Copyright © 2023 GoldWave Inc.
Please do not publish, upload, or include this document on a website.
April 2023
GoldWave is a professional, full featured,
digital audio editor. Use it to play,
record,
import,
edit,
restore,
process, analyze,
and convert audio on your computer.
Updates and announcement are posted on the GoldWave website, Twitter, and Facebook.
Evaluation Usage Limit
If you are using an evalution version of GoldWave, the upper
right status bar displays a command count (unlicensed usage),
which gives you a rough idea of how much you've used the program. The
evaluation version is limited to 200 commands each session and 2500
commands total. When the session limit is reached, a reminder message
appears whenever you use a control in the
Control window. Exiting
and restarting GoldWave lets you use another 200 commands without
interruption. The program stops working when the total command
count is reached. Please
purchase a license
to remove evaluation limits.
You can give copies of the evaluation version of GoldWave to anyone
you think might find it useful. See
distribution information for details.
GoldWave includes a complete set of audio processing features.
Familiarity with using Windows, such as windows, dialogs, toolbars, scroll bars, etc., is recommended before reading this manual.
Familiarity with using apps, such as tapping, scrolling, 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: Installation, 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 items. Tap the ☰ button in the upper left corner to display the menu. File | New, for example, means to select the New item 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 item from the Windows Start menu instead.
Options, settings, features, commands, and paramaters are given in a fixed width font or in quotes.
A green arrow box provides one or more ways of accessing the feature in the program.
A information box emphasizes helpful information and techniques.
An exclamation mark box emphasizes warnings and other important information.
The following sections give instructions for installing and configuring GoldWave on your computer or device.
The minimum device requirements for GoldWave are:
The total amount of memory or hard drive space required depends on the size of audio files you'll be editing. One minute of CD-quality audio requires 20 megabytes of temporary storage. Editing a full CD requires a minimum of 1.5GB of storage and possible much more for Undo storage. By default, GoldWave uses memory for temporary storage. If you are working with large files, use Options | Storage to change the storage setting to "Hard drive" instead.
The following section gives instructions for installing GoldWave on your system. Before installing GoldWave make sure that audio playback and recording devices are installed and working in the Windows Control Panel.
Close any opened instances of GoldWave before installing an update.
To install GoldWave from a downloaded file, simply run the download. It prompts you to provide a destination folder to install the program files. A desktop shortcut and Windows Start menu items are created automatically, if selected.
New versions of GoldWave will be available from the web site goldwave.com
GoldWave supports a portable installation to run it from a USB drive on any computer without reinstallation.
To create a portable installation:
If GoldWave finds the settings file in the folder where it is installed, it loads and saves settings from that file and does not use the file on the computer, provided the file can be written to. Note that some plug-ins may still use the settings file on the computer or may not function correctly.
To choose an audio devices to use for playback
and recording,
use the properties
button on GoldWave's Control window or use
Options | Control Properties, then choose
the Device tab.
use Options | Playback or Options | Recording menu item.
Select appropriate devices from the drop down lists of installed playback and
recording devices.
Use Shared
quality unless you require using specific
hardware sampling rates.
Use the Test buttons to make sure you can hear playback and that
recording is working.
To associate a file type with GoldWave, such as wav or mp3 files:
Older versions of GoldWave may have to be uninstalled for the correct version to be used. See Windows help for more information about default programs and file type associations.
A license is an activation code that unlocks the evaluation version and removes all limits and reminder messages. A Lifetime license never expires and works in future versions, giving you free upgrades. A One Year license expires after one year, then a new license is required to continue using the program. Internet access is not required (or used) for activation and the license is not tied to a specific computer. All license information must be kept confidential. If a license becomes public, it will be revoked by GoldWave Inc. and will not be replaced.
To purchase a license, please see the website.
To enter a license into the program, use Register on the Options menu or use the Enter License button above the status bar near the bottom of the Main window .
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 toolbar buttons, and status bars (see Main Window figure below). It groups together and manages all the Sound windows.
Note: If this manual is printed, be sure to enable printing of background images.
The Main window contains the menu, toolbar, Control window, and Sound windows, as shown below.
The toolbar 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 when the mouse pointer is positioned directly over it. Right-click on Effect buttons to display a list of presets for that effect.
Use Toolbar in the Options menu to configure the toolbar.
Click-and-drag the left edge of a toolbar to move it. To reorder toolbars, right-click on the toolbar and choose Allow Toolbar Reordering.
The status bars show attributes of the Sound window, including the channels, length, selected region, playback position, modified status, zoom level, and general file format information. By clicking the mouse pointer over any status item with an inverted triangle on its right side, the unit or format for that status item can be changed. If you click the mouse pointer over the "Length" item, for example, a menu lists file lengths in terms of storage size, time, and samples. Clicking on the "Channel" item displays a channel selection window.
▲Channel | ▼Length | ▼Selection Range | ▲Playback Position | Coordinates |
Modified | ▼Zoom | Format Description |
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 white graph is the left channel and the bottom red graph is the right channel (see Channels for other colours). The selected part of the sound is highlighted with a blue background between two cyan markers. Initially the entire sound is selected. A vertical line with a left pointing triangle shows the current playback position within the sound. This line is the playback marker (or 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. White gradient edges indicate what part of the sound is currently displayed and zoomed.
Use the mouse buttons or the keyboard to change the selection. See Editing Overview for details. You can configure the window size and axes format of Sound windows using the Options | Window command and set the function of the left mouse button. The Options | Colours command sets the colour scheme.
The window is divided into three rows. The top third is the selection area. Touching in this area sets the selection. See Editing Overview for details. The middle third is the view and zoom area. Touching or pinching in this are scroll or zooms the waveform. See the View Menu Commands for details. The bottom third is the playback area. Touching in this area changes the playback position. You can configure some of the features of Sounds windows using the Options | Window on the menu. Use Options | Colours to change the colour scheme.
The Control window interacts with your sound hardware. 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.
The playback marker (or cursor) is a vertical line with a right pointing triangle that moves across the Sound window during playback.
Move the marker in the following ways:
To hide the marker when playback stops, check the setting in Options | Window.
Use Set Playback Position to see or set the playback marker's current position. The position can be absolute from the beginning of the file, or relative depending on mode setting. Enter a time to change the position.
Modes | |
---|---|
Mode | Description |
Absolute | The time entered is the absolute time from the beginning of the file. |
Add | The time entered is added to the current position. Use this to move the marker ahead a given amout of time. |
Subtract | The time entered is subtracted from the current position. Use this to move the marker back a given amount of time. |
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 drop-down list to reduce the load on the computer's processor to give more time to other programs. Use the notification drop-down list to set the audio notification played when processing is finished. Notifications are enabled only when processing takes more than 10 seconds.
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 menu items are explained below. The mouse wheel works only when the Main window is active and only on the currently active Sound window. Click-and-drag the middle mouse button to scroll the waveform left or right.
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.
Increases or decreases the playback volume by changing the Volume 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.
Time Formats | |
---|---|
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. |
XXXXXXXXsmp | Specifies time as a number of samples. This number is divided by the sound's sampling rate to calculate the actual time. This notation is supported only in New and Insert Silence. |
Time Examples | |
---|---|
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. |
24000smp | 24000 samples, which is half a second of audio at a sampling rate of 48000Hz. |
The Control window is the interface for playback and recording. On the bottom half of the window are visuals that 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. Use the expander button on the far right to reveal some of these and other controls. A status visual is located to the right of these controls. A level visual is located in between or below. Volume and speed controls are located in the next row.
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. Use the expander button on the far right to change the size of the window.
The desktop version of GoldWave refers to these settings as "properties". In the mobile version, they are referred to as "settings".
Use Control Properties to configure playback, recording, volumes, visuals, and devices. Settings are described in the following sections. After installing GoldWave, you should take a moment select the playback and recording devices under the Device tab and familiarize yourself with the settings under the Play and Record tabs.
Use Play Properties to configure the three play buttons, set the speed for rewind and fast forward, and set the marker preview time. Play button settings are given in the table.
Play Button Settings | ||
---|---|---|
Setting | Image | 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. |
Continue | ![]() |
Continues playback from the current playback position and stops at the finish marker. If the position is past the finish marker, playback begins at the start marker. |
Continue to end | ![]() |
Continues playback from the current playback 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. Be sure to check the Loop box below to enable looping. This is useful for musical accompaniment, looped instrument samples, or testing loop points. |
Loop point | ![]() |
Starts playback just before the finish marker and loops back to the start marker and continues playing briefly. Use this setting to test the current selection markers for smooth loop points. |
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. The loop image appears over the image of a button that has looping selected. |
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 audio.
Skip
Sets the amount of time to move the Playback marker
when skipping backward or forward.
See Control | Skip Backward for more information.
Marker preview (scrubbing)
Marker preview (scrubbing) plays a very short section of audio
whenever a selection marker is moved.
Preview duration
specifies the amount of time to play just after
the start marker or just before the finish marker. Setting the time to 0
disables previewing. Use the drop-down list to select when marker preview
is triggered. By default previewing is done only when the marker is moved
with the keyboard
arrow keys. Select the second item to enable previewing after mouse dragging
as well.
Record Properties contains all of the recording related settings and features. Use these to monitor the recording sources, start recording automatically when a sound is detected (level activated), delay recording until a certain time (timer), and more.
To change the recording source or device, use the Device tab.
Recording Settings | |
---|---|
Setting | Description |
Use new file duration |
Sets the duration for recording a new sound. See
Entering Times.
If this box is unchecked, then a window appears when you choose the Record New
![]() |
Dictation mode |
Allows you to quickly switch between recording and playback (Punch In/Out). The playback marker
is moved to the recording marker whenever recording is stopped. Recording starts at the playback marker instead
of the start marker. Use this feature to record dictation and rewind (or reposition the playback marker) and re-record
to fix mistakes. Use the Record Dictation ![]() ![]() This setting overrides the Recording Mode to use Unbounded recording and disables loop recording. New audio is recorded over the existing audio, replacing it. Note that recording is stopped without warning when playback is started. Do not use this mode for live recordings that should not be interrupted. |
Monitor input on visuals | Connects the recording source to the
visuals so you can adjust volume levels before recording.
Monitoring works only when the current sounds's sampling rate is
compatible with the recording device or no sounds are opened.
See Recording Sounds for information about selecting a different recording source and setting volumes. To hear what is being recorded, enable Windows recording monitoring. |
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 audio during recording. Use this setting 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 setting with
the Timer setting 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. If you start recording in an existing file, the original file will be overwritten when recording ends and recording cannot be undone. |
Power down system | Automatically turns off the computer after saving the recording. Use this setting with the Auto save and Timer settings to shutdown the computer after a scheduled recording. |
Record Mode | |
---|---|
Setting | Description |
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 setting to record for a fixed length of time. |
Bounded and looped | The is similar to the above setting, 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
Timer
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.
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 automatically synchronizing recording to a sound source or efficiently captures airport or police radio communications containing mostly silence that does not need to be recorded. It starts recording automatically when the sound source is above a given level and pauses recording when the sound is below the level.
Level Activated Settings | |
---|---|
Setting | Description |
Threshold | Specifies how loud a sound should be before recording begins. The value must be high enough so that noise does not trigger recording and low enough so that other sounds will. Start with a value around -20dB or record some background silence for several minutes to get a baseline and use Maximize Volume (Normalize) effect to get the peak level and set the threshold value above that. Be sure to keep the device recording volume the same. Any changes to that volume will affect the threshold (other Windows program may change the recording volume). |
Minimum duration | 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. |
Prebuffer | 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. |
Time stamp cues | Marks the date, time, and position of each recording activation with a cue point. Use the Cue Points tool to view and edit cue points. Use the edit box to specify the format for the date (this is done using the C strftime function). Some format specifiers are given in the next table and examples in the following table. |
Time Stamp Specifiers | |
---|---|
Specifier | Description |
%a | Short weekday (Sun, Mon, ... ) |
%A | Weekday (Sunday, Monday, ... ) |
%b | Short month (Jan, Feb, ... ) |
%B | Month (January, February, ... ) |
%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 |
Time Stamp Specifiers Examples | |
---|---|
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 |
Volume Properties is available only when using DirectSound mode. Use it to adjust recording volumes and select or unselect recording sources. Make sure the Volume device selected is the same as the recording device selected in the Device tab (it is not matched automatically if more than one recording device is available).
A volume fader, edit box, and checkbox are 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 setting is not checked and that the master volume is not zero.
You can use the Monitor input on visuals setting under the Record tab to activate the visuals without recording.
Note that volumes are changed instantly remain changed even if Cancel is used to close the Properties window.
To select a different recording device, use the Device tab.
Use Visual Properties to configure real-time visuals. Up to 10 visuals may be active, depending on the number of channes in the file and the settings selected. The default settings show a status visual, a level visual, and two graph visuals for the left and right channels. 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 shows the current output or input as horizontal bar meters. The left and right graph visuals display audio in a variety of ways, as described in the Description of Visuals table below.
Resize the Control window to make visuals larger or smaller.
Visual Settings | |
---|---|
Setting | Description |
Visuals | Sets the number of visuals to display. The 2 Mixed option combines multichannel into stereo. The center and low frequency channels are mixed in with the left and right channels. Usually each channel is graphed separately in each visual. Some visuals, such as the VU Meter or X-Y Graph may use more than one channel. |
Frame Rate | Sets the number of times per second that visuals are updated and drawn. A value of 60 or less gives good results. Use higher values to get an extra detailed spectrogram or envelope. The actual frame rate is limited by your system's processing power. Use a lower frame rate for older systems or when the Control window is large. |
Status, Level, Channel... |
Use these drop down lists to select a visual for the status
level, or channel graphs. Visuals are described in the
table below. The number of
visuals shown is set by the Visuals setting above.
Some visuals have properties such as axes ranges, colours, display modes, etc. Use the Properties button to the right of the drop down list to set the properties. |
Quick Select Menu | Use this 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. |
Visual FFT Settings | See FFT Settings for more information about these settings. Minimum dB sets the lowest dB level that visuals will show. |
Description of Visuals | |
---|---|
Visual | Description |
3D Bars | 3 dimensional logarithmic frequency 11 band bar graph. |
Analog Meter | Scaled amplitude needle meter. |
Bars | Logarithmic frequency 11 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. |
Bulge | Symmetrical, colourful frequency graph. |
Channel Separation | The difference between the left and right channels shown as a frequency graph. |
Envelope | Amplitude envelope. |
Music Staff | Pitch of the audio with music staff overlay. |
Ring | Coloured frequency graph radiating from low frequency at the center to high frequency outwards. |
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. |
VU Meter | Horizontal peak and current amplitude level meter. |
Waterfall | Flowing, coloured spectrogram. |
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. |
Bulge displays a symmetrical colour scaled frequency graph where higher magnitude frequencies are shown with taller lines and more intense colours. The magnitudes (heights) are drawn on a logarithmic scale.
Bulge Settings | |
---|---|
Setting | Description |
Colour scale | Sets the colour gradient for the graph. |
Linear, Logarithmic |
Sets the scale for the X axis. Use Linear to draw frequencies on a simple linear scale. Use Logarithmic to expand low end frequencies drawn in the center of the visual. |
Slow fade | Makes the graph fade out gradually while expanding it vertically. |
Channel Separation displays a colour scaled frequency graph representing the difference between the left and right channels. The greater the difference in a frequency, the taller the bars and the higher the colour intensity. The heights are drawn on a logarithmic scale and multiplied by 2.
