has the least significant byte stored first, allowing some optimizations in processing. This ordering is used on Intel and PC systems.
Digital audio can have one or more channels. Single channel audio, referred to as a monaural (or mono) audio, contains information for only one speaker and is similar to AM radio. Two channel audio, or stereo audio, contains data for two speakers or two ears, much like FM stereo. Stereo sounds can add depth, but they require twice as much storage and processing time as mono sounds. Most movie theatres have advanced audio systems with 4 or more channels, which are capable of making sounds appear to come from certain directions. GoldWave currently supports mono and stereo audio only.
Since digital audio is limited by the sampling rate, the number of bits, and the number of channels, a digital waveform can never be an exact replica of the real world waveform. These limitations can lead to a number of problems, such as aliasing, clipping, and quantization noise.
Aliasing occurs when the sampling rate is not high enough to correctly capture the shape of the sound wave. The recorded sound will have missing tones (Figure: Aliasing, top) or new tones that never existed in the original sound (Figure: Aliasing, bottom). These problem can be eliminated by using higher sampling rates or by using anti-aliasing filters.
Higher sampling rates increase the number of sampling points. To see how this works, try adding a three points between each sampling point in the figure and redraw the graph. The recorded sound will more closely resemble the input.
Anti-aliasing filters remove all tones that cannot be sampled correctly. They prevent high pitched tones from being aliased to low pitch. Many sound cards include anti-aliasing filters in hardware.
Clipping errors occur when the level is too high to be stored in the bits available. For example, if the maximum level for a 16 bit sample is 32767, and the actual level is 40000, then it must be clipped to 32767 to fit (see the figure). This generates distortion. To eliminate clipping, adjust the recording volume before recording. By using the Control window's Monitor input feature, you can adjust the volume to a suitable level. The volume is low enough when the VU Meter visual does not reach the top of the red region.
Clipping can occur when processing effects, such as increasing the volume with the Change Volume effect. GoldWave does not clip audio internally, but it has to be clipped when sending the audio to the sound card or when saving a file. You should use the Maximize Volume effect with a 0dB setting before saving to ensure that no clipping occurs in the file.
The real world has an infinite number of states, but the digital world has a very limited number of states. Quantizing is process of assigning an amplitude to this limited quantity of states. Quantization noise occurs when an amplitude is rounded to the nearest state for the given number of bits. This is illustrated in the Bits section. Using 2 bits causes large rounding errors and only roughly follows the real waveform. Using 3 bits has much smaller rounding errors, but is still nowhere near as smooth as the real world waveform. Using 8 bits gives a much smoother waveform, but the round errors can still be heard. Using 16 bits gives errors so small that they are almost imperceptible.
To minimize internal and external noises, make sure your sound card is installed as far away from your graphics card as possible. If the sound hardware is integrated into the computer's main board, consider purchasing and installing a separate sound card instead. Integrated hardware tends to have a much higher noise level.
Keep all microphones and audio cables away from your monitor, computer case, or other sources of electrical noise. Use shielded cables and make sure everything has a common ground connection. Plug equipment and devices into the same power strip so that they share the same ground. Equipment without a grounded plug (three pronged plug) may require special grounding. Sometimes rotating the plug helps reduce a hum or buzz.
System configuration can also affect audio quality. Due to architectural problems with PCs and excessive virtual memory swapping by Windows, you may notice an occasional gap when recording. Restarting Windows, updating the sound driver, shutting down all other programs, or installing more RAM helps to minimize that problem.
The waveform levels or states can be interpreted in a number of different ways. Reading the states in their binary form is impractical, so the states are mapped to different scales that are more human-readable. These include a simple amplitude scale, a percentage scale, and a decibel scale. As the figure shows, a 1.0 amplitude level is the same as a 100% level and a 0dB level.
The amplitude scale simply maps the states to a linear range of -1.0 to +1.0, with zero being silence. The amplitude typically is given as a positive value when used in effects.
The percent scale is 100% at the maximum and 0 at silence. It is essential the same as the amplitude scale converted to a percentage by multiplying by one hundred and adding a percent sign.
The decibel scale is unusual in that it is 0dB at the maximum and negative infinity at silence. It is a logarithmic scale, which is closer to the way human hearing perceives sound levels. You'll notice from the above figure that there are no positive levels. Levels below the maximum are negative. Only values above the maximum are positive (not shown). When changing the volume, positive values increase the level and negative values decrease the level.
Use the equation below to convert a decibel level to a percentage level.