This visual can be used see the difference between two signals if each were placed in a separate channel of a stereo sound.
Channel Separation Settings | |
---|---|
Setting | Description |
Colour scale | Sets the colour gradient for the graph. |
Linear, Logarithmic |
Sets the scale for the X axis. Use Linear to draw frequencies on a simple linear scale. Use Logarithmic to expand low end frequencies drawn in the center of the visual. |
Slow fade | Makes the graph fade out gradually. |
Select one of the listed items to change the colour scale used to display the visual.
Colour scales are used by 3D Bar, which displays a three dimensional frequency bar graph, by Bulge, which displays a double mirrored colour frequency graph, by Envelope, which displays an amplitude envelope graph, and by Ring, which displays frequencies as rings.
Colour Scale Settings | |
---|---|
Setting | Description |
Rainbow | Uses a full colour gradient. From lowest to highest the colours are: black, purple, blue, green, yellow, orange, red, white. |
Cold | Uses a white/blue gradient from gray to light blue. |
Hot | Uses a red/orange gradient from dark red to light orange. |
Gray | Uses a gray-scale gradient from dary gray to white. |
Solid white | Uses a solid white colour with no gradient. |
Solid cyan | Uses a solid cyan colour with no gradient. |
Discrete rainbow | Uses solid colours for different magnitude or amplitude levels. Changes from one level to the next are more distinct and easier to notice. |
Blue | Uses a blue gradient from dark blue to light blue. |
Use Music Staff to transcribe music. It displays peak pitch information over time on a grand staff. The horizontal axis is time, the vertical axis is pitch (notes), and the colour intensity represents volume of the pitch, explained in Mode below.
This visual works best with simple, monotonic audio. More complex audio will be more difficult to read and it will take time to learn how to interpret the graph. Due to the logarithmic scale of the music staff and the linear scale of the FFT, higher pitches are resolved at much higher resolution than lower ones. Setting FFT Size higher (13) may improve accuracy for low pitch tones. Use Pitch to raise the music by one octave to improve the resolution.
Use Time to stretch the music. Use filters to remove noises or other sounds or to boost a frequency range to make the visual clearer.
Use the Speed Fader to adjust the overall key of the music to shift the spectrum up or down to more closely align the grid/sharpes.
Music Staff Settings | |
---|---|
Setting | Description |
BPM | Sets the measure in beats-per-minute. A vertical line is draw at this interval. |
Divisions | Set the number of divisions within the beats-per-minute measure. Darker lines are drawn at this sub-interval. |
Range To |
Limits the frequency/pitch range to search. Any pitches outside the range are not shown. Use this setting to narrow down the range for vocals or a specific instrument. |
Threshold | Sets the minimum volume for the pitch to be detected. Any pitches with a volume below this level are not shown. |
Mode |
Sets the graph mode.
|
Ring displays a colour scaled frequency as rings with the low frequencies in the center and higher frequencies going outward.
Ring Settings | |
---|---|
Setting | Description |
Colour scale | Sets the colour gradient for the graph. |
Linear, Logarithmic |
Sets the scale for the radial axis. Use Linear to draw frequencies on a simple linear scale. Use Logarithmic to expand low end frequencies drawn in the center of the visual. |
Shape |
Sets the shape of the graph.
|
Expand | Makes the graph expand and fade outward from the center. |
Spectrogram displays frequency information over time. The horizontal axis is time in seconds (s), the vertical axis is frequency in Hertz (Hz), and the colour represents the frequency's magnitude (dB). The louder a certain frequency is, the more intense its colour.
Spectrogram Settings | |
---|---|
Setting | Description |
Automatic full frequency range | Automatically sets the frequency (vertical) axis range to match the current sampling rate of the file. For a sampling rate of 44100Hz the range is set from 0 to 22050Hz (the Nyquist rate). |
Fixed frequency range | Allows the frequency (vertical) axis range to be set manually, giving more frequency detail within that range. |
From (Hz), To (Hz) |
Sets the frequency range when Fixed frequency range is selected. |
Scroll speed | Sets the number of vertical strips to draw per frame, increasing the time resolution of the spectrogram. |
Show axis | Shows numbers on the horizontal and vertical axes and the colour scale legend at the bottom of the visual. |
Gray-scale | Uses a gray-scale gradient instead of colours. |
Spectrum displays a simple frequency graph. The horizontal axis is frequency in Hertz (Hz) and the vertical axis is magnitude in decibels (dB).
Spectrum Settings | |
---|---|
Setting | Description |
Automatic full frequency range | Automatically sets the frequency (horizontal) axis range to match the current sampling rate of the file. For a sampling rate of 44100Hz the range is set from 0 to 22050Hz (the Nyquist rate). |
Fixed frequency range | Allows the frequency (horizontal) axis range to be set manually, giving more frequency detail within that range. |
From (Hz), To (Hz) |
Sets the frequency range when Fixed frequency range is selected. |
Show axis | Shows numbers on the horizontal and vertical axes. |
Logarithmic | Changes the frequency (horizontal) axis from linear to logarithmic. |
Solid | Fills the area under the spectrum graph with a solid colour. |
Peak line | Displays a red peak graph showing the highest magnitudes since playback or recording started. Use the stop button to reset the peak line the next playback or recording is started. |
Average line, alpha |
Displays a yellow average graph showing an approximation of the average magnitudes
since playback or recording started. It is not a true average, but a low pass filtered
difference of magnitudes in time as follows:
M = alpha * Mprevious + (1 - alpha) * Mnew Use alpha to change the rate at which the average changes. A value of 0.999 causes the average to change very slowly. A value of 0.800 causes it to change quickly. Use the stop button to reset the average line the next playback or recording is started. |
Status displays elapsed time and playback and recording status.
Status Setting | |
---|---|
Setting | Description |
Display | Changes the elapsed time format to show hours, minutes, or seconds. |
Stripe displays colour scaled verticle stripes of increasing frequency from left to right and/or amplitudes from top to bottom in time, with the colour proportional to the magnitude.
Ring Settings | |
---|---|
Setting | Description |
Colour scale | Sets the colour gradient for the graph. |
Bands | Sets the number of vertical stripes to graph. |
Layer | Sets the graph layer to show. |
Tone Meter plays a tone and displays a solid colour based on the peak amplitude of the waveform. This visual is designed for visually impared users to provide an auditory amplitude meter. Shared playback quality must be selected to allow the visual to generate tones.
The properties include settings for four tiered tones. Each tone has a different trigger level, frequency, waveform, and colour. The Level value sets the upper limit to trigger the tone. The tone is used only if the peak amplitude is below or equal to the given level. If the peak is above that level, then a tone with a higher level is used. Each level must be higher than the previous level, so the trigger for level 2 must be greater than the trigger for level 1, etc.
Tones are played at a fixed interval. Tone interval changes the time between tones. By default a tone sounds once a second. Duration sets how long the tone plays. Volume changes the volume of the tone.
Sound tone only when Scroll Lock key is on provides a quick way to turn on and off the tone by using the Scroll Lock key on the keyboard as a toggle switch.
VU Meter displays the current peak volume of the waveform on a horizontal bar with a green to red gradient.
VU Meter Settings | |
---|---|
Settings | Description |
Decay time | Sets the amount of time it take for the meter to drop from peak maximum volume to nothing (silence). |
Peak hold time | Sets the amount of time the peak indicators (the vertical segments that stay on at the top level of the meter) remain at their peak positions before dropping back. |
Show axis | Shows decibel numbering on the meter. |
Reset Clip | Clears the red clip detection indicators on the far right of the meter. Clip indicators are automatically reset when playback or recording is restarted. They can be reset by clicking the mouse on them as well. |
Waterfall displays a 3 dimensional color frequency graph. The horizontal axis is frequency in Hertz (Hz), the vertical axis is magnitude in decibels (dB). The upward scrolling of the graph is time.
Waterfall Settings | |
---|---|
Setting | Description |
Automatic full frequency range | Automatically sets the frequency (vertical) axis range to match the current sampling rate of the file. For a sampling rate of 44100Hz the range is set from 0 to 22050Hz (the Nyquist rate). |
Fixed frequency range | Allows the frequency (vertical) axis range to be set manually, giving more frequency detail within that range. |
From (Hz), To (Hz) |
Sets the frequency range when Fixed frequency range is selected. |
Axis | Shows numbers on the horizontal axis. |
Grid | Draws grid lines on the graph. |
Fade | Fades the graph as it scrolls to the back. |
Logarithmic | Changes the frequency (horizontal) axis from linear to logarithmic. |
1 second dark transparent walls | Draws a dim transparent rectangle every second, dividing the graph into 1 second sections. |
Height (%) | Changes the vertical height of the graph with respect to the height of the visual's window. A value of 50% makes the maximum, peak magnitude height half as high as the window. |
Scroll angle | Change the vertical scrolling angle. 90° scrolls straight up. |
Waveform displays an amplitude versus time graph. X-Y Graph displays a left channel versus right channel graph.
Waveform Settings | |
---|---|
Setting | Description |
Monochrome line | Draws the graph in one colour with lines connecting the points. |
Coloured points | Draws the graph using coloured points in a green to red scale. The greater the amplitude, the more red the point. |
Swap X and Y | Switches the horizontal and vertical axes. |
Slow fade | Slowly fades from one graph frame to the next. |
Grid | Shows grid lines over the graph. |
Device Properties contains settings for playback recording, and joystick or pedal devices.
Playback and Record areas show the currently selected playback and recording devices. If more than one device is installed, you can select a different device from the drop down list. You can change playback and recording quality by selecting different bit depths from the Quality lists. Use PCM 16 bit quality unless your sound card supports higher bit depths. GoldWave takes exclusive control of the audio device unless you select Shared quality. The Shared setting forces GoldWave to share the audio device with other programs using the sampling rate, channels, and resolution determined by the system. The system attributes are shown next to the "Quality" label. Use the Configure button on the System tab to change the system properties of an audio device. Shared quality is not ideal for recording because it may not allow recording at the sampling rate of the file, resulting in resampling.
If you select a LOOPBACK device for recording, select Shared quality for best driver compatibility. See Recording Streaming Audio for more information about recording what you hear or Internet streams. Be sure to disable any sound driver enhancement features to ensure you get a clean copy of the audio.
Use the Test buttons to perform a simple test of a device. When testing a playback device, you should hear audio on the speakers or headphones. If not, check the Windows volume settings or select a different playback device. When testing a recording device, you can determine the supported sampling rate and quality and adjust the volume level.
The Playback area has additional settings for latency and initialization. Latency controls the amount of audio stored 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 drivers are used. This setting does not apply to recording.
Alternative initialization solves problems with certain drivers and plug-ins. Use this setting if GoldWave freezes when previewing an effect plug-in or if playback does not start properly in general. It begins playback on a separate processor without pausing or holding up the main program. This makes the interface seem more responsive when starting playback.
Muting controls how unselected channels are handled. By default, unselected channels are not played. Use this setting to always play all channels or to play one selected (solo) channel in stereo.
The Record area has a volume fader and mono settings. The Volume fader (not available in DirectSound mode, use Volume tab instead) adjusts the volume level for the device. You can adjust the volume anytime during recording.
The Attributes setting determines the sampling rate and number of channels used when starting a recording
with the Record New button .
Attributes Settings | |
---|---|
Setting | Description |
Use device attributes | The sampling rate and number of channels is determined by the device. Use this setting to avoid any conversions when recording. The audio data is captured directly from the device without further processing. |
Use new file attributes | The sampling rate and number of channels is determined by the values last entered into the New command. Use this setting to always record mono or stereo or at a specific sampling rate when a recording device supports multiple channels or rates. Audio data is processed (resampled or mixed) to make it compatible with the attributes. |
Joystick/pedal 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.
Joystick/Pedal Control | |
---|---|
Mode | Description |
None | Joystick/pedal control is disabled. |
Foot pedal or buttons | A foot pedal or game controller buttons controls playback. Use this for controlling playback in GoldWave while typing transcription in another program. If the pedal has more than one button, they can be assigned to rewind, fast forward, etc. by using the Configure button. GoldWave supports most USB HID devices with simple button inputs, such as the VEC Infinity IN-USB-2 foot controller and most USB HID "joystick" or "programmable" foot switches, such as Delcom or vPedal. |
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. |
Use the Configure button to configure foot pedal controls. Different brands of foot pedals use different switch combinations. After choosing the Configure button, choose an action button to assign, then hold down the pedal for at least one second.
See Also: Default Save Format, Recording Streaming Audio
Configure button in the
Device tab via Options | Control Properties
or via the Properties button in the Control
window. "Foot pedal or buttons (HID)" must be selected for Joystick/pedal.
Use this window to assign the pedals or buttons to certain actions like playback, rewind, etc.
To assign a pedal, choose the action button to assign, such as Play, then hold the pedal to use for that action for at least 1 second. The pedal indicators turn red during this time, then turn green when the pedal is assigned. If you do not press a pedal within 5 seconds or you choose the action button again, then no pedal is assigned. More than one pedal can be pressed at a time to assign that combination of pedals to an action. If two actions use the same pedal or same combination, then precedence is determined by the order shown (the lower action is considered unassigned).
Foot Pedal Options | |
---|---|
Option | Description |
Toggle | Pressing and releasing the pedal starts the action. The action continues until the pedal is pressed and released again. The pedel does not have to be held the for the action to continue (reduces constant foot pressure and fatigue). |
Hold | The pedal must be pressed continuously to perform the action. When the pedal is released, the action stops. |
Hold & skip back | The pedal must be pressed continuously to perform the action. When the pedal is released, the playback marker is move back two seconds. Tapping the pedal skips back 2 seconds each time. Use this option for a single switch pedal. Only the Play action has this option. |
The Skip actions skip backward or forward 2 seconds with each tap of the pedal.
See Also: Control Overview, Control Device Properties
Use System Properties to change the audio interface used for playback and recording and list information about the system. This information can help locate problems with drivers, hardware, or the current setup.
GoldWave supports DirectSound and Core Audio/WASAPI. Core Audio/WASAPI is selected by default, but DirectSound may be used instead if playback or recording probles occur. DirectSound was discontinued in Windows Vista and no longer provides direct access to the audio hardware, so it is not recommended. Using WASAPI in exclusive mode is the only way to play or record audio directly through the sound hardware at the original sampling rate and quality of the file.
Choose the Configure button to display the system's audio configuration window.
Choose the Information button to gather information about installed playback and recording devices.
If a device is listed as disabled, disconnected, or not present:
After opening a sound (see File | Open), use one of the
play buttons, such as Play All
,
to play it. Each button starts playback at a different place, which can be configured
under the Play tab of the Control Properties window.
Right-click on one of the play buttons to quickly change the 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 moving vertical line on the waveform (playback marker). You can move the start and finish selection markers to the playback position by using the bracket keys, [ and ] or Edit | Selection | Move Start.... 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 | Add Cue Point command.
If you do not hear anything during playback, check the following:
If an error occurs during playback, make sure the correct playback device selected and the audio hardware is capable of playing at the sampling rate and quality of the file.
Try the following:
If nothing helps, try playing a file in Windows Media Player to make sure playback is working on your computer.
While a sound is playing, 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. 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. When one of these buttons is used to start playback, the region played is
determined by the Play 3 setting.