When setting the volume in an effect, the value may be interpreted as a relative level or an absolute level. When changing the volume, it is usually relative. When specifying a threshold, it is usually absolute. Relative changes are cumulative. So if you apply a volume change with 0.5 amplitude (50% or -6.02dB), then the amplitude decreases to half its current level. If you apply that change again, then it decreases to one quarter of its original level. In other words, the change is relative to its current level. Given an original amplitude of A, the first change yields a result of A x 0.5 and the sample is replaced by that value. The second change takes that value and multiplies it by 0.5 again, so we get the final result of (A x 0.5) x 0.5 or A x 0.25. Most of the effect settings in GoldWave are cumulative. For relative changes, using 1.0, 100%, or 0dB does not alter the sound at all.
Absolute levels are used in places where a threshold is required, such as in Silence Reduction, or in rare cases were the absolute level is set, such as Maximize Volume. Absolute changes are not cumulative. If you maximize the sound with 0.5 amplitude (50% or -6.02dB), then that is what the peak level will be no matter how many times the effect is used. For absolute changes, using 1.0, 100%, or 0dB may alter the sound if it is not currently at that level.
Sound windows in GoldWave show sound as a waveform of amplitudes on a time axis. However, sound can be viewed in an entirely different way by examining its frequency and pitch content or frequency spectrum. Sounds are broken down into a combination of simple fundamental (sinusoidal) tones, each with a different frequency. This is useful for examining bass and treble levels or for isolating and studying certain sounds.
Average human hearing spans a frequency range from about 20Hz to about 17000Hz. The figure below shows some common sounds and the frequency range they cover.
Many people wonder why it is difficult to remove vocals from
music. From the figure, you'll see there is a large overlap in the
frequency range of speech and music. Removing the vocals would
also remove a significant part of the music. A similar problem
occurs when removing hiss noise, since it often covers the entire
spectrum.
Most basic stereo systems have bass and treble controls, which offer limited control over a frequency spectrum. Bass applies to low frequency sounds, such as drums, cellos, low piano notes, or a hum noise. Treble applies to high frequency sounds, such as a clash of cymbals, a tweet of a small bird, high notes on a piano, or a hiss noise.
More expensive stereo systems have Graphic Equalizers, which provide better control over a frequency spectrum. Instead of controlling just two bands (bass and treble), you can control many bands.
GoldWave provides even more control over frequency spectrums with filter effects such as Parametric EQ, Low/Highpass, Bandpass/stop, Equalizer, and Noise Reduction.
The frequency range of a digital sound is limited by its sampling rate. In other words, a sound sampled at 8000Hz cannot record frequencies above 8000Hz. In fact, the sound cannot even have frequencies above 4000Hz. According to the sampling theorem, the maximum frequency is limited to half the sampling rate. Any higher frequencies will be aliased, to lower ones, causing noise if appropriate filters are not used.
CD audio is designed to cover the full range of human hearing, which has a maximum of under 22kHz. In order to successfully record this range, the sampling theorem states that a sampling rate of at least twice the maximum must be used, so a rate of at least 44kHz is required. The actual rate is 44100Hz for standard CD players.
Several of GoldWave's Control visuals convert sounds into a range of frequency bands using a radix-2 fast Fourier transform (FFT) algorithm. When the results are drawn using colours, the graph is referred to as a spectrogram. When the results are drawn with lines, it is often referred to as a frequency spectrum or frequency analysis.
Frequency analysis graphs are displayed in the Noise Reduction and Parametric EQ filter effects. These help you to locate frequencies that you want to remove or enhance.
GoldWave applies a windowing function to the data before performing the FFT (see Control Visual Properties section). This reduces "discontinuity" errors that occur when dividing data into small chunks. A Kaiser window is used by default.
To make the spectrum more realistic to human hearing, magnitudes are scaled logarithmically. This means that if one frequency "sounds" twice as loud as another, it is graphed with twice the height (or the corresponding colour for the spectrogram).
Uncompressed audio files tend to be large. CD quality audio requires ten megabytes per minute. That is not a problem with large computer hard drives available today, but it is a problem if you want to save many songs on a portable player or if you want to transfer files over the Internet. Unlike most computer data, audio data does not compress very well using typical compression methods such as those found in programs like PKZIP or WinZip. These methods preserve the data exactly so there is no loss of quality. Such compression is called lossless compression.
To make audio files smaller, complex algorithms have to used. Most of these algorithms sacrifice some quality so that when the data is decompressed, you do not get exactly the same quality you had originally. This type of compression is known as lossy compression. Ideally the quality that is lost is not perceptible, so you do not notice the difference.
The most common method of lossy audio compression is MPEG Layer-3, better known as MP3. It is capable of getting near CD quality audio in less than one tenth the size, which is about one megabyte per minute. More recent algorithms, such as Ogg Vorbis and Windows Media Audio, get even better quality in a smaller size.
Software and hardware that compress audio using complex algorithms are referred to as codecs (from coder/decoder). Compression is the same as encoding and decompression is the same as decoding.