Most computers have more than one recording input, such as microphone or line-in. To select and adjust a recording input, use the Device tab (and the Volume tab for DirectSound mode) of the Control Properties window. To record what you hear (such as an Internet stream), see Recording Streaming Audio. Make all connections before running GoldWave. Otherwise some devices or sources may not be listed. To see the input before recording, use the Monitor input on visuals setting under the Record tab of the Control Properties window.
Use the Record New
button to create a new file and start recording. The sampling rate and number of channels is determined by the
Attributes setting in Device Properties.
Recording stops automatically when the duration has passed.
If you stop recording earlier, the new file is trimmed to the length of the recording.
Use the Record tab of
the Control Properties window to set the default new file duration.
Use the Record Selection
button to record into an existing sound. 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 and the unrecorded part of the selection is filled with silence.
To record dictation and easily switch between playback and recording, check the Dictation mode
setting on the Record tab of the Control Properties window.
You can then use the Record Dictation button
to resume recording from the current playback position.
Make sure you see activity on the horizontal VU Meters while recording. The source volume should be adjusted so that it peaks in the orange or low red area, but not all the way. If there is no activity or the level is very low, change the input and/or volume or select a different recording device, explained below.
You can make room for recording in the current sound by using the Edit | Insert Silence command.
A recording pause
buttons appears in place of the record button so
that you can pause and unpause recording.
Many recording settings are available in the Record tab of the Control Properties window. Right-click on the record button to quickly access some of these settings.
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. After recording, mix the two files together.
For long recordings, turn off any power management settings that may power down or sleep the computer.
Software Loopback Method
Select a LOOPBACK recording device
in GoldWave. GoldWave lists a separate LOOPBACK recording device for every
compatible playback device in the system. If no LOOPBACK devices are listed,
be sure to select the Core Audio (WASAPI) mode on the
System Properties tab.
The following conditions:
Hardware Loopback Method
In DirectSound mode, the streaming source is
usually called "What You Hear", "Stereo Mix", "Wave Out", or something similar
and that input can be selected on the
Volume tab of
the Control Properties window in GoldWave.
In Core Audio/WASAPI mode, the streaming device is disabled by default (if present at all) and has to be manually enabled. Try using Software Loopback (above) before enabling the device.
To enabled a disabled device:
After the device has been enabled, it can selected for recording on the Device tab of the Control Properties window
A recording input may be physically connected to the output, so check the following as well.
Loopback Cable Method
If neither software loopback nor hardware loopback work, then a loopback or
splitter cable from the speaker output to the line input is required.
After connecting the cable, select the Line recording device in GoldWave. Make sure monitoring (above) is turned off to avoid feedback and echoes.
If an error occurs when recording, make sure the correct recording device is selected in GoldWave and the audio hardware is capable of recording at the sampling rate and quality selected.
Try the following:
If your recording device does not support a sampling rate required, record at a supported rate, then use Resample to convert it to the required rate.
If nothing helps, try recording in the Windows Sound Recorder accessory included with Windows (see Windows Help for details) to make sure recording is working on your 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 bottom speed fader changes the playback speed of the audio device. Move the fader right to increase the speed or 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. To change the speed of the file, use the Time effect instead.
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. The function of the left mouse button can be set using Window options. 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:
A selection can be dragged-and-dropped to a different part of the same sound or into a different sound. Use the dragging icons located at the top of the selection to start dragging.
The Range icon drags the selection range, allowing
you to maintain the same selection duration without moving the start and finish markers separately. The sound
within the selection is not affected. Only the selection markers are moved. You can drag the range to another
Sound window to select the same duration in that sound.
The Contents icon drags the selection contents,
allow you to move that section of audio to a different time or into a different Sound window. Note that when the
selection is dragged to another Sound window, the contents are copied and the selection is not deleted from the
original Sound window.
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 Zoom 1:1 or 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.
Use File Merger to join many files together.
Use Split File to divide a large file into smaller section.
The Edit | Mix command mixes one sound with another so they both play at the same time.
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. One of the songs must be copied to the clipboard before using the command.
Song 1 | Fade out | |
Fade in | Song 2 | |
← Duration → |
In some cases more control is needed.
For extra control, use the Shape Volume to create custom fades before mixing.
GoldWave supports both hard drive based editing and memory (RAM) based editing. These features are described below. Memory storage is enabled by default. Use 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 one or more temporary files 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 memory 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 memory (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 memory. If you edit or record large files, Windows may start swapping memory to the hard drive, which significantly degrades performance and may cause defects when recording.
If 90% or more of physical memory is in use or if the amount of storage required by the file is more than 75% of the amount of physical memory currently available, then Hard Drive storage is used instead to preserve system stability.
Also note that in the event of a system crash, it will not be possible to recover a file stored in memory.
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 installed file format plug-ins. 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, MPEG1 Layer 3, or iTunes compatible M4A.
The table indicates supported features within the file type: cue points (Cues), file information such artist, title, and other metadata (Info), Surround Sound (SrndS), and file sizes larger than 4GB.
Supported File Types | |||||
---|---|---|---|---|---|
Extension | Comments | Cues | Info | SrndS | >4GB |
.aiff .aifc .afc |
Apple / Macintosh sound files. Compressed files are not supported. Cue points are supported. File text information is supported in ANSI only. NAME, COPY, ANNO, AUTH, genr, ©url, ©trk, ©day, and ©art are preserved. | ![]() |
![]() ANSI |
![]() |
![]() |
.ac3 | AC3 compressed files. GoldWave does not support these files directly, but it may open them automatically using system decoders. If not, installing the AC3Filter for DirectShow (search Google) may work. | ![]() |
![]() |
![]() |
![]() |
.ape | Monkey's Audio compressed files. Requires the APEFile plug-in. | ![]() |
![]() ANSI ID3v1 |
![]() |
![]() |
.asf .avi |
Microsoft audio and/or video files. GoldWave can extract the audio portions of these files, but cannot save or create them. See .wma and .wmv below. | ![]() |
![]() .asf only |
![]() |
![]() |
.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. | ![]() |
![]() |
![]() |
![]() |
.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. | ![]() |
![]() UTF8 |
![]() |
![]() |
.iff | Amiga 8SVX files. NAME, COPY, ANNO, AUTH, and CHAN are all preserved. Limited support for file text information. | ![]() |
![]() some |
![]() |
![]() |
.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. GoldWave may use system decoders or QuickTime decoders to extract the audio portion from the file (if present). See the .mp4 and .m4a types for more information. Files cannot be saved in this format. | ![]() |
![]() |
![]() |
![]() |
.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 using unicode (UTF-16). The ID3v1 tag is read, but not written. | ![]() |
![]() |
![]() |
![]() |
.m4p | Copy protected/encrypted iTune/MPEG4 sound files. These files cannot be opened in GoldWave. Upgrade them to iTunes Plus to remove the copy protection/encryption. Upgraded files become .m4a files, which can be opened in GoldWave. | ![]() |
![]() |
![]() |
![]() |
.mp4 .m4a .aac |
Unencrypted iTune/MPEG4/AAC sound files. GoldWave opens these files through the Media Foundation system decoders and allows saving if the AAC encoder is installed. File information is preserved. | ![]() |
![]() ANSI |
![]() |
![]() |
.opus | Opus compressed sound files. Gives better quality compression than MP3 and covers a wide range of audio applications. Files are always encoded at 48,000 Hz. Refer to the Opus website for more information. File text information is supported. Opus uses a non-standard channel layout for 4 or 5 channels. See the vorbis channel order (used for Opus) for details. | ![]() |
![]() UTF8 |
![]() 5.1, 7.1 |
![]() |
.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. See the vorbis channel order for details. | ![]() |
![]() UTF8 |
![]() 5.1, 7.1 |
![]() |
.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 (but may be +/-10.0) for floating point data and -32768 to +32767 for integer data. No support for file text information. | ![]() |
![]() |
![]() |
![]() |
.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. | ![]() |
![]() |
![]() |
![]() |
.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 fact, LIST INFO, LIST adtl, and cue are detected. All others are ignored. Cue points are supported. File text information is supported in ANSI only. |
![]() ANSI |
![]() ANSI |
![]() |
![]() |
.wma | Windows Media Audio files. Supports several different bitrates from low bandwidth to high quality lossless. File text information and cue points are supported. | ![]() name only |
![]() |
![]() |
![]() |
.wmv | Windows Media Video files. The first audio track is extracted from the video file. File text information and cue points are read. Files cannot be saved in this format. | ![]() |
![]() |
![]() |
![]() |
.wv | WavPack compressed files. GoldWave does not support these directly. Installing the DirectShow Filter from the WavPack website may allow GoldWave to open and convert them. | ![]() |
![]() |
![]() |
![]() |
.xac | Extended Audio Container. Currently used only by GoldWave. | ![]() UTF8 |
![]() UTF8 |
![]() |
![]() |
![]() |
![]() |
![]() |
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.
If this window appears when opening MP3 files or iTunes, then the decoders or plug-ins required to open the files are not installed on the computer. GoldWave cannot open the file properly unless those are installed.
File Format Settings | |||||||||||||
---|---|---|---|---|---|---|---|---|---|---|---|---|---|
Setting | Description | ||||||||||||
File | Sets the file type or plug-in to use to open the file. | ||||||||||||
Attributes |
Specifies the structure, range, and layout of the audio data supported
by the file type or plug-in. See
the Attributes section for details.
Common format attributes are listed below.
|
||||||||||||
Rate | Sets the sampling rate for the audio data. This value does not affect how the data is translated. If the wrong rate is selected, the sound will either play too slow or too fast. Use Playback Rate to change the rate later. A CD audio recording has a sampling rate of 44100Hz. A Dialogic VOX or telephony file usually has a rate of 6000Hz or 8000Hz. | ||||||||||||
Data | Displays data at the beginning of the file in hexadecimal and ASCII form. This is useful only if you need to examine the raw contents of the file to identify its format. |
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 Playback Rate.
If the file is saved later, use a different filename and type using Save As so that GoldWave will be able to open the file next time. Or to make GoldWave assume a format for a particular file type extension, use the Undetectable Types tab in Options | File Formats to associate a format with the extension.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.
GoldWave checks for new plug-ins only during startup, so if a new plug-in is installed, you must restart GoldWave for it to be detected.
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:
Use these settings to set MP3 related attributes directly.
MP3 Settings | |
---|---|
Setting | Description |
VBR quality |
Uses a variable bitrate (VBR) when encoding the file.
Use Bitrate range to set the minimum and maximum
bitrates specified. The range of available bitrates depends
on the sampling rate. Selecting a lower or higher sampling
rate lists lower and higher ranges of available bitrates.
Set it to Off to use a constant birate. |
Channels | Sets the number of channels in the file and how stereo encoding is handled. Use Mono to create a single channel file. The other options create a stereo file. Use Stereo to create a typical stereo file. Use Joint Stereo to get better compression when the left and right channels contain similar audio. Use Dual Channnel when the left and right channels contain completely different audio. |
MPEG bits | Sets bits in the MPEG header. Include CRC tells the encoder to include CRC data after the header. Copyright means the audio is copyrighted. Original means the audio is new or original material. |
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 Change Volume changes the "size" of a sound. Changing the colour of a font would be similar to changing the pitch of a sound using Pitch
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 Change Volume 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 Time. 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. Maximize Volume 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.
The Effect Chain Editor tool allows several simple effects to be chained together and processed as a single unit for faster processing.
Batch Processing allows any number of edits and effects to be applied many files.
Many effects have similar controls such as presets and shape boxes. These are explained in detail in the following sections.
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.
Most effects have a Default preset, which can be changed so that the same settings are used every time the effect window is shown. See Options | Window for a setting to automatically update the Default whenever an effect is used.
To add a new preset:
To delete a preset:
To change a preset:
If you change any of the presets installed by GoldWave, including the Default preset, they will be reset to their original settings if you reinstall or update GoldWave. Use a different preset name to retain a preset across updates.
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. To move a point only horizontally or vertically, hold the Ctrl key
when clicking and dragging a point. Dragging is confined to the first direction dragged.
If you initially move the point horizontally, then vertical movement is locked.
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.
Shape Controls | |
---|---|
Control | Description |
Graph Window | Displays the current shape and graph. Click the mouse in the window to add points. Right-click to remove a point. |
Point | Sets the current point and associates X, Y, and Remove Point to that point. |
X | Sets the X value for the current point. |
Y | Sets the Y value for the current point. |
Add Point | Adds a new point between the current point and the next point. |
Remove Point | Removes the current point. |
Time | Adjusts the time within the selection for the graph. May set the preview start time. |
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 the Play button is pressed, 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.
Previewing can be paused by holding the Ctrl key when pressing the Stop button or with F7.
In most cases, any changes in the effect settings while previewing are applied in real-time so you can hear the changes as they are made. For more complex effects and most shape based effects, the Update button must be used to update previewing with the changes.
The Update button is used only for previewing. It does not apply the effect to the sound. Use the OK button to process the file.
When performing Fast Fourier Transform (FFT) processing, the sound is divided into small blocks and processed one block at a time. The FFT size value controls the size of the these blocks. The number of samples to process is calculated by taking the value as a power of 2. A value of 10 gives 2 to the 10th power, or 1024 samples. By increasing the number of samples, frequencies are processed at a higher resolution, which helps to eliminate chirping and other mechanical sound distortions, but it tends to add more echo. Usually values from 11 to 12 give the best trade-off between distortions and echo.
To smooth out transitions from one block to the next, it is necessary to overlap blocks. The Overlap value controls how much of the FFT analysis of one block overlaps the next. A high value makes the transition between each block smoother. It also requires more processing time since overlapping samples are recalculated several times. A low value may result in rougher transitions, but processes faster. For complex audio and tempo or pitch modifying effects, higher values may give better quality.
FFT Settings | |
---|---|
Setting | Description |
FFT Size | Controls the number of samples to process in an FFT analysis block (power of 2). |
Overlap | Controls the amount of overlap for each FFT analysis block. Usually set to 4x (75% overlap). |
Effect plug-ins are modules developed by other companies that can be used within GoldWave. These appear under the Effect | Plug-in menu. DirectX and VST plug-in wrappers are available for GoldWave to add support for many of the existing DirectX and VST Audio Plug-ins. Other plug-ins are designed to work with GoldWave directly and will appear as separate items under the plug-in menu, each with its own submenu of effects. In some cases settings for effect plug-ins can be changed using the Options | Plug-in | Effect menu.
If any errors or exceptions occur while using a plug-in, you'll need to contact the plug-in creator for assistance. GoldWave Inc. cannot provide support for plug-ins created by separate (third party) developers.
This section outlines the accessibility features in GoldWave. Almost all of GoldWave's functionality is accessible through the keyboard. See Keyboard Commands and Options | Keyboard.
If a screen reader is active when starting GoldWave, the program automatically changes its interface slightly to be more screen reader friendly.
Screen reader mode can be switched on manually by using GoldWave Setup in the Windows Start menu and checking the Force screen reader mode box.
The status visual in the Control window becomes a text box that gives the elapsed time, the current status, and the left and right peak levels (in parentheses). The levels are reset whenever playback or recording is restarted. 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. Press R to reset the peaks. Press P to pause updates to give time for the screen reader to read it. These keys only work when the status visual has the keyboard focus and the keys have not been assigned to other functions in GoldWave.
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 Toolbar images in menu box under Options | Toolbar.
Also when a screen reader is active, GoldWave defaults to shared playback instead of exclusive playback so that the screen reader can share the audio device with GoldWave.
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 certain parts of a sound file is done through keyboard navigation, which involves playing the file and moving the playback marker 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 marker is moved depends on the zoom level, explained in the View section below.
The View | Follow Playback menu item must be checked for keyboard navigation to work.
Keyboard Navigation | |
---|---|
Keystroke | Action |
Space | Starts playback or stop it (toggles playback). The region that is played depends on the Play 3 button settings. |
J, K, L | Rewinds, plays, and fast forwards respectively. Playback always starts at the playback marker position. |
Shift+J, Shift+K, Shift+L | Makes the playback speed slower, normal, and faster respectively. |
F2, F3, F4, F5, F6, F7, F8 | Plays 1, plays 2, plays 3, rewinds, fast forwards, pauses, and stops respectively. See Play Properties for the play button settings. |
Ctrl+G | Sets the playback marker to a specific time (go to). |
H | Starts playback relative to the mouse's horizontal position in the waveform. |
Shift+[ | Plays some audio up to the start marker. Marker preview controls the amount of audio played or three seconds is played if that value is set to zero. |
Shift+] | Plays three seconds of audio up to the finish marker. Marker preview controls the amount of audio played or three seconds is played if that value is set to zero. |
Ctrl+[ | Plays from the start marker to the finish marker. |
Ctrl+] | Plays from the finish marker to the end. |
Left | Move the playback marker backward. |
Right | Move the playback marker forward. |
Page Up | Moves the playback marker one screen backward. |
Page Down | Moves the playback marker one screen forward. |
Home | Moves the playback marker to the start marker's position. |
End | Moves the playback marker to the finish marker's position. |
Ctrl+Home | Moves the playback marker to the beginning of the sound. |
Ctrl+End | Moves the playback marker to the end of the sound (playback stops). |
In addition to using the Edit | Selection | Set and the Edit| Selection | Move... 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.
Keyboard Selection | |
---|---|
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) | Moves the start marker to the playback marker. |
] (right bracket) | Moves the finish marker to the playback marker. |
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 Trim Both 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 marker and selection markers are moved with each keystroke. The playback marker 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 playback marker 10 seconds backward or forward. The Left or Right key moves the playback marker 1 second backward or forward. The selection keys move the start or finish marker a tenth of a second (0.1s or one hundredth of the view). 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 (see Entering Times). You can alter the length later with the Edit | Trim menu or Insert Silence.
To change the default save format shown in the status bar, see File Formats. To change the recording bit depth, see Device Properties
Open presents a list of files in your sound folder. The sound folder can be set using the Storage in the Options menu. 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.
Use Open URL to enter the location of a remote file on a website to open. An active network connection is required. The file must have a finite duration. If a stream is opened, the program can never finish opening the file.
The URL is similar to a website address and must begin with the protocol, followed by the website, followed by the file's location, such as:
https://www.somewhere.com/somefolder/somefile.mp3
Only enter URLs of trusted websites. Always be vigilant when accessing any remote content.
This command may not be present on unsupported systems or devices.
This submenu lists recently used files. Select an item from the submenu to quickly reopen one of the files. Use Clear List or Storage Options to clear the list.
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.
Use Information to assign or change 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, .opus and .mp3. Some file types only store a subset of all the items given in the File Information window. Cover art is not supported in .wav files. 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.
Do not replace artwork with high resolution photos from a camera! Such photos can easily exceed 3MB in size, which is excessive and completely unnecessary for artwork. Use image editing software (Windows Paint) to reduce the size and resolution of the photo. The artwork should be cropped square. A resolution of 512 by 512 pixels or less is recommended. Save as JPEG.
Any additional information and metadata within a file not shown in this window may not be preserved by GoldWave.
Use the Copy and Paste buttons to copy information from one file and paste it into another. Use the Clear button to erase all text and the artwork.
The File Format section lists the file formats supported by GoldWave and the level of text information retained. Formats supporting unicode text allow international (non-latin) character sets to be used.
When using Batch Processing or when splitting a large file with Split File, Track number can be set to ## (two pound signs). During processing or splitting, the pound signs are replaced with the sequential number of the file being processed or split.
Use Infinity Link and GoldWave Link to transfer files between the desktop version of GoldWave and GoldWave Infinity. The link works on a local network only. GoldWave and GoldWave Infinity must be connected to the same network (usually to the same router).
To make a connection:
Once connected, use the Send file... button to select a file to send to Infinity. Or use the Send file... button in Infinity to send a file to the desktop.
The Receive folder is the folder where received files will be saved.
If Allow overwriting of files received is checked, files sent from GoldWave Infinity can overwrite files in the Receive folder having the same name. If unchecked, files cannot be overwritten.
If Open received files is checked, then after a file is received it is opened for editing.
If a connection cannot be established, check the following:
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 Save As to save it in a different type or save them in a separate Cue File.
An option to confirm saving is available in Window in the Options menu.
Use Save As to save a sound using a different filename, a different file type, or with a different number of channels.
Save As can be used to convert a sound to a compressed format, such as MP3, M4A/AAC, WMA, Ogg, etc. To save as MP3, select the "MP3 (*.mp3)" type and choose the Attributes button to select one of the many attributes. The smaller the bitrate (kbps number), the smaller the file will be. Note that quality may be reduced as well. To save as M4A (iTunes/AAC), select the "Media Foundations (*.m4a)" type. This option works only on versions of Windows that include the AAC encoder.
Use File Formats in the Options menu to assign a default format if you always prefer to use a specific file type and attributes.
The Converted Save menu provides another way to save the sound in a different format.
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.
The Select Audio Attributes window lists all combinations of attributes supported by a given file type. Use the Filters boxes to find the attributes required. Only attributes that match the filters are listed. The Attributes and File Compression sections explain the different kinds of attributes that may be listed.
An item in the list must be selected before choosing OK.
Use presets to save and quickly select commonly used attributes. Some presets are included for Wave, MP3, M4A, and FLAC files.
Choose Download to allow GoldWave to download the LAME encoder automatically. You must have an active Internet connection that does not require the use of a proxy server. Your firewall setup must allow GoldWave access to the network through HTTP.
Choose Browse if the encoder is already installed to provide the location of the encoder to GoldWave.
Choose Cancel to save the file in a different format.
For information about installing the LAME MP3 encoder manually, please see the website. A separate patent license may be required to use the encoder. GoldWave Inc. does not provide a patent license.
Save All saves all modified sounds. A confirmation window appears before all changes 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.
Use the Converted Save submenus to save the sound in different formats. The current format of the sound is not changed. Use Edit Presets to add or remove file types and attributes to the menu structure.
Use this window to add presets to the Converted Save submenu. To add a preset, select the file type first, then select the attributes, then enter a name for the preset in the Presets box and choose the add button. To remove a preset, select the file type, then select the preset from the Presets drop down list and choose the remove button. See presets for more information and using presets. Refer to the Attributes and File Compression sections for more information about attributes.
Batch Processing is a powerful tool for processing and converting a set of files automatically.
Use Batch Processing to:
Each tab contains settings to configure edits and effects, conversions, 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.
Files are processed one at a time in the order they are listed. Each file is processed as follows:
To process raw files that GoldWave cannot open automatically, such as .vox or .pcm, use Options | File Formats to add a default format for that type.
Source Tab
Use these controls to specify files to process. Select
Current Sound window
to process the currently active Sound window opened
in GoldWave. Only items on the Process
tab are applied. No other changes are made. The file is not converted and the information
is not changed. Select All Sound windows to process all opened Sound windows in
GoldWave in the same manner.
Select Files and folders to create a list of files to process. 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 Clear button removes all files and folders from the list.
Process Tab
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 Clear
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 or use the Up or Down
button.
To add an effect, use the Add Effect button to display a tree list of all effects available and their presets. Select the preset you want to use. If you require custom effect settings, you must create new presets for the effect outside of Batch Processing. To do that, open a file, display the effect window, adjust the settings, then add a preset. Chains may 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 sets the part of the file (the selection) to use for all subsequent effects and edits during processing. The selection can be specified using time, percent, or cue point names or indexes. If using the Cue option to set the selection, the first cue point matching the given name is used. If a number is given, cue points are searched for a name matching the number. If no such cue point is found, then the number is used as an index in the list of cue points. So if the number 6 is given, for example, a cue point with the name "6" is searched for first. If that name isn't found, then the sixth cue point is used. The Quick selection drop down list contains many examples of selection settings.
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 Batch Processing.
To add or edit a comment, either double click a line in the process list or select a line
and press the Ctrl+N key. If you are changing or adding a comment to
an existing preset, be sure to update the preset by choosing the preset
Add button.
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.
After performing all these steps, the process list will contain eight edit and effect items. Use the Convert and Destination tabs to set the file type and destination folder for the 10 second sample files (so the original files are not overwritten). Add all the songs in the Source tab and begin processing.
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 Copy, then use Batch Processing and add a Paste or Mix edit command to the Process list.
Convert Tab
If the Convert files to this format box is checked, then files
are converted to the format specified on this tab. Otherwise
no conversion is performed and a processed file will have the
same format as the original file, if possible. If the same format
cannot be used, then an error is reported.
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, use the Custom button 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.
Destination Tab
To stored all processed files in the folder
where they currently reside, select Store all files in their
original folder.
To store all processed files 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, select Preserve subfolder structure to ensure that the relative subfolders are maintained when storing processed files in a different destination folder. If this item is not checked, all files are stored in the destination folder and no subfolders are 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 checkbox. GoldWave fully processes original files before overwriting them. Check the Only if older than the original checkbox to process and overwrite files that have not been processed or modified recently. Files are processed only if the destination file does not exist already or if the destination file is older than the original file. Selecting the Delete original files checkbox removes the original files after processing. Note the warning below.
Overwriting or deleting original files is not recommended. Processing lossy formats such as MP3 and iTunes M4A reduces the quality. Always keep a copy of the high quality original files.
Use Create log file to save the processing details shown in the progress window to a text log file that can be viewed and searched later (in the Notepad accessory, for example).
Information Tab
Use this tab to control how information is processed. See
File | Information.
Information Tab Settings | |
---|---|
Setting | Description |
Retain current text information | Keeps the file's current information unchanged. |
Replace text information and picture | Replaces all of the information with new information provided using the Set Info button. Any blank entries are removed from the file. The picture is replaced or removed as well. |
Replace specified text information and picture only | Replaces the information with new information provided using the Set Info button. Blank entries are not removed or changed, except the Title. GoldWave uses the file's name as the Title text when a file currently has no Title. If a new picture is provided, it will replace any picture in the file. Otherwise any existing picture will not be changed. Note that you must set the picture before each batch processing as it is not retained from one session to the next. |
Remove text information | Removes all text information from the processed file. The Title is not set. |
Remove text information and pictures | Removes text, cover art, and other pictures. |
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. If the Title is left blank and the file does not have a title, GoldWave sets it to the file's name.
Depending on the settings used, all processed files may have exactly the same information (except for ## track numbering and title mentioned above), so care must be taken when specifying file specific information such as the 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.
Add Folder button on the Source tab of Batch Processing window.
Use this window to add an entire folder or file type to Batch Processing source list.
Enter the folder path in the Folder box or use the folder button to browse for a folder.
Specify the file type in the Type filter box. To add all files in the folder, select the *.* item from the drop down list. To add Wave files only, select the *.wav item. To add MP3 files, select *.mp3. Other unlisted items can be entered manually, such as *.abc.
Use the Include all subfolders checkbox to include all subfolders within the given folder. The entire tree of subfolders will be added to the list for processing. Use the Preserve subfolder structure checkbox on the Destination tab of Batch Processing to maintain the relative subfolder structure when storing processed files.
Add Edit button on the Process tab of Batch Processing window.
Use this window to add edit commands to Batch Processing process list. Supported commands are listed in the Edit command listbox. Additional settings may appear in the Settings area when a command is selected. Some commands do not have any additional settings.
When changing the selection, use the Quick settings drop down list to see examples of selections. If using the Cue option, the first cue point matching the given name is used. If a number is given, cue points are searched for a name matching the number. If no such cue point is found, then the number is used as an index in the list of cue points. So if the number 6 is given, for example, a cue point with the name "6" is searched for first. If that name isn't found, then the sixth cue point is used.
For more information about each command, look up the command in the Edit Menu Commands.
See Also: Editing Overview, Batch Add Effect Window
Add Effect button on the Process tab of Batch Processing window.
Use this window to add an effect to the Batch Processing process list. The tree list shows all plug-in modules.
To use effect settings that are not available in any of the current presets, open a file and use the effect to create a new preset with the settings you need.
To select an effect from a different plug-in module, scroll down the list and select the module name to expand its branch.
To use the effect on part of the file only, add an edit command to set the selection first.
See Also: Editing Overview, Edit Menu Commands, Effect Menu Commands, Presets
Add Logic button on the Process tab of Batch Processing window.
Use this window to add logic to the Batch Processing process list. Construct a logic statement or add labels by using the drop-down list boxes and edit boxes. See the tables below for details.
Logic Statements | |||||||||||||||||||||||
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
Statements | Description | ||||||||||||||||||||||
None | Use this statement to unconditionally change the flow of processing by performing a command. | ||||||||||||||||||||||
Label | Labels are used by the goto command to skip ahead or loop back to a specific point in the processing list. | ||||||||||||||||||||||
If value of |
Compares the value of two parameters using the selected comparison operator. Parameters are listed below.
|
Logic Commands | |
---|---|
Command | Description |
goto | Skips ahead or loops back to the given label in the processing list. Enter the label in the box below this drop-down list. Use the Label statement to add a label in the list. |
cancel processing this file | Stops processing the current file and starts processing the next file. |
error | Stop processing and displays an error message. Enter the message in the box below this drop-down list. |
terminate all processing | Stops all processing. Processing of the current file is stopped and no other files are processed. |
See Also: Editing Overview, Edit Menu Commands, Effect Menu Commands, Presets
Batch Processing on the command line uses presets, so the first step is to use Batch Processing to create a preset that does the processing required.
The -process command line parameter starts GoldWave in Batch Processing mode and processes files given on the command line. The basic syntax is as follows:
"C:\Program Files\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. Other parameters include -proclog:filename to set a log file, -region:start,length to set the initial selection for processing, and -clipboard:filename to set the initial contents of the edit clipboard, and -outfolder:pathname to set the destination folder.
To convert all Wave files to MP3 on the command line, for example, 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"
To include all subfolders when processing, include the -subfolders command line parameter:
-process:MP3 "C:\My Music\*.wav" -subfolders
Make sure all folder settings under the Destination tab are set appropriately in the preset before using this parameter and specify the absolute pathname of the folder (do not use relative pathnames). To specify a different destination folder than the one contained in the preset, include the -outfolder command line parameter. The Preserve subfolder structure setting must be set in the preset if the hierarchy is to be preserved. Quotes must be used around the entire parameter if spaces are used:
"-outfolder:C:\My Music\Batch Destination"
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:filenameQuotes are required around the entire parameter if the filename contains spaces:
"-proclog:C:\Log Files\Batch.log"
All messages and errors are appended to the log file in unicode (UTF-16). If you do not require a log file, but still need GoldWave to always close, use nul as the filename:
-proclog:nul
To process many commands in a single instance, store the list of command lines in a text file and use the @ symbol to specify that file in the process parameter. For example:
"C:\Program Files\GoldWave\GoldWave.exe" -process@batch.txt -proclog:batch.log
If a log file is specified, it will be used by default for all command lines in the list file. A different log file may be specified within the list file too.