To control the level of compression in GoldWave, use File | Save As and select a different file type and/or attributes. The lower the bitrate (the kbps number, see below), the smaller the file will be, usually at reduced quality.
When opening, editing, and saving a file repeatedly, it is best
to use a lossless file format. Every time a lossy compressed file is
re-opened and saved, some quality is lost. For long term editing,
use the Wave type with "PCM 16 bit" attributes or one of the lossless
compressed formats such as FLAC or Windows Media Audio with "lossless"
attributes.
Many compressed audio formats measure the compressed size as a bitrate. The bitrate is the number of bits per second (bps) required to store the audio. Usually the number is given in kilobits or one thousand bits. Divide that number by 8 to determine the number of kilobytes required per second.
Internet connection speed (bandwidth) is often measured in bitrates as well. A 56k modem is capable of receiving 56 kilobits per second. If you want people to stream your MP3 audio over a modem, you'll need to compress the file using a maximum bitrate of 56kbps. Due to the connection overhead and Internet protocol, a lower rate would have to be used to ensure the audio can be downloaded fast enough. For DSL and Cable Internet connections, the standard 128kbps MP3 rate can be used.
Audio files can contain a wide range of sounds, from noisy cymbal clashes to silence. Algorithms typically get much better compression on silence or simple audio sections than on complex, noisy audio. This means that the bitrate depending on whether constant bitrate or variable bitrate compression is used.
Constant bitrate is useful for broadcast systems where the transmission rate is fixed. It also make is easy to seek to arbitrary positions within the audio stream or file.
Variable bitrate gives the best compression and quality. However, it makes it difficult to seek within the stream or file since there is no direct relation between time and size.
In addition to all the standard menu keystrokes, such as Alt+F O to open a file, Alt+E C to copy, Alt+E P to paste, etc., GoldWave includes a number of additional keyboard shortcuts. These are summarized in the following table. Use Options | Keyboard to reassign shortcuts.
| Keystroke | Action |
| Ctrl+A | Selects the entire sound. |
| Ctrl+B | Pastes the clipboard into the sound at the beginning. |
| Ctrl+C | Copies the selection into the clipboard. |
| Ctrl+D | Displays the Crossfade window. Crossfades the clipboard with the selection. |
| Ctrl+E | Pastes the clipboard into the sound at the end. |
| Ctrl+F | Pastes the clipboard into the sound at the finish marker's position. |
| H | Starts playback relative to the mouse's horizontal position in the waveform. |
| J, K, L | Rewinds, plays, and fast forwards respectively from the current cursor's position. Playback stops at the finish marker but can be continued. Playback stops at the start marker when rewinding. |
| Shift+J, Shift+K, Shift+L | Makes the playback speed slower, normal, and faster respectively. |
| Ctrl+J | Jumps the start marker to the next cue point. |
| Ctrl+Shift+J | Jumps the start marker to the previous cue point. |
| Alt+J | Jumps the finish marker to the next cue point. |
| Alt+Shift+J | Jumps the finish marker to the previous cue point. |
| Ctrl+K | Overwrites the selection with the clipboard. |
| Ctrl+M or Shift+Ctrl+Ins | Displays the Mix window. Mixes the clipboard with the sound at the start marker's position. |
| Ctrl+N | Creates a new sound. |
| Ctrl+O | Opens a sound. |
| Ctrl+P | Pastes the clipboard into a new Sound window. |
| Ctrl+Q | Drops a new cue point at the current playback, recording, or start marker position. Cue naming is controlled by the Auto Cue settings under the Cue Points tool. |
| Ctrl+Shift+Q | Drops a new cue point at the current playback, recording, or start marker position and displays the edit window. |
| Ctrl+R | Replaces the selection with the clipboard contents. |
| Ctrl+S | Saves the file. |
| Ctrl+T | Trims the sound. Removes all audio outside the selection. |
| Ctrl+U | Displays the AutoTrim window. Removes silences within the ends of the selection. |
| Ctrl+V | Pastes the clipboard into the sound at the start marker's position. |
| Ctrl+W | Sets the select to the view (as in Select View). |
| Ctrl+X | Cuts the selection and copies it into the clipboard. |
| Ctrl+Y | Redo. Reverses last undo. |
| Ctrl+Z | Undoes last change. |
| Ctrl+Shift+B | Selects both channels of a stereo file. |
| Ctrl+Shift+L | Selects the left channel of a stereo file. |
| Ctrl+Shift+R | Select the right channel of a stereo file. |
| Shift+0 | Zooms 10:1 horizontally. |
| Shift+1 | Zooms 1:1 horizontally. |
| Shift+2 | Views 1 second horizontally. |
| Shift+3 | Views 10 seconds horizontally. |
| Shift+4 | Views 1 minute horizontally. |
| Shift+5 | Views 1 hour horizontally. |
| Shift+A | Horizontally zooms all the way out. |
| Shift+E | Displays the Set Marker window. |
| Shift+M | Stores the locations of the start and finish markers. |
| Shift+P | Zooms to previous horizontal zoom. |