Each command line must be stored on a single line in the list file. Do not break a command line across multiple lines. A list file would look like this:
-process:Trim *.wav -region:10.0,30.0 "-process:Convert to MP3" "C:\My Music\iTunes\*.m4a" -subfolders -proclog:itunes.log "-process:Match Volumes -18dB" *.wav
Exit closes all Sound windows then closes GoldWave. Any playback or recording is stopped. You are asked to save any changed files.
Edit commands remove, insert, copy, move, mix, or replace sections of sound. For an introduction to the concepts and terms used in this section, refer to the Editing Overview section.
Use Undo to reverse the most recent change made to a sound. The undo system keeps copies of the preprocessed audio in temporary storage, which can require a significant amount of storage when working with large files in long editing sessions. Use Options | Storage to change temporary storage settings and the number of undo levels allowed. To disable Undo, set the number of undo levels to zero.
Use Redo to reverse the most recent Undo. It restores the last undone change without any processing. Use this after accidentally undoing a change (such as a recording) to recover the change or to do a quick before-and-after comparison. Redo is possible only immediately after Undo. If any other changes are made, Redo cannot be used again until Undo is used first. Many levels of Undo and Redo are possible. In other words, you can undo several changes and redo each of them.
Use Cut to remove the selection from the sound and place it in the clipboard. The contents of the clipboard can then be mixed or placed into another sound using any of the other editing commands, such as Mix, Paste, Paste New, Crossfade, or Replace.
To remove the selection without placing it in the clipboard, use Delete instead.
If only some channels are selected in a stereo or multichannel sound, then only those channels are removed. Since it is not possible for one channel to be longer than another, the ends of the cut channels are padded with silence.
Use Copy to copy the selection to 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 any of the other editing commands, such as Mix, Paste, Paste New, Crossfade, or Replace.
To copy individual channels, use the Select Channels menu to select one or more channels.
In the Copy edit command in Batch Processing, audio may be copied from a separate file using the Copy from the following file setting.
Use Copy To to copy the selection to a new file. This may be used 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 to specify the filename, type, and attributes for the file. This is the same as Save Selection As. To automatically split a large file into several smaller pieces, use the Cue Points tool and Split File.
To save individual channels, use Select Channels menu to select one or more channels.
Use Paste New to create a new Sound window containing the sound in the clipboard. To enable Paste New, use Copy first. The new sound will have the attributes and length of the clipboard sound. This is useful when you need to edit, process, or save the clipboard audio.
To save the selection directly to a new file without copying or pasting, use Copy To.
After copying audio to the clipboard using Copy, use these commands to insert it audio into another sound. Paste inserts the clipboard at the start marker's position.
The length of the sound is increased so that the clipboard sound will fit and any audio after the insertion position is shifted so that it comes after the inserted audio. The clipboard sound is automatically converted to match the attributes of the sound, when possible.
The Paste At menu list commands to insert the clipboard at positions other than the start marker. Use Beginning to insert at the beginning of the file. Use Finish Marker to insert at the finish marker's position. Use End to insert it at the end of the file.
To replace the selection, use Replace instead.
To join many files together, use the File Merger tool.
Copying a small selection and pasting it several times creates a stutter effect.
Use Mix to blend (layer or combine) the clipboard audio with the selection. To enable Mix, audio must be copied to the clipboard first using Copy. Mixing essentially allows two sounds to be played at the same time, such as vocals and music or voice overs.
See the Mixing overview for step-by-step instructions.
To automatically fade the music in and out for voice overs, use Voice Over instead.
Mix Settings | |
---|---|
Setting | Description |
Time where mix will begin (s) | Sets the time from the start marker's position where the clipboard audio will be mixed. When mixing vocals in the clipbard, adjust the start time to synchronize the vocals with the music. |
Volume (dB) | Set the volume of the clipboard audio for the mix. A value of 0 is full volume. To change the volume of the selection, use Change Volume before Mix. |
Invert | Inverts the clipboard audio when mixing, subtracting the audio amplitudes instead of adding them. Usually this has no audible effect unless the selection and clipboard are almost identical. |
Use Crossfade to fade and mix the ends of two sounds, fading out one song while fading in another. To enable Crossfade, a song must be copied to the clipboard first using Copy.
See the Crossfading overview for step-by-step instructions.
The selection is used when performing a crossfade. Use Select All before using Crossfade unless you intend to crossfade within a certain part of the sound.
Crossfade Settings | |
---|---|
Setting | Description |
Duration (s) | Sets the length of time of the crossfade between the songs. It is the amount of time that the songs overlap, while one song fades out while the other fades in. |
Clipboard position | Choose End of selection to crossfade the clipboard song after the selection. The selection fades out and the clipboard song fades in. Choose Beginning of selection to crossfade before the selection. The clipboard song fades out and the selection fades in. |
Fade curve | Sets how the audio is faded. If a song is already faded out at the end, choose None. The fade is shown graphically. |
Settings may be previewed before applying them. Note that any changes made while previewing are not used until previewing is restarted.
See Also: Blend At Marker, Mix
Use Replace to replace the selection with the clipboard. To enable Replace, use Copy first. The selection is remove 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 and audio after the selection is shifted so that it comes after the replaced selection. To avoid changing the length of the file or altering the timing of the sound following the selection, use Overwrite instead.
Use Overwrite to overwrite part of the sound with the clipboard beginning at the start marker's position. To enable Overwrite, use Copy first. 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 nothing is shifting. 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 tempo, timing, or alignment of the sound following the selection needs to be preserved, such as for a video track.
Use Delete to remove the selection from the sound. The selection is not copied to the clipboard and the contents of the clipboard is not affected. Use Delete instead of Cut when the selection is not needed.
If only some channels are selected in a stereo or multichannel sound, then only those channels are removed. Since it is not possible for one channel to be longer than another, the ends of the deleted channels are padded with silence.
Use Trim commands to remove everything outside the selection or to remove leading or trailing silences. Trim is similar to cropping images to remove unwanted parts. Note that if only some channels are selected in a multichannel sound, the end of those channels is padded with silence. As an alternative, you can use the Copy To command to save the selection to a separate file.
Use Trim Beginning to remove audio before the start marker. If not all channels are selected, the area is replaced with silence instead.
Use Trim End to remove audio after the finish marker. If not all channels are selected, the area is replaced with silence instead.
Use Trim Silence to remove leading and trailing silences from the ends of the selection. Unlike other trim commands, Trim Silence works within the selection and does not remove anything outside the selection. Be sure the selection contains all the leading and trailing silences to be removed. Noise Reduction and Maximize Volume are recommended before using this effect to ensure a consistent silence level.
To remove all silences throughout the selection, use Silence Reduction instead.
Trim Silence Settings | |
---|---|
Setting | Description |
Silence to keep (s) | Sets the amount of 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. |
Threshold (dB) | Sets the highest level of background noise that should be considered silence. If the audio has a lot of background noise or hiss, use values above -30dB. If the audio is clean, use values below -40dB. |
Leading silence, Trailing silence, Both |
Sets the end of the selection to trim. Leading silence removes
the silence at the beginning of the selection.
Trailing silence removes
the silence at the end of the selection. Both removes silences
from both ends.
If only some channels are selected in a stereo or multichannel sound, then only that channel is trimmed. Since it is not possible for one channel to be longer than another, the ends of the trimmed channels are padded with silence, which results in trailing silence. All channels must be selected. |
Use Mute to replace 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.
Use Blend At Marker to smooth the transition between edit points. After cutting or pasting, the waveform at the selection endpoints may not match, causing a pop or click. Blend At Marker takes a small section of audio on either side of the marker and overlaps it with fading so that the section on the left is faded out and the section on the right is faded in. This ensure a gradual transition between edit points.
Note that the length of the file is reduced because of the overlap. When blending music, audio tracks from videos, or other time/tempo sensitive audio, be sure to compensate for the blend duration.
To crossfade between two songs with more control over the fade shape and duration, use Crossfade instead.
Blend At Marker Settings | |
---|---|
Setting | Description |
Duration of blend (s) | Sets the amount of time to overlap on each side of the edit marker. A value of 1.0 takes one second on the left and one second on the right and blends them, creating a one second blend. The length of the file decreases by one second due to the overlapping of the two parts. |
Marker | Sets the markers affected by the blend. After pasting in the middle of a file, both the Start and Finish markers may need to be blended. After cutting audio, only one the Start marker needs to be selected. |
Curve | Sets the fade curve for blending. For music and highly varying audio, choose "Equal power". For uniform audio, such as pure tones, choose "Linear". |
Use Find to search the sound for silences, peaks, or pop/clicks, or a text phrase in dialogue. Use the Find drop-down list to select what to find. Use the Where drop-down list to specify where to start searching. Select Past selection to find the next occurance. If the finish marker is currently at the end of the file, then the entire file is searched. Enter the search criteria in the boxes provided.
Search Criteria | ||
---|---|---|
Find Type | Setting | Description |
Silence | Threshold | Sets the minimun level of audio. Anything below that is considered silence. |
Duration | Sets the minimum length of time the audio must remain below the thresold to be considered silence. Using a value of 1.0, for example, ignores silences less than one second. | |
Peak/clipping | Threshold | Sets the level of audio that must be exceeded to be considered a peak. Anything below that level is ignored. Use a thresold of 0dB (or slightly below) to find areas of clipping before saving a file. |
Duration | Sets the minimum length of time the audio must remain above the threshold to be considered a peak or clipping. Usually this value will be 0 or very small, spanning just a few samples. For a 44100Hz file, a value of 0.00007 would be about three samples (3/44100). | |
Pop/click | Tolerance | Sets the percentage of change in the waveform that is to be considered a pop or click. Usually pops or clicks have a very sharp change in the waveform of about 700% or more. Start with a higher value and decrease it as required to avoid finding false positives. Some solo instrumentals (drums, trumpet, etc.) naturally have sharp changes where reliable pop/click detection is not always possible. |
Text | Phrase | Sets the search text. This can be a single word, but several words usually give better matches. Everything should be spelled out, including numbers and symbols. Enter "twenty four" instead of "24" or "plus" instead of "+". |
Recognizer | Sets the speech recognizer. On most systems only the "Microsoft Speech Recognizer" is available. | |
Confidence |
Sets the confidence level for recognition. If you find that too many false matches are found, increase the confidence
value. A value of 100% requires a very high level of confidence for each match. However some matches may be missed.
Using a confidence below 50% usually results in too many false positives.
Speech recognition rarely works well with music or singing. Ideally the audio should contain dialogue only with low background noise. Use Noise Reduction to minimize the background noise or use Stereo Center to reduce the music and extract the vocals. Due to the limitations of current speech recognition technology, false positives or missed matches may still occur. |
Use the Zoom in on selection if found checkbox to automatically zoom in (up to 1:1) on the selection when found. If this box is not checked, then the current zoom level is used and the view is scrolled to the selection instead.
Use Insert Silence to insert some blank space in the sound. Use this command to make room for recording or to insert a delay. The Duration sets the length of the silence. Values may be entered in any of the supported time formats. The Location sets the starting position of the silence. The silence may be inserted at the start or finish marker's position or at the beginning or end of the sound.
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.
Use the Channel submenu to select or unselect channels. In most cases, only selected channels are edited, processed, or played. Use this feature to copy certain channels from a sound or apply an effect to only some channels. When recording or using effects such as Resample, Playback Rate, or stereo and multichannel effects, the channel setting may not apply and all channels may be modified.
Click on the Channel item in the status bar to select channels or use Options | Toolbar to add the Select Channels button to the toolbar.
Selects all channels.
Lists all channels that can be selected. Channels with a leading checkbox are currently selected.
Use the Selection submenu to change the selection or moving the start and finish markers.
Use Set to change the selection range by setting start and finish markers to an exact time or sample position. To specify a time, choose the Time based range option and enter the times in hours, minutes, seconds, and thousandths of a second. Enter 1:23:45.678, for example. To specify a sample position, choose the Sample based range option and enter the positions. Tip: Right click on one of the up/down controls to change the precision of the controls.
To align the length of the selection to a CD sector or 1 kilobyte, select the appropriate option. The finish marker will be adjusted to align length when OK is pressed.
See Editing Overview for many more ways of setting the selection.
Use Previous to set the selection to its previous range. The last five selection ranges are stored automatically whenever the selection is changed. Using this command repeatedly sets the selection back to each of those five ranges. 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 have been deleted).
Use Move Start To Elapsed and Move Finish To Elapsed to move the start or finish marker at the current playback or recording position. Use the bracket keys, [ and ] to move the start and finish markers respectively. If the start marker is moved past the finish marker, the finish marker is moved to the end of the file. If the finish marker is moved ahead of the start marker, then the start marker is moved to the beginning of the file.
Use Move Start To Beginning to move the start marker to the beginning of the file. Use Move Start To Finish to move it to the finish marker's position. This sets the selection to nothing.
Use Move Finish To Start to move the finish marker to the start marker's position. This sets the selection to nothing. Use Move Finish To End to move it to the end of the file.
Use Recall Selection Range to set the selection to the range previously stored using Store Selection Range. Hold the Ctrl key to recall a range stored within a different Sound window.
Use Save Selection Range to save the current selection range. Use Recall Selection Positions reset the range to the stored range. Hold the Ctrl key to store the range so they can be recalled in a different Sound window.
Turn on Snap To Zero-Crossing to reduce pops and clicks caused between edit points. 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 adjacent, causing a click.
Snap To Zero-Crossing helps minimize the problem by ensuring 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 and multichannels sounds can have very different channel amplitudes, finding an ideal zero-crossing position may not be possible. Use the Edit | Channel menu to limit the snap feature to a single channel or use the "Sharp faded ends" preset in Shape Volume to force the ends of the selection to have zero amplitude (the unselected ends will have to be faded when deleting the selection).
If the zoom level is close enough that the true shape of the waveform is shown (such as Zoom 1:1), the snap feature is automatically turned off so that markers can be placed at any position.
Use the Cue Point submenu to edit and set cue points. Use cue points to mark or describe special places or points of interest within a sound.
Edit Cue Points displays the Cue Points tool.
Use Add Cue Point to place a cue point at the current playback or recording position. The name of the cue point is set to the time position by default, but that can be changed in the Auto Cue window using the "Cue Naming" setting.
Use Add Cue At Start to add a cue point at the start marker's position. Use Add Cue At Finish to add a cue point at the finish marker's position.
Use Add And Edit Cue Point to add a cue point and display the Edit Cue Point window.
Use Move Start To Next Cue to move the start marker to the next cue point to the right of the marker's current position or to the end of the file if no cue points are there.
Use Move Start To Previous Cue to move the start marker to the previous cue point to the left of the marker's current position or to the beginning of the file if no cue points are there.
Use Move Finish To Next Cue to move the finish marker to the next cue point to the right of the marker's current position or to the end of the file if no cue points are there.
Use Move Finish To Previous Cue to move the finish marker to the previous cue point to the left of the marker's current position or to the beginning of the file if no cue points are there.
Use Split File to divide the file into smaller files using cue points as split points. Displays the Split File window.
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.
Censor replaces the selection with a tone (beep), static, clipboard audio, or gibberish. Use this effect to cover up profanity, cleanly overwrite dialogue with other dialogue, or make dialogue almost unintelligible.