| Shift+R | Moves the start and finish markers to the stored locations. |
| Shift+S | Horizontally zooms in on the selection. |
| Shift+U | Horizontally zooms to the user defined level. |
| Shift+V | Vertically zooms all the way out. |
| Shift+Y | Displays window to specify horizontal zoom. |
| Del | Deletes the selection, permanently. |
| [ (left bracket) | Drops the start marker at the current playback position. |
| ] (right bracket) | Drops the finish marker at the current playback position. |
| Shift+[ | Plays three seconds of audio up to the start marker. |
| Shift+] | Plays three seconds of audio up to the finish marker. |
| Ctrl+[ | Plays from the start marker to the finish marker. |
| Ctrl+] | Plays from the finish marker to the end. |
| Shift+\ | Moves the start and finish selection markers to their previous positions. |
| Left | Scrolls the Sound window view left. |
| Right | Scrolls the Sound window view right. |
| Page Up | Scrolls the Sound window view left one page. The amount of time scrolled depends on the current view zoom level. If you used the green playback button in view mode to play the file, playback is restarted at the new scrolled position. |
| Page Down | Scrolls the Sound window view right one page. See the Page Up key for more details. |
| Home | Moves the Sound window view to the start marker's position. |
| End | Moves the Sound window view to the finish marker's position. |
| Ctrl+Home | Moves the Sound window view to the beginning of the sound. |
| Ctrl+End | Moves the Sound window view to the end of the sound. |
| Shift+Home | Moves the start marker to the beginning of the sound. |
| Shift+End | Moves the start marker to the finish marker's position. |
| Ctrl+Shift+Home | Moves the finish marker to the start marker's position. |
| Ctrl+Shift+End | Moves the finish marker to the end of the sound. |
| Shift+Left, Shift+Right | Moves the start marker left or right. |
| Ctrl+Shift+Left, Ctrl+Shift+Right |
Moves the finish marker left or right. |
| Shift+Up | Horizontally zooms in. |
| Shift+Down | Horizontally zooms out. |
| Ctrl+Up | Vertically zooms in. |
| Ctrl+Down | Vertically zooms out. |
| Space | Plays or stops a sound (toggles playback). The region that is played depends on the Green play button settings. |
| F4, F5, F6, F7, F8 | Plays, rewinds, fast forwards, pauses, and stops respectively. |
| Shift+F4 | Plays the sound with the Yellow play button settings. |
| Ctrl+F9, Ctrl+F8, Ctrl+F7 | Starts, stops, and pauses recording respectively. |
| F11 | Displays Device Controls Properties window. |
| F1 | Starts help. |
| Ctrl+F4 | Closes the Sound window. |
| Alt+F6 | Switches between Main window and Device Controls window. |
| Ctrl+F6 | Switches between Sound windows. |
The Expression Evaluator is a versatile tool for manipulating and generating audio data.
The Destination box specifies the Sound window where results of the evaluation will be stored. The drop down list contains all Sound windows in the form "X - Filename", where X is the wave identifier number of the Sound window. For example, a Sound window with the title "Hello.wav" could appear as "1 - Hello.wav" in the list. By default, the destination is set to the current Sound window. You can change the destination, if more than one Sound window is opened, by using the up and down keys or by selecting it with the mouse from the drop down list.
The Source box lists currently opened Sound windows. By selecting a source from this list, the function waveX(n) is placed in the expression. X is the wave identifier number, as explained above.
The large Expression box located at the top of the window is where an expression is entered. A list of valid operations and functions is given in a following section. In most cases, expressions will be some function of n or t, just as in regular math, where y is usually a function of x (i.e. y = f(x)). In the expression evaluator, we can have destination = f(t), where f(t) is any expression you enter.
To create a simple tone, for example, you would enter the expression "sin(600*t)". You can even alter an existing sound. To double the volume of "1 - Hello.wav", for example, you would select it as the destination and enter the expression "wave(n)*2".
To enter an expression, you can:
The evaluator uses several special variables, which you can initialize in the "Incremented variables" and "User constants" groups. These variables are discussed later.
After you have specified the destination, expression, and initial values, choose the Play button to preview it. Choose the OK button (or just press the Enter key) to begin evaluation. If you entered an expression incorrectly, a message will be displayed. If the expression is valid, a processing window will appear. Since the evaluation process takes time, you can stop it at any time with the Cancel button. No changes will be made to the sound if you cancel processing.
You can copy, cut and paste expression in the Expression box using
the usual keystrokes (Copy = Ctrl+C, Cut = Ctrl+X, Paste =
Ctrl+V). You can also copy and paste expression from the online
help.