Censor Settings |
|
---|---|
Setting | Description |
Censor type | Sets the sound to use to replace the selection. Most of the censor types are simple tones or noises. Clipboard and Gibberish are special types. Clipboard replaces the selection with audio currently in the clipboard. Use Copy in the Edit menu to copy audio to the clipboard before using Censor. Gibberish scrambles the selection by moving and reversing short sections of audio, making dialogue difficult to understand. |
Censor volume (dB) | Sets controls the volume of the censored audio (the tone, clipboard, or gibberish). Usually this is set near full volume. |
Source volume (dB) | Sets the volume of the original audio. Usually this is set to the lowest value (Off). |
Crossfade time (s) | Sets how long it takes to fade from the original audio to the censored, which eliminates sudden transitions (possible clicks) between original to censored regions. Usually this is set to a small value. |
The copied dialogue must be exactly the same length as the replaced dialogue for it to be completely overwritten. If the clipboard audio is shorter, then the trailing end of the replaced dialogue will still be present. If the clipboard audio is longer, it will be truncated to fit in the selection. This effect does not alter the length of the selection. The timing of the surrounding audio is not affected, which maintains synchronization in music or video. Use Replace to replace a different size selection.
See Also: Mute, Overwrite, Replace, Presets
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.
Use Doppler to dynamically alter or bend the pitch of the selection altering the speed at which the waveform is played. Note that both pitch and tempo are changed.
Shape Controls allow the pitch/speed to be varied over the selection from 0.25 to 2.0 times normal. Use 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 chipmunk ("Faster") or a giant ("Slower").
See Also: Pitch, Time, Shape Volume, Presets
Use Dynamics to alter 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 above 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: Auto Gain, Compressor/Expander, Presets
Use Echo to create an echo or reverb effect in the selection. The settings include number of echoes, echo delay, echo volume, feedback, stereo, and tail generation.
Echo Settings | |
---|---|
Setting | Description |
Echoes | Sets the number of times the sound is repeated and mixed at a diminished volume with the given Delay. |
Delay (s) | Sets 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. |
Volume (dB) | Sets the loudness of the first echo. The volume of subsequent echoes is compounded in the same way the delay is. Usually values should be less than -6dB. |
Feedback (dB) | Sets the loudness of the feedback. Feedback makes an echo sound deeper and richer by regenerates each echo, and the echo of those echoes, and so on, creating many more echoes than before. |
Stereo | Causes the echo to bounce between the left and right channels. |
Generate tail | 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 setting is off, echoes will end abruptly rather than trailing off gradually. |
See Also: Flanger, Reverb, Presets
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 Save As on the File menu and change the type and attributes.
Compressors and limiters are used to decrease or limit the dynamic range of audio. In simple terms, 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.
Compressor/Expander Settings | |
---|---|
Setting | Description |
Multiplier (dB) | Sets 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. |
Threshold (dB) | Sets 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. |
Attack (s) Release (s) |
Set the amount of time required to fully activate the effect. 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. |
Expander Compressor |
Sets the type of processing required. Use Expander to change volumes of quiet sounds. Use Compressor to change volumes of loud sounds. |
Anticipate attack | Scans ahead for audio that crosses the threshold and activates the effect. 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 0.100 seconds later. |
Use smoother | Smoothes out any sudden volume changes which may occur during processing with small attack/release times. |
Process channels independently | Controls how each channel is processed. When checked, each channel is processed separately and different gains may be applied to each channel. When unchecked, one gain is applied to all channels, usually the lowest to avoid clipping. |
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: Auto Gain, Dynamics, Presets
Filters are used to remove a range of frequencies or amplitudes from a sound and can produce a variety of effects. The submenu contains several filter related effects.
Use Auto Offset Removal automatically remove a vertical shift or dc offset in the waveform. Offsets occur when audio is wired through several external devices that do not share a common ground. Computers with integrated sound hardware often have significant offsets as well. If the waveform is constantly above or below zero during silence, then it has an offset that must be removed. Offsets can cause pops and clicks between edit points and other problems.
Unlike Offset, this effect does not need to scan the entire selection and offsets cannot be set to specific values. Values to cancel out existing offsets are automatically calculated and used. Even varying offsets are removed.
Auto Offset Removal Setting | |
---|---|
Setting | Description |
Duration for offset calculation (s) | Sets the length of time to analyze the audio for an offset. For constant, stable offsets, set this value to the maximum. For fluctuating offsets, use lower values. |
See Also: Offset, Dynamics, Presets
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.
Bandpass/Stop Settings | ||
---|---|---|
Group | Setting | Description |
Initial frequency range |
From (Hz), To (Hz) |
Sets the initial frequency range of the pass or stop band. |
Final frequency range |
From (Hz), To (Hz) |
Sets the final frequency range of the pass or stop band. The band range fades from the initial setting to the final setting over the length of the selection. Dynamics must be checked to set these values. |
Settings |
Bandpass, Bandstop |
Sets the type of filter to use. Select Bandpass to keep only the frequencies within the range. SelectBandstop to block all the frequencies in the range. |
Static, Dynamic, Steepness |
See Low/Highpass filter for details. |
See Also: Low/Highpass, Equalizer, Noise Reduction, Spectrum Filter, Presets
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.
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 between +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 Spectrum Filter or Parametric EQ.
See Also: Low/Highpass, Equalizer, Noise Reduction, Spectrum Filter, Presets
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 muffled 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.
Low/Highpass Settings | ||
---|---|---|
Group | Setting | Description |
Cutoff frequency | Initial cutoff (Hz) | Sets the constant cutoff frequency for static filtering or the initial frequency for dynamic filtering. |
Final cutoff (Hz) | Sets the cutoff frequency at the end of the selection, allowing you to fade from one cutoff to another over the selection. Dynamics must be checked to set this value. | |
Settings |
Lowpass, Highpass |
Sets the type of filter to use. Select Lowpass to keep only the frequencies below the cutoff frequency. Select Highpass to keep only the frequencies above the cutoff frequency. |
Static, Dynamic |
Sets whether the filter is constant or changing. Select Static to keep the cutoff frequency constant throughout the selection during processing. Select Dynamic to fade the cutoff frequency from the initial value to the final value over the duration of the selection. | |
Steepness | Sets 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. |
See Also: Bandpass/Stop, Equalizer, Noise Reduction, Spectrum Filter, Presets
Use Noise Gate to reduce or eliminate noise in quiet or silent parts of a recording. Brief noises, such as mouse clicks or microphone bumps can be muted as well. A noise gate cannot remove noise from louder parts of the recording. Use Noise Reduction for that instead. To completely delete silences from a recording, use Silence Reduction.
Use Maximize Volume before using this effect to ensure volume is optimal for the presets.
Noise Gate Settings | ||
---|---|---|
Setting | Description | |
Threshold (dB) | Sets the level of audio that is considered noise. Audio below the threshold is reduced. The threshold should be set as low as possible so that only noise is removed. If the threshold is set too high, short and quiet sections of audio may be muted too. | |
Attack (s) | Sets the amount of time required to fade out to complete silence when the audio is below the threshold. | |
Release (s) | Sets the amount of time required to fade in to full volume when the audio rises above the threshold again. | |
Reduction (%) | Sets the level of reduction in the quiet parts. Normally this value would be set to 100 to replace the quiet parts with complete silence. A value of 25 reduces the noise only slightly (-3dB). | |
Ignore (s) | Sets the duration of brief sounds to ignore, even if they are above the threshold. Any sound less than this duration is muted. Use this to remove clicks, pops, and other very short sounds. Set this to zero to ensure that all sounds above the threshold are retained. |
See Also: Noise Reduction, Silence Reduction, Presets
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.
If the noise to be removed contains pops and clicks, use Pop/Click to eliminate those before using Noise Reduction. Any pops or clicks in the noise profile/envelope may cause a significant reduction in quality.
A frequency analysis window with Shape controls displays frequency in Hertz on the x axis and frequency magnitudein decibels on the y axis.
The frequency analysis provides graphical information about all frequencies within the sound at the given Time. For multichannel sounds, each channel is shown it's own colour. Moving the scroll bar located below the analysis window changes the time, showing the frequency analysis of a different part of the sound. The detail of the analysis depends on the FFT size setting, explained below.
Noise is removed using a reduction envelope. The shape of the envelope must closely match the shape of the noise 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 eeduction envelope setting. Use clipboard (see below) generally gives the best results, but requires copying the noise to the clipboard before using Noise Reduction. Also try the presets.
Noise Reduction Settings | |||
---|---|---|---|
Group | Setting | Description | |
Reduction envelope | Use shape | Creates an envelope based on the shape defined by the Shape Controls. Click the mouse in the analysis graph to draw the shape. A simple horizontal line at about 80dB removes a quiet hiss from a sound. In some cases, you'll need to trace the outline of the analysis graph or draw completely different shapes to reduce the noise. | |
Use current spectrum | Creates an envelope based on the shape of the graph shown in the frequency analysis window. Use this to remove a complex buzz or hum. The analysis Time must be set to a place where the noise is heard by itself. | ||
Use average | Creates 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 Copy a piece of noise into the clipboard. The piece must contain only the noise to be removed from the rest of the file. The noise can even be copied from a different file. After copying the noise, remember to change the selection to the part of the file you want to apply the noise reduction. | ||
Shape controls |
Point, X, Y, Time |
See Shape Controls. | |
Settings | FFT size | Sets the detail of the frequency analysis and the noise reduction envelope. Usually values of 11 or higher give the best results. See FFT Settings in Effects Overview for more information. | |
Overlap | Sets the amount of audio to overlap from one calculation to the next. A value of 4x works well. See FFT Settings in Effects Overview for more information. | ||
Scale (%) | Sets reduction envelope scale. A value of 100 uses the envelope as it is. A value of 200 doubles the envelope amplitude, which doubles the amount audio removed from the sound. A value of 50 halves the envelope amplitude, which halves the amount removed. Normally it should be set to 100. | ||
Output noise only | Makes the effect perform the exact opposite of noise reduction so that only the noise remains in the output. This is useful when previewing the effect to hear what is actually being removed from the audio. Do not check this box when removing noise. |
See Also: Equalizer, Spectrum Filter, Presets
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. 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.
See Also: Equalizer, Noise Reduction, Spectrum Filter, Presets
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.
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 and instrument solos, particularly trumpet solos. Always start with the maximum tolerance setting for those types of sounds.
Pop/Click Setting | |
---|---|
Setting | Description |
Tolerance (%) | Sets how abrupt a change in amplitude can be before it is considered a click. It is best to start with a value of 1000% or higher. 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. |
The filter requires a minimum selection of 4000 samples (about one tenth of a second at CD quality) to establish a baseline. Using the filter on a shorter selection has no effect.
See Also: Noise Reduction, Find, Presets
Use Silence Reduction to automatically remove silences from a sound. Use it to save storage space or to remove long pauses in speeches or police/airport radio recordings. Noise Reduction and Maximize Volume are recommended before using this effect to ensure a consistent silence level.
Silence Reduction Settings | |
---|---|
Setting | Description |
Silence threshold (dB) | Sets 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. |
Duration (s) | Specifies how long the silence must be before it is reduced. Any silences shorter than this remain unchanged. |
Reduce to (%) | Sets 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 (s) | Sets 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. |
Full crossfade | If is checked, the ends of the audio where silences were removed are gradually crossfaded over the entire length the remaining silence. If unchecked, a short crossfade of one-tenth of a second is used to join together non-silent sections where silences were removed. If there is a high level of background noise that varies, then full crossfade is recommended. However, if you hear unexpected overlapping fragments after removing silences, then this option should not be used. |
See Also: Find, Trim Silence, Noise Gate, Presets
Use Smoother to reduce hiss and crackle.
Smoother Setting | |
---|---|
Setting | Description |
Length | Sets the length of the smoother filter, which is the number of samples to be averaged. The larger the value, the more averaging is applied to the audio and the duller it will sound. |
Volume | Increases the volume of the processed audio to offset the the loss caused by averaging. Larger Length values require a higher volume. |
See Also: Pop/Click, Noise Reduction, Presets
A spectrum filter is a general purpose audio filter similar to Parametric EQ, but with much greater control. Instead of using individual bands, the entire frequency spectrum is controlled using Shape Controls to adjust 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 displays 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. Moving the time scroll bar, located below the analysis window, changes the graph to show the spectral analysis of a different part of the sound. Each channels is shown in a different colour.
Master gain sets the overall gain of the filter, which is equivalent to shifting the entire shape up or down.
Initially the shape line is horizontal 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.
Spectrum Filter Settings | ||
---|---|---|
Group | Setting | Description |
Mode | Static | The shape of the filter remains constant through processing and does not change over time. |
Dynamic |
Allows the shape to change over time by adding keyframes on the time line.
This mode displays additional controls to manage keyframes with different shapes. Keyframe selects the current keyframe on the time line. Use the Remove Keyframe button to remove the currently selected keyframe. Use Keyframe time to change the position of the keyframe on the time line or simply drag-and-drop the keyframe diamond in the keyframe slot. To add keyframes, change the time either by adjusting the time fader bar or enter it into the Time box, then change the shape or X and Y values. A new keyframe is added automatically at that time. Remember to save all your keyframes as a preset or they will be lost when the effect window closes. |
|
Graph range | Min (dB) | Sets the lower range of the y axis of the graph. |
Max (dB) | Sets the upper range of the y axis of the graph. Setting Min to -5 and Max to 5, for example, sets the graph to show a narrower range between -5dB to 5dB, allowing shape points to be set more precisely with the mouse within that range. | |
Settings | FFT size | Sets 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 (Gibbs phenomenon) |
Overlap | Sets the amount of audio data to overlap from one calculation to the next. The lowest value gives the fastest processing and generally works well. |
See Also: Equalizer, Noise Reduction, Parametric EQ, Presets
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.
Spectrum Filter Settings | ||
---|---|---|
Group | Setting | Description |
Volumes | Source (%) | Sets 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. |
Flanger (%) | Sets 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 (%) | Set 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. | |
Stereo | Makes the flanger and feedback audio alternate between the left and right channels, giving a more pronounced stereo effect. | |
Frequency modulator / vibrato | Variable delay | Sets the maximum variable delay in milliseconds. A value of 40 causes the delay to vary from 0 to 40 milliseconds. |
Frequency | Sets the speed at which the delay varies. 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. | |
Fixed delay | Sets a constant delay that is added to the Variable delay 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. | |
Sine modulator, Triangular modulator |
Sets the function for varying the delay. Select Sine to vary the delay on a sinusoidal pattern. Select Tiangular to vary the delay on a simple linear pattern. | |
Volume modulator / tremolo | Depth (%) | Set the relative amount to vary the flanger volume. If Flanger volume is 80%, for example, a depth of 100% varies the volume from 0% to 80%. A Depth of 25% varies it from to 60% to 80%. |
Frequency | Sets the speed at which the volume varies. A value of 2 will vary the volume over its depth twice a second. For a value of 0.2, the full depth is reached every 5 seconds. | |
Sine modulator, Triangular modulator |
Sets the function for varying the volume. Select Sine to vary the volume on a sinusoidal pattern. Select Tiangular to varys the volume on a simple linear pattern. |
See Also: Echo, Reverb Presets
Frequency Blend combines past frequencies and pitch shifted frequecies to generate unique audio effects, such as a sustained reverb effect or a pitch echo effect where the pitch continously increases or decreases over time.