To speed up evaluation, make sure that you are using RAM storage
(see Options | Storage).
Knowledge of the structure of digital audio is essential to understand how the evaluator works. To illustrate this structure, let's assume we have the following sound:
Title bar: Hello.wav Total length: 2.0 seconds Sampling rate: 8000Hz Start marker: 0.5 seconds Finish marker: 1.2 seconds
Digital audio is stored as a series of amplitudes, which are often referred to as samples (see figure). The evaluator interprets each sample as a value between -1 and 1, inclusive. Only samples between the start and finish markers are considered valid; all other values are assumed to be zero. The number of samples selected is defined as N.
Each sample has a relative index number, n, and a time, t. Since the time of each sample depends on the sampling rate, it is usually written in terms of the unit of time between each sample, T. You many have noticed that the time, t, is related to the index number, n, by the equation t=nT. The figure shows how all these variables relate to the structure of the sound.
Let's assume we have entered the expression "sin(t)". Since expressions are evaluated over the selection range, the initial value for t is automatically set to start marker's position of 0.5. Choosing the OK button evaluates the expression from t = 0.5 to t = 1.2 in steps of 1/8000 of a second, as defined by T. This means that the expression is calculated for each sample in the selection, changing each sample as follows:
The sample index is useful for modifying an existing sound. If we want to double the amplitude of Hello.wav, for example, we need to multiply each sample by two and store it back into the sound. In this case, Hello.wav will be both the destination and the source. To set it as the destination, we simply select it from the Destination list. To use it as a source, we need to determine its wave identifier number. These numbers are provided in the Source list. Assuming it is listed as "3 - Hello.wav", we now know that its wave identifier number is 3. This number is necessary for the evaluator's wave function, which has the following syntax:
| waveX( n ) | where: | X = wave identifier number n = sample index number |
Note that when the destination and the source are the same,
no number is needed after the wave function. If
"wave(n)" is used, then the destination sound is assumed
to be the source.
In the evaluator, the index number, n, is relative to the start marker.
This means that the start markers position is added to the index
number (i.e. n+Start). For the example in the figure above, a relative
index of n=0 has an absolute index of 4000.
The final expression is "wave3(n)*2". Choosing OK evaluate this expression from n=0 to n=5600 in steps of 1 (note that N = 5600) . This produces the following changes (remember than n is relative to the start marker's position):
Note that N and n are always integers. The evaluator rounds indices to the nearest integer, so the expression "wave3(.7)" would be calculated as "wave3(1)".
You can use the sample index number and the wave function to mix two or more wave together. If you have several sounds opened, you can obtain the wave identifier number for each sound from the Source list. If the sounds you wanted to mix were identified as 2 and 3, you would enter the expression:
Care must be taken when indexing signals with different sampling rates. If wave1 is a voice recorded at 11025Hz and wave2 is music recorded at 22050Hz, then with wave1 as the destination you'd need to use this expression to correctly mix them:
Ideally, wave2 would have to be low-pass filtered first. If wave2 is the destination, the expression would be:
The variable N has several uses, such as reversing a sample. If wave2 is a new sound that has the same sampling rate and length of wave1, then setting the destination to wave2 and using the expression
makes wave2 the reverse of wave1.
User constants can be set to any values you choose. None of these values change during evaluation. They just provide a way of easily changing parameters within an expression without having to edit the expression directly. The constant f is typically used for frequency values. By using the expression
you can generate any tone by specifying the frequency in the f box and the amplitude in the y box.
The following equations convert between time and sample index number. The start value is the position of the start marker (in seconds).
The Presets tree list in the Expression Evaluator window organizes expressions in a number of groups, such as Dial Tones, Effects, Noise, and Waves. You can create new groups or add expression to existing groups.
To retrieve an expression:
To add an expression:
button in the Presets box.To delete an expression:
button in the Presets box.The following table summarizes evaluator operators and functions.
Table B.1: Evaluator Operators and Functions
| Label | Operation, function |
| (, ) | Parenthesis |
| +, *, -, / | Add, multiply, subtract (negate), and divide |
| % | Modulus operator (remainder) |
| <, > | Greater and less than operators. Compares left and
right operands. Gives 1 if true or 0 if false.
For example, x > y evaluates to 1 if x
is greater than y or 0 otherwise.
The expression |
| ^ | To the power of, yx |
| pi | Constant (3.14159...) |
| cos | Cosine |
| sin | Sine |
| tan | Tangent |
| acos | Arccosine |
| asin | Arcsine |
| atan | Arctangent |
| cosh | Hyperbolic cosine |
| sinh | Hyperbolic sine |
| tanh | Hyperbolic tangent |
| sqrt | Square root |
| abs | Absolute value |
| sgn | Sign (-1, 0, or 1). |
| log, ln | Log base 10, natural logarithm |
| exp | Exponential base e |
| step | Unit step ( 0 for t < 0, 1 for t >= 0 ) |
| int | Integer value |
| rand(n) | Random number between 0 and n |
| limit | Limits the value +/-1. If the absolute value is greater than 1, then its magnitude is set to 1. |
| waveX(n) | Sound amplitude at n. X specifies the Sound window as given in the Source list. If no X is specified, the destination Sound window data is used. |
Several signal generation expressions are listed below. Words given in italics represent numeric values that you must enter. To try one of the following expression, perform the following steps:
sin(2*pi*261.7*t)
Use the play button to preview the expression before processing the entire file.
| Type | General Expression | Examples |
| Sine wave | sin(2*pi*frequency*t) | Middle C: sin(2*pi*261.7*t) Telephone dial tone for "5": (sin(2*pi*1336*t) + sin(2*pi*773*t)) / 2 |
| Saw wave | 1 - 2*abs(1 - 2*frequency*t%2) | 200Hz tone:
1 - 2*abs(1 - 2*200*t%2) |
| White noise | amplitude - rand(2*amplitude) | Full volume white noise:
1 - rand(2) |
| Square wave | int(2*t*frequency)%2*2-1 | 400Hz tone:
int(2*t*400)%2*2-1 |
| Sweep | sin(2*pi*t^rate) | Slow sweep up to 20kHz:
sin( 2*pi*160*(t%5)^3 ) |
| Exponential decay | (1 - minimum)*exp(-t) + minimum | 50% decay a 500Hz sine wave:
(0.5*exp(-t) + 0.5) * sin(2*pi*500*t) |
One way to create your own digital filter is to use MatlabTM (The Student Edition of Matlab, by The Math Works Inc., published by Prentice-Hall, ISBN 0-13-855974-0). It has many built-in commands that generate filter coefficients. The coefficients can then used in the Expression Evaluator command.
In preparation for down-sampling, you can use Matlab to generate the coefficients of a 4th order Butterworth low pass filter that will remove noise above half the Nyquist frequency (one quarter the sampling rate). Enter:
[b,a] = butter(4, 0.5)
The result should be similar to:
b = 0.0940 0.3759 0.5639 0.3759 0.0940 a = 1.0000 0.0000 0.4680 0.0000 0.0177
To implement this filter in the evaluator, assume that the sound to be filtered is in the Sound window titled Sound.wav.
wave1(n)*0.0940 + wave1(n-1)*0.3759 + wave1(n-2)*0.5639 + wave1(n-3)*0.3759 + wave1(n-4)*0.0940 - wave2(n-2)*0.4860 - wave2(n-4)*0.0177
Unless your turntable has a built-in amplifier or a headphone output, you'll need an external amplifier, receiver, or preamp to boost the turntable output. The figure below shows all the connections required. You may need two RCA cables with male connectors and a RCA to 1/8" miniplug stereo Y-adaptor cable.
If you have not connected your headphones or speakers to your computer, do so before proceeding.
Step 1
Connect the turntable output to Phono-in on a receiver or to the input
of a preamp, as shown.
Step 2
Connect the receiver or preamp audio output to the light blue
Line-In socket on your computer. It is marked with an arrow pointing
into the rings, as shown.
Make sure you connect to the correct socket on your computer.
If you accidentally plug the receiver/preamp output into the computer's
speaker socket you may damage the hardware. Connect your
speakers or headphones to your computer first.
Step 1
Take some time to thoroughly clean the album and carefully inspect and clean
the stylus.
Also review the information about avoiding Noise.
Eliminating noise before it is recorded saves a lot of restoration
work later and gives much better quality in the end.
Step 2
Run GoldWave, then press the F11 key or use the
Options | Control Properties
command, then select the Device tab. Make sure the correct
playback and recording devices are selected. Usually the default
settings are fine, unless you have more than one sound device.
Step 3
Select the Record tab on the Control Properties window, then check the
Monitor input box. This will activate
the visuals on the Control window so you can adjust volume levels
whithout having to record.
Step 4
Select the Volume tab on the Control Properties window. Make sure
the correct Volume device is selected. It should be the same as
the recording device.
Check the Select box for the Line (or Line In) item. Make sure the volume level is at least 50 initially. Make sure no other items are checked as they could be a source of noise. Choose OK to close the Control Properties window so that all new settings are used.
Step 5
Play the album on your turntable to ensure that all the connections
are good and that the input level is set correctly. The Level Visual
in the Control window should occasionally touch the red region. If you
find that the Level Visual goes too high or never goes above the green
region, then adjust the volume as explained in step 4. Note that
the volume fader on the Control window only changes the playback
volume. You must adjust the recording volume under the Volume
tab of the Control Properties window.