Frequency Blend Settings | |
---|---|
Setting | Description |
Pitch shift (%) | Sets the amount of pitch shifting. A value of 50% shifts the pitch down one octave. A value of 200% shifts the pitch up one octave. This shift is applied to the feedback loop, causing a continuous shift over time, provided the feedback volume is not too low. |
Feedback | Sets the volume of the pitch shift feedback. A volume near 0dB generates a continuous pitch shift over time. Using the lowest setting generates a single, fixed pitch shift. |
Sustain | Sets the rate at which frequencies are blended over time. A value of 0 disables blended frequencies. Higher values slow down the rate of change of frequencies, which may enhance harmonics or extend stronger frequencies over time. |
From |
Sets the range of frequencies affected. Use this to limit the sustain or pitch shift to a narrower range of frequencies. |
Wet volume | Sets the volume of the processed audio mixed with the final output. |
Dry volume | Sets the volume of the original audio mixed with the final output. |
See Also: Doppler, Pitch, Reverb, Presets
Interpolate (see figure below) uses linear interpolation to smooth out the waveform within the selection. Use Interpolate only on a tiny selection to remove a pop or click. Do not use 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.
See Also: Channel Mixer, Dynamics, Channel Menu
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.
Mechanize Modulator Options | |
---|---|
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. |
Multiband Compressor controls the volume of specific bands of frequencies that pass a given threshold level. When the volume level of audio in a given band crosses the threshold level, the volume is decreased (or increased) based on a ratio setting. GoldWave goes beyond just compression by allowing a wider range of dynamics processing. Each band can be configures as an expander, limiter, or gate. GoldWave allows ratios of less than 1 for non-typical processing as well.
Use this effect to de-ess a recording or reduce silibance and plosives that occurs in a basic recording setup or with low quality devices. The effect can also brighten or dull parts of the audio.
This top graph displays a frequency graph and the width of 4 frequency bands. The time of the spectrum is given in Spectrum time and can be changed to view a the spectrum at a different place within the file. The scroll bar below the graph also changes the time. Three edit boxes allow you to set the width of each of the 4 bands or you can drag one of the band separators.
Controls and graphs for each of the 4 bands appear below the graph. These are similar to controls in the Compressor/Expander effect, but apply only to the assigned frequency band.
You may want to use Maximize Volume or Loudness to ensure the audio has no clipping or has a standard level so that volume level and threshold settings will be applied consistently.
Multiband Compressor Settings | |
---|---|
Setting | Description |
Spectrum time | Sets the time within the audio for the spectrum graph. Use this to see how high the frequencies peak within each band throughout the audio. This setting has no effect on the audio and is used for graphing only. |
Band 2, Band 3,Band 4 | Sets the frequency ranges for each of the bands. |
Mode |
Sets the dynamics mode for the band:
|
Gain | Sets the overall gain for the band. This gain is applied regardless if the threshold is crossed or not. Also referred to as the makeup gain. |
Threshold | Sets the volume level to activate. When the peak volume crosses this level, the band settings become active and the Ratio is applied to the volume. |
Knee | Sets the width of the transition from inactive to active. This smooths out the volume change around the threshold level. |
Ratio | Sets the volume scale factor. A ratio of 2 (2:1), for example, causes a level that is 2dB above the Threshold to become 1dB above instead. |
Attack | Sets the time required for the band ratio setting to be fully applied. |
Release | Sets the time required for the band ratio setting to become deactive. |
Process channels independently | When checked, each channel is processed separately and volume changes may be different for each channel. When unchecked, volume changes are averaged over all channels, preserving stereo balance. |
Synchronize value changes | When checked, any changes to a value in one band will be copied to all bands. Use this feature to set common values for all bands. |
See Also: Dynamics, Compressor/Expander, Presets,
Multichannel Mixer swaps, mixes, or combines channels. Each channel has a group of controls where the volume level of every channel can be set. Presets include multichannel mixdown, channel swapping, front to back swapping, and more.
After mixing, use Maximize Volume to ensure the audio has no clipping, but has full dynamic range.
Multichannel Mixer Settings | |
---|---|
Setting | Description |
Channel group | Contains a set of volume (%) controls for setting the mix levels of every channel. |
See Also: Channel Mixer, Pan, Balance fader, Presets
Offset 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. If you notice that silent sections of a sound are not at zero in the waveform, use this effect to adjust them to zero.
Positive values shift the graph up and a negative values shift it down. Use the Scan Offset button to automatically calculate the offset to use to remove any existing offset. After scanning completes, the value for each channel is set such that it will cancel out the offset in that channel. If the value is 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.
See Also: Dynamics, Filter dc offset during recording, Presets
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.
Spectrum Filter Settings | |
---|---|
Setting | Description |
Scale (%) | Sets the relative pitch by a percentage value. A value of 50% is equivalent to a downward shift by one octave. A value of 200 is equivalent to 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, Fine tune |
Sets the relative pitch in semitones (notes on a piano).
Given a tone at middle C, a the semitone value of 2
changes the tone of D. A value of -1 changes the tone to B.
A value of 12 makes the note one octave above middle C.
Fine tune sets 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 tempo |
If selected, a different, more complex algorithm is used to keep
the length and tempo the same. 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.
For information on the FFT size and Overlap, see FFT Settings. |
Higher quality pitch changes may be possible using Time. Try the "Similarity pitch up" or "Similarity pitch down" presets in the Effect Chain Editor tool.
See Also: Doppler, Frequency Blend, Playback Rate, Time, Presets
The Plug-in submenu lists all installed and enabled plug-ins. PION, DirectX and VST plug-ins are listed under Plug-In Over Network, DirectX and GWVST32 submenus if installed.
GoldWave checks for new plug-ins only during startup, so if a new plug-in is installed, you must restart GoldWave for it to be detected.
See Also: Effect Plug-ins, Options | Plug-in | Effect
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. The
Rewind
button on the Control window plays backwards as well, without processing the file.
Reverb adds fullness and richness to the sound by simulating the acoustic reverberation and echoes within a chamber or room.
Reverb Settings | |
---|---|
Setting | Description |
Reverb time | Sets the size of the reverb. A longer time simulates a larger chamber or room. |
Volume (dB) | Sets the loudness of the reverb. Values less than -18dB give good results. |
Delay scale | Sets the delay of the reverb for fine tuning. Use 1.0 for a standard reverb. |
See Also: Echo, Flanger, Presets
The Stereo submenu contains effects for stereo or multichannel files, such as swapping channels and left/right panning.
Channel Mixer 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. All selected left and right channels of a multichannel file are processed in pairs. The Center and LFE channels are not affected.
Channel Mixer Settings | ||
---|---|---|
Group | Setting | Description |
Left channel | Left volume (%) | Sets the percentage of volume of the original left channel to mix in the final left channel. |
Right volume (%) | Sets the percentage of volume of the original right channel to mix in the final left channel. | |
Right channel | Left volume (%) | Sets the percentage of volume of the original left channel to mix in the final right channel. |
Right volume (%) | Sets the percentage of volume of the original right channel to mix in the final right channel. |
See Also: Reduce Vocals, Pan, Balance fader, Presets
MaxMatch 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 independently, then uses the Maximize Volume effect on all channels to ensure there is no clipping. 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 all channels to have the same average and full dynamic range at the same time. Most channels will have a dynamic range of slightly less than 1.0 or 0dB.
Pan uses Shape Controls to dynamically adjust the stereo balance of the sound. The shape graph is divided into green and red regions, representing the left and right channels respectively. All selected left and right channels are processed in pairs. The shape line, initially located between the regions, is the center for panning. Bend and/or move the line to dynamically alter the balance. The figures below show a couple of examples of panning shapes.
Figure: Pan from left 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.
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.
Show balance calculates and displays the current peak balance in yellow on the graph. For a typically 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.
Change volume only 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.
See Also: Shape Volume, Balance fader, Presets
Reduce Vocals reduces vocals in stereo recording by subtracting the left and right channels and by using a bandstop filter. A true stereo file is required with mono vocals in the left and right channels. Any variation of the vocals in the left and right channels prevents perfectly removal. Any instruments that are identical in the left and right channels are removed as well.
To extact the vocals and remove the music, use Stereo Center instead.
Reduce Vocals Settings | ||
---|---|---|
Group | Setting | Description |
Channel cancellation volume | Volume (dB) | Sets the volume of the subtracted channels in the final output. |
Bandstop filter volume and range | Volume (dB) | Sets the volume of the bandstop filter in the final output. This is mixed with the subtracted channels to restore some of the stereo music. |
From (Hz), To (Hz) |
Sets the range of the bandstop filter. A narrow range retains more of the music, but removes less of the vocals. A wider range retains less of the music, but removes more of the vocals. |
Usually subtracting the left and right channels destroys the stereo sound, giving mono output. By using the bandstop filter, the effect is able to retain some of the low and high end stereo, enhancing the output.
Try the presets to learn what the different settings do.
See Also: Channel Mixer, Stereo Center, Presets
Stereo Center uses a frequency based method to separate vocals from music or to separate the perceived stereo center subchannel from the side subchannels and allows independent volume control of all subchannels.
Stereo Center Settings | ||
---|---|---|
Group | Setting | Description |
Center channel | Volume (dB) | Sets the volume of the perceived center subchannel, which usually is the vocals. To keep the vocals and remove the side subchannels, set this volume to 0.0dB and the side subchannel volumes below to -100dB (off). |
From (Hz), To (Hz) |
Sets the range of the filter used to extract the center subchannel. Any frequencies outside the range are excluded from the center subchannel. To retain bass and treble when adjusting the volume of the vocals, use a range from about 200Hz to 12500Hz. | |
Side channels | Left (dB) | Sets the volume of the left side subchannel. |
Right (dB) | Sets the volume of the right side subchannel. | |
FFT Settings |
FFT Size, Overlap |
See FFT Settings. |
Credit: Part of this effect uses a method by the developer of virtualdub.
See Also: Channel Mixer, Reduce Vocals, Presets
Time changes the playback speed or alters the tempo of the selection. This effect has many uses: it stretches or compresses a sound to fit in a certain time, it slows down instrumental music for easy transcription, or it changes the tempo of one musical passage to match rhythm and beats of another.
Time Settings | |
---|---|
Setting | Description |
Change (%) | Sets the relative change in time. A value of 50% slows down time, stretching the sound. A value of 200% speeds up time, shortening the sound. |
Length (s) | Sets a new length for the selection. Use this to make a sound fit a certain time, such as squeezing a 35 second radio commercial into a 30 second spot. The new length must be from one quarter up to four times the original duration of the selection. Settings beyond that range cannot be processed. |
Tempo (BPM), From, To | Changes from one tempo to another. Use the From and To values to specify the original tempo and the new tempo respectively. |
Algorithm |
Three different time altering algorithms are provided, each with
certain advantages and disadvantages:
|
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.
See Also: Doppler, Pitch, Playback Rate, Speed fader, Presets
Voice Over mixes foreground audio contained in the clipboard (use Copy first) with background audio contained in the file. Voice Over automatically fades the background audio in and out based on silent and non-silent regions within the foreground audio.
Use this effect to:
Voice Over Settings | |
---|---|
Setting | Description |
Fade out time (s) | Sets the amount of time it takes for the background audio to fade out before the foreground audio is mixed in. |
Fade in time (s) | Sets the amount of time it takes for the background audio to fade in when the foreground audio ends or contains a long enough span of silence. |
Fade volume (dB) | Sets the target volume of the background when it is faded out. Set it to the lowest value (Off) if no background audio is required when foreground audio is present. Usually values under -20dB are recommended when background audio is needed. |
Voice volume (dB) | Sets the volume of the foreground audio. Values around -2dB to 0dB are recommended. |
Silence level (dB) | Sets the level at which the foreground audio should be treated as silence. Recordings often contain some amount of background noise. To ensure proper detection of silence and accurate automatic fading, use values of -34dB or higher. Using the lowest (Off) value is not recommended. See the notes below to ensure consistent silence level detection. |
Silence allowed (s) |
Sets the duration of silence allowed in the foreground audio
before the background audio is
faded in. Usually narrations contain brief pauses where fading in is
unnecessary and undesired. Adjust this setting to avoid fading in and
out too frequently.
Note that this value is added to the fade in and out times to determine the total amount of silence required before the background audio starts to fade in. If Fade out time is 1.0, Fade in time is 1.0, and Silence allowed is 3.0, then the foreground audio would have to be silent for a total of 5.0 seconds before the background audio is faded in. |
The Voice Over window shows a graph of the current fade and mix settings. The clipboard (foreground or voice) audio is shown in red, the original file (background or music) audio is shown in blue, and a time line is shown under the graph. By default, background audio is faded out, which is shown by the blue region sloping downward. The steepness of the slope is controlled by the Fade out time setting. How low the background volume goes is controlled by the Fade volume setting.
Next the foreground audio is mixed in, which is shown by the red blocks. The volume (height) of the foreground region is controlled by the Voice volume setting. The space between the two red blocks represents the amount of silence allowed in the foreground audio before the background fades in again. It is controlled by the Silence allowed setting.
Finally, when the foreground audio ends or contains sufficient silence, the background audio fades in again, as shown by the blue region sloping upward. The steepness of the slope is controlled by the Fade in time setting. The level at which the foreground audio is considered silence is controlled by the Silence level setting and is shown as a yellow line in the graph.
If you use the Play button to start previewing, the graph displays a real-time representation of the faded and mixed audio.
See Also: Mix, Crossfade, Fade In, Fade Out, Shape Volume, Presets
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 effects change the sound's data.
Use Auto Gain to even out the volume to a consistent level across the selection. When recording a speech, interview, or telephone conversation, for example, the volume tends to vary depending on the position of the microphone relative to the person speaking. If the person or microphone moves around, the volume fluctuates. With telephone recordings, one person often sounds louder than the other.
Auto Gain automatically varies the volume level to increase it when it is low and decrease it when it is high (but it cannot correct overloaded or clipped audio).
Auto Gain Settings | |
---|---|
Setting | Description |
Target volume (dB) | Sets the final peak volume of the audio. The gain is increased or decreased so that the peak level always hovers around this value. A value close to or slightly below 0dB (100%) is recommend for maximum volume. Values less than 0dB can be used to limit or clamp the peak level to a certain volume. The "Peak reducer" preset shows how peaks can be reduced to 90% without affecting the average volume. |
Update interval (s) | Sets the time between volume adjustments. Values under one second give the best results. Use a smaller value if the audio level varies quickly. Use a larger value if the audio level is mostly even already, but needs occasional adjustments. |
Attack/release (s) |
Sets the time it takes for the volume to change from one interval to the next.
Larger values smooth out volume changes so that they are more gradual.
Setting this value to zero applies the new volume instantly every
update interval. Usually values less than Update interval
work best.
Note that when using a non-zero setting, peaks may exceed the target volume briefly as the volume gradually changes from a higher level to a lower one.
|
Maximum gain (dB) | Sets the maximum amount the volume can increase. Audio containing many noisy silences can result in sudden bursts of noise if the Silence level setting is not set high enough. By limiting the maximum gain, explosive volume increases for quiet sections are reduced. |
Silence level (dB) | Sets the level of noise to be considered as silence. Any audio below this level is ignored and not adjusted. In other words, Auto Gain is turned off while the audio level is below this threshold. Care must be taken when setting this value. Setting it too low greatly amplifies background noise. Set it as high as possible while still getting good gain results. |
One drawback of Auto Gain is that background noise is amplified along with the foreground audio. In recordings where the background noise is consistent, but the foreground audio varies, the end result will be consistent foreground audio with varying background noise. Use Noise Reduction or other Filter effects in GoldWave before using Auto Gain to reduce or eliminate that problem.