If you get no activity at all, check all connections, make sure your receiver/preamp is turned on, and make sure the correct recording device and volume source are selected as explained in steps 2 to 4. Check to see if your amplifier or preamp has an output level control that needs to be adjusted.
Step 6
Use the File | New command and
select one of the "CD Audio" preset quality setting. Enter the initial length
for the recording, such as 25:00 to record the entire side of the
album. Note that you can use the
Unbounded recording mode to
record for any duration.
Step 7
Click the red
record button or press Ctrl+F9 to start recording in
GoldWave. Start playing the thoroughly cleaned, dust free album on
the turntable. Press the
red
stop button to stop recording when done.
Step 8
Play the recording to see how it sounds. If you are not satisfied with
the quality, make volume adjustments, re-clean the album, check connections,
etc. Use Edit | Undo to undo the recording
and start recording again. If the quality is satisfactory, continue to
the next step.
Step 9
Trim any leading and trailing silences by selecting the entire file with the
Edit | Select All command,
followed by the Edit | AutoTrim command.
Depending on how noisy the recording is, a threshold of −30dB or greater may
be needed. Start with −40dB. If you still see flat areas on the ends of
the waveform, increase it to −30dB.
If the recording is very noisy, manual trimming may be required. To do that, use View | 10 Seconds. Right-click on the waveform to set the start selection marker to a position a second or so before the music begins. Click on the timeline to start playback to check the position. Scroll to the end of the recording by using the scroll bar below the waveform. Right-click to set the finish selection marker at a position a second or so after the music ends (where the waveform goes flat). Use Edit | Trim to remove the silences on either end of the selection.
Step 10
Save the recording by using the File | Save
command and providing a name for your recording.
Step 1
Use the Effect | Filter | Pop/Click
command to remove any pops and clicks from the recording. Use a
tolerance setting of 2000 at first. If you find that some pops/clicks
are still present, select a short
area of the recording where the click occurs then use the Pop/Click
filter again with a lower tolerance setting. Using a low tolerance
on the entire recording is not recommended since it may distort
some sounds, such as trumpet solos. Use
Edit | Select All to
select the entire recording when done.
Step 2 (optional)
Use the Effect | Filter | Smoother
command and select the "Reduce hiss" preset to reduce crackle and hiss.
Step 3
Play the recording to find a few seconds of silence where only
background noise can be heard. Select
about a second of that noise, then use
Edit | Copy to copy it to the clipboard.
Select the entire recording by using
Edit | Select All.
Use Effect | Filter | Noise Reduction
to display the Noise Reduction window. Choose the "Use clipboard" envelope
option. Preview the settings by choosing the play button to ensure the
quality sounds good. If
you notice too much tingling or warbling, then lower the Scale setting
and Apply the settings while previewing. Choose OK to remove the noise from
the recording.
Step 4 (optional)
Use the Effect | Compressor/Expander
command and select the "Noise Gate 3" preset, then choose OK. This will eliminate
any remaining noise in the silences between songs.
Step 5
Use Effect | Filter | Equalizer
to display the Equalizer window. Experiment with the presets and adjust
the bands to boost or reduce bass and treble. Preview the audio until
you get the desired results, then choose OK to process the recording.
Step 6
Use Effect | Volume | Maximize Volume
and select the "Full dynamic range" preset, then choose OK.
Step 7
Save the restored recording by using the File | Save
command.
Step 1
Use Tool | Cue Points
to display the Cue Points window.
Step 2
Choose the Auto Cue button. On the Auto Cue window, choose the Mark
Silence button. Set the threshold to −40dB. Set the silence length
to 1 second. You may want to use a longer or shorter time
depending on how long the silences are between each song. Set
the minimum separation to about 1 minute (1:00). Use a longer or
shorter time depending on the length of the shortest song. Choose
OK to automatically set cue points at the beginning of each
song.
If no cue points appear in the list, choose Auto Cue again and increase the threshold (−30dB). If too many cue points appear, choose the Delete All button and use a lower threshold (−45dB) in Auto Cue. Note that if the album is a live recording with no silences between songs, you'll have to set cue points manually. Close the Cue Points window, then Play the recording and use Edit | Marker | Drop Cue to drop cue points at places where you want to split the recording into separate tracks.
Once cue points have been added, the Split File button becomes enabled.
Step 3
Choose the Split File button on the Cue Points window.
Provide a destination folder where each song/track will be
saved. Check the "Use CD compatible wave format and alignment"
box. Choose OK to create a set of track files for each
song.
Step 4
Use CD-R software, such as Easy CD Creator, Nero, etc. to write
the track files to a blank CD as audio tracks. Note that you cannot
just copy the files to the CD using Windows Explorer. That will
just create a data track. Each song must be written as a separate
audio track.
End of Tutorial
| Problem | Cause/Solution |
| Cannot open large files | Make sure that hard disk storage is enabled in
Options | Storage.
Make sure that you have plenty of free RAM and hard drive space. CD quality sound requires 10MB per minute and 20MB per minute when editing. |
| Cannot play sounds | Sound driver is incorrectly
installed or needs to be updated.
Check that the Windows Sound Recorder accessory can play sounds. If it doesn't, the driver is not installed correctly. Make sure that your audio device is selected by using the Device Properties tab under Control Properties. |
| Cannot record sounds | See above.
Make sure you have the correct recording source selected. See Recording Sounds under Control Overview. Make sure your audio hardware is capable of recording. Sound may be in use by the playback device; click on the stop button. Recording device may be in use by another program; close the other program. |
| System seems slow or visuals do not update smoothly |
Your system may not be fast enough to draw the visuals. Try reducing the frame rate in Visual Properties tab under Control Properties or set all visuals to blank. |
| System freezes or crashes or an exception occurs | Make sure that you have a Pentium III or better system.
If the crash occurs during recording or playback, install an updated sound, video, and motherboard drivers. If the crash occurs when you try to run GoldWave, use the Start | Programs | GoldWave | GoldWave Setup to disable some of the plug-ins or OpenGL. You may have encountered a problem. If you can duplicate the problem after restarting your system, contact GoldWave Inc. for more assistance. |
| After a crash, there is less free space on the hard drive | Delete files in the temporary storage folder specified under Options | Storage. These files usually have names like GWXXX.TMP. Note that you may be able to recover the files by opening them in GoldWave and specifying the Raw type with "32 bit IEEE float, stereo (or mono), little endian" attributes. |
| Visuals are out of synchronization | Many audio drivers return inaccurate positions. Make sure you have the most recent sound driver. |
| Pops, clicks, or stuttering during playback |
Your sound card driver is rubbish. Get a new sound driver, or
a new sound card, or tell your sound card manufacturer about the problem.
Pops and clicks can occur at the beginning or ending of a sound if the first or last sample is not silence. Fading in/out a small selection can sometimes fix this. Enable the Edit | Marker | Snap to zero-crossing feature to avoid pops and clicks when editing. |
| Expression Evaluator slow | Make sure that RAM storage is selected in
Options | File. Remember to close and reopen
your sounds for this setting to apply.
Your crusty old 386 system does not have a co-processor. Go to a computer junk yard and find a Pentium system. |
| Editing seems to be getting slower and disk activity is increasing | Files on your hard disk are becoming fragmented. Use the Windows Disk Defragmenter system tool. |
| Distortion in recording | You need to lower the recording volume. See
Volume Properties tab under
Control Properties for
information about selecting and controlling recording volume.
Check that all connections are correct and firm (do not connect a "line-out" to "mic-in", for example). |
| Gaps in recording and playback | The Windows virtual memory manager can sometimes
cause gaps due to excessive swapping. Installing more RAM
may help.
Your system may be too slow to display the visuals. Try reducing the frame rate in the Visual Properties tab under Control Properties or set all visuals to blank. |
| Sound won't play for more than a few seconds. | Make sure you've selected the entire file. |
Why do I get only silence when I try to record?
You have to select the correct recording source. See Volume Properties tab under Control Properties.
How do I record from an audio CD?
If you have a SCSI or MMC compliant CD-ROM drive (most are), use the Tool | CD Reader command. If you want to record using your sound card, you need to select the CD-ROM as the recording source for your sound card. See Recording Sounds for more information.
How can I see and adjust recording volume levels without recording?
Use the Monitor input option under the Record Properties tab.
How can I avoid stuttering and glitches in recording?
Do not use disk compression. Set visuals to Blank. Close all other programs. Restart your system before (or between) recording for long periods of time. Update your sound driver.
How do I select part of a sound?
See Editing Overview for more information.
How can I tell if the finish marker is in the right place without playing the entire selection?
Right click on the waveform just before the finish marker and choose the "Play from here" menu item. Or, go to the Play Properties tab and select the Finish option for one of the play buttons. That will play a few seconds before the finish marker.
How do I edit individual samples with the mouse?
See Redrawing the Waveform with the Mouse.
Why does GoldWave show the shareware messages?
GoldWave is not free software. If you are a licensed user and are still seeing shareware messages, you'll need to enter your license into the program again. Directions for entering the license are included with the license.
Can I convert sound files to MIDI?
No. MIDI files do not contain digital audio. They contain notes and timing information for instruments. In other words, they contain instructions for playing the music, but not the music itself.
How are you?
Fine, thanks.
How can I split a large recording into individual songs?
Use the Cue Points tool with the Auto Cue Button and the Split File Button.