For multichannel files, each channel is processed independently. While this ensures that the channels will have the same volume level, it may produce unbalanced stereo for short update intervals. If multichannel audio is not needed, convert the file to mono or use Channel Mixer in GoldWave to mix the channels into mono before using Auto Gain.
See Also: Compressor/Expander, Maximize Volume, Presets
Change Volume makes the selection louder or quieter.
To make the volumes of several different songs sound similar, use Loudness instead.
Change Volume Setting | |
---|---|
Setting | Description |
Volume (dB) | Sets the relative volume. Positive values make the selection louder. Negative values make the selection quieter. A value of 0dB leaves the volume unchanged. If you are unfamiliar with the decibel scale, adjust the fader and watch the percentage value. |
See Also: Auto Gain, Maximize Volume, Mute, Presets
Fade In gradually increases the volume throughout the selection.
Fade In Settings | |
---|---|
Setting | Description |
Initial volume (dB) | Sets the starting volume. Use the lowest value to fade from silence. The volume increases to full (0dB) over the selection. |
Logarithmic, Linear |
Sets the shape of the fade. Logarithmic fades in more rapidly than Linear. |
See Also: Crossfade, Fade Out, Shape Volume, Presets
Fade Out gradually decreases the volume throughout the selection.
Fade Out Settings | |
---|---|
Setting | Description |
Final volume (dB) | Sets the final volume. Use the lowest value to fade to silence. The volume decreases over the selection. |
Logarithmic, Linear |
Sets the shape of the fade. Logarithmic fades in more rapidly than Linear. |
See Also: Crossfade, Fade In, Shape Volume, Presets
Match Volume makes the volumes of separate files seem similar.
This effect is based on a simple root-mean-square calculation and may not give the best results for some types of audio. Please use Loudness for best results based on human perception of loudness.
When creating a CD, for example, you may notice that each song 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 Match Volume, 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.
Match Volume Settings | |
---|---|
Setting | Description |
Average (dB) | Sets the overall average volume of the selection. |
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. |
Use Match Volume on each file to set the Average to the same value. Use Batch Processing to apply this effect to a group of files.
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.
To match the left and right channel levels within a single file, use the MaxMatch 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 Maximize Volume, Match Volume 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).
See Also: Loudness, Auto Gain, Compressor/Expander, Maximize Volume, MaxMatch, Presets
Loudness sets the volume of a file to an international standard level defined in ITU-R BS.1770-4.
When creating playlists you may notice that some songs have a different volume level and you have to adjust the volume control from one song to the next. By using Loudness, you can set the volume of each song to a standard level to avoid such variations. The LUFS (Loudness Units Full Scale) number provides a way to compare and match volumes of different files. Files with the same LUFS number will seem to have to same overall volume.
Loudness | |
---|---|
Setting | Description |
Volume (LUFS) | Sets the standard volume of the selection. The standard recommends -23 LUFS, however that tends to be much lower than most songs currently available. You can scan several songs in your playlist to see the current levels, then pick one of the lower levels to use on all files. |
Allow clipping | Uses the given volume, regardless if clipping is required. |
Reduce volume to avoid clipping | Reduces the volume just before processing begins to ensure that the audio will not be clipped. A lower volume is used for the entire file, so it will not be at the standard volume you've given. However this ensure no distortion occurs if the volume is too high for the file. |
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. |
Calculate Loudness Range | Rescans the selection to calcualte the loudness range (similar to the dynamic range). This is a much slower calculation, so it is not performed automatically. The loudness range is defined in EBU Tech 3342. |
Use Loudness on each file to set the Volume to the same value. Use Batch Processing to apply this effect to a group of files.
Unlike Maximize Volume, Loudness may result in clipping distortion if the volume 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 will occur. Use a lower volume to avoid clipping or select the appropriate clipping option.
This effect should not be used with Maximize Volume. Use one or the other, but not both (one cancels the other).
See Also: Auto Gain, Compressor/Expander, Maximize Volume, MaxMatch, Presets
Maximize Volume sets the peak level of the selection. First it searches the selection for the current peak level. Then it displays the level and the position of the level within the file. You can then specify a new maximum peak level.
Change Volume Setting | |
---|---|
Setting | Description |
Maximum (dB) | Sets the new peak volume. The volume of the entire selection is changed so that the peak will match that value. A value of 0dB normalizes the volume to full dynamic range, making the sound as loud as possible without clipping distortion. |
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.
Do not use Maximize Volume after using Match Volume.
See Also: Auto Gain, Match Volume, MaxMatch, Mute, Presets
Use Shape Volume to reshape the volume envelope of the selection using Shape Controls. The volume envelope is defined by the shape line, initially horizontal at 1.0, representing normal volume. Bend or move the line to dynamically change the volume over the selection. Add a point below 1.0 to decrease the volume. Add a point above 1.0, in the red section, to increase 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.
Note that Voice Over makes this much easier.
See Also: Auto Gain, Crossfade, Fade In, Fade Out, Pan, Voice Over, Presets
Playback Rate 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. To change the playback speed without changing the pitch, use the Time effect.
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 Playback Rate for the change to be permanent. Some file attributes may have a fixed sampling rate. A warning is shown in that case. An MP3 file with "Layer-3, 44100Hz, 192kbps, stereo" attributes, for example, is always saved with a 44100Hz sampling rate. You must use Save As to select attributes with the rate you want.
See Also: Doppler, Pitch, Time, Resample, Speed fader, Presets
Resample 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.
Resampling is done using a high quality polyphase algorithm with a filter length of 192.
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.
See Also: Time, Playback Rate, Presets
Before reading this section, review the terms introduced in the Interface Overview and Control Overview sections.
Control playback and recording with these commands. Use these commands to play the entire file, just the selection, or start playback from a number of common positions without having to move the Playback marker manually. Or start recording in a new file, record over the current selection, or record in dictation mode.
Plays the entire active sound.
Plays the current selection in the active sound.
Plays the unselected sections in the active sound. Playback will be limited to the view when zoomed in so that the entire beginning of the sound does not have to play.
Resumes playback from the current Playback marker position and stops at the Finish marker.
Resumes playback from the current Playback marker position and continues to the end of the sound.
Plays the sound currently shown in the view.
Plays the sound currently shown in the view and continues to the end of the sound.
Starts playback a few seconds before the Start marker and stops at the Start marker.
Starts playback a few seconds before the Finish marker and stops at the Finish marker.
Starts playback at the Finish marker and continues to the end of the sound.
Starts playback before the Start marker (limited by the view) and then plays the selection twice, then continues to the end of the sound.
Starts playback about a second before the Finish marker then loops to the Start marker and plays about another second.
Plays backwards. Set the speed of Rewind in the Play tab of the Control Properties window.
Plays faster. Set the speed of Fast Forward in the Play tab of the Control Properties window.
Pauses playback. The visuals are paused.
Stops playback. Visuals may be reset or cleared.
Moves the Playback marker back and continues playing (if playback is active). Use the Play tab of the Control Properties window to set the skip time.
Moves the Playback marker ahead and continues playing (if playback is active). Use the Play tab of the Control Properties window to set the skip time.
Creates a new file and starts recording. See Recording Sounds for details;
Starts recording in the selection of the active sound. See Recording Sounds for details;
Allows switching between playback and recording. While recording, choosing a playback button stops recording. While playing, choosing the Record Dictation button stops playback and starts recording where playback stopped. See Recording Sounds for more information.
Pauses recording so that it can be resumed from the current elapsed time later.
Stops recording. Unused space will be set to silence or trimmed, depending on the recording mode. See Recording Sounds for more information.
Shows the Control Properties window.
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 on part of the sound, a scroll bar appears at the bottom of the Sound window for moving to different parts of the sound while still keeping the same level of magnification. Clicking-and-dragging the waveform with the middle mouse button (wheel button) is another way of moving around. The current level of magnification is displayed in the Main window's status bar next to the modified status. The magnification is given as a time, which is the amount of audio currently shown in the view. Click on the status item to display a scale factor or set the magnification.
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.
Use Specify to magnify the Sound window graph to any specified level. The level can be given as a time length or as a ratio.
Use the Length option to specify the amount of time to show in the graph. For example, use 1:00 to show one minute of audio.
Use the Ratio option to specify the number of samples samples 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.
Start time sets the position in the file to begin drawing the magnified waveform. If the given level is not possible, the closest valid level is used.
Use Selection to magnify the selection, 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.
Use Preset to magnify to the view to a favourite level. The sound is magnified to the level of detail specified in Options | Window. The level can be set to any preferred value.
Use Previous to return the view to the previous zoom level. Use this to switch back and forward between two different zoom levels.
When checked, Follow Playback scrolls the view 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 marker 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 beginning at the start marker's position. Use View | Finish to see the audio at the end of the selection.
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.
Amplitude Statistics shows various statistics related to the selection's amplitude, such as peak, offset, loudness, and range. Values that need attention are shown with a red triangle. Clicking on a peak or clipped item zooms in on the location of the value in the selection. Clicking on the clipped value cycles through all the possibly clipped samples, zooming in on each one.
See Also: Auto Gain, Compressor/Expander, Offset, Loudness, Maximize Volume
Use CD Reader to digitally copy 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). Due to the wide variety of interfaces, inconsistent device standards, and a long standing bug in the Windows USB driver, 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 Download 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 Title button (or Alt+R) to manually rename it.
Use the
Save Titles button to save all titles and disc information so that it
can be retrieved automatically later.
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 Tracks button saves all selected audio 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 Range 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 CD database server.
CD Reader Options | |
---|---|
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. A single speed CD reads data at 150kBps. To read at 10x, for example, set the speed to 1500kBps. |
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. |
Database Options | |
Server | Sets the server address of the database to use for downloading CD information. Note that the original freedb.org server is no longer operational. Please use freedb.goldwave.ca. Or you can use gnudb.gnudb.org/~cddb/cddb.cgi. |
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 text and track number in title | Prefixes titles with the given text and track number when displaying titles. Enter text in the edit box with at least two # symbols for the track numbers, such as "## ", "Disc 1 ## " or "01-##-". The text must contain characters that are safe for filenames, so "Disc:1/##/ " is invalid. |
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. |
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
Save button on Read Tracks tab of the CD Reader window.
Use this window to set the destination folder, name, and format for all tracks read from the CD.
Destination folder specifies the location to store the tracks. Type in a folder name or use the folder button to browse for a folder.
The name of each track is the same as the name shown in the track list of the CD Reader window. If you've download track names from a database, then the Replacement character is used to replace any special characters, such as \, /, or : with safe characters so the files can be created properly.
Use the File type and the Attributes lists to specify the format to use for the tracks. You can select one of many popular compressed formats, such as MP3, M4A, or WMA, to save hard drive space. Use FLAC for lossless storage.
To overwrite any files with the same track name, check the Overwrite existing files box.
Use this command to compare two files to each other and graph the differences between them. Select two files from the list (you must open at least two files before using this command). Note that the order the files are selected may make a slight difference in the comparison results. Similar files with missing pieces or gaps or delays cannot be compared accurately due to sudden changes in synchronization. Settings are listed below.
To compare two channels, use the Edit | Channel menu to select a channel, then use Copy and Paste New to extract each channel.
The Result tab shows the differences of the two files over time. The top graph shows a frequency spectrogram of the difference. Blue shades indicate a stronger frequency in the first file. Red shades indicate a stronger frequency in the second file.
In the lower graph the left side shows the dB axis and the difference is plotted as a solid gray graph. The right side is the time/speed difference, plotted as an aqua line. Files on a similar time scale have a horizonal line. Files that vary temporally will have a diagonal line.
A "difference score" gives the overall average difference. The higher the score, the greated the difference.
Compare Files Settings | |
---|---|
Setting | Description |
From (Hz) To (Hz) |
Sets the frequency range to compare. Only frequencies within the From to To range are compared and all others are ignored. Use this setting to ignore low-end and high-end noises, such as hum, buzz, and audio compression artifacts. |
Sync time (s) | Sets the amount of time to match (or synchronize) the beginning of the two files. Use this setting to ignore any silence or noise (use Noise level setting below) at the beginning of the files and align them as closely as possible. Set it to zero to disable initial synchronization and compare the files without any initial time adjustment. |
Noise level (dB) | Sets the leading noise level to be considered silence when syncing the files. Audio is skipped at the beginning of the file while it is below this level. Use this to skip a fade in or a soft start of a file to get better initial synchronization. |
Resynchronization method |
Sets the type of resynchronization method to use to realign the files in time. Resynchronization is necessary
when files were recorded on different turntables or tape decks that have slightly different mechanical speeds.
|
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 border will be red.
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 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:
The Delete All button removes all cue points in the file. Use this button before using the Auto Cue button if you want to remove all existing cue points before automatically generating new ones.
The Cull button removes all cue points with the same position. Only the first cue point is retained and all other cue points with the same sample position are removed.
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
Use the Split File feature to divide a large file onto smaller files
using the cue points as split positions. This button displays the
Split File Window.
Auto Cue Button
Use Auto Cue to create cue points automatically based on fixed
intervals or detected silences. This button displays the
Auto Cue Window.
Use Auto Cue to generate cue points automatically at fixed intervals or at silences detected within the audio. Existing cue points are not changed or removed. Use the Delete All button on the Cue Points window first if you do not want to use any of the existing cue points.
The Split File feature can be used later to split a file into separate pieces based on these cue points.
Mark Silence Button
Displays setting for generating cue points at silences, such as marking gaps
between songs. A single cue point is added at each detected silence.
Mark Silence Settings | |
---|---|
Setting | Description |
Threshold (dB) | Sets the upper 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 considered 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 (s) | 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 (s) | Sets the minimum amount of time between one cue point and the next. If all the songs are longer than 2 minutes, for example, then 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. Use the Calculate button to calculcate the separation based on a given number of cue points. |
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 Button
Displays setting for generating cue points at fixed intervals, such as
having cue points every 5 minutes. Cue points are added at the specified
interval, starting at the given time.
Spacing Settings | |
---|---|
Setting | Description |
Start time (s) | 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 (s) | 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.). Use the Calculate button to calculcate the interval based on a given number of cue points. |
Duplicates
This setting determines how duplicates are handled. A duplicate cue point is one that is close to an existing cue point. Only
the position of the cue point is considered. Cue points with the same name, but different positions are not considered duplicates.
If "Allow duplicates" is selected, then new cue points will be created on top of existing ones. If "No exact duplicates" is selected,
then new cue points having the exact time as existing cue points are not added. If one of the other options is selected,
then new cue points close to existing cue points are not added.
Cue Naming
These settings control how names are assigned to the new cue points as they
are generated.
Cue Naming Settings | |
---|---|
Setting | Description |
Time based | Names cue points 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. |
Numbered | Names cue points based on three digit sequential numbers. The names will be 001, 002, 003, etc. The starting number depends on the number of cue points already in the file. |
Lettered | Names each cue point alphabetically, with three letters, such as AAA, AAB, AAC, etc. The starting name depends on the number of cue points 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 default 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.
Use this window to set or change the attributes of a cue point described below.
Use the playback buttons or the keys F4, F5, F7, and F8 to play, rewind, pause, or stop playback respectively.
Edit Cue Settings | |
---|---|
Setting | Description |
Name | The tip text that appears when the mouse is over the cue point in the cue points slot of a Sound window. A name is required, so it cannot be left blank. The name may be used for as filename in the Split File feature. If the name is set to "]X[" or includes the text "[-Exclude-]", that cue point and section of the file is skipped when splitting. |
Colour | Sets the colour of the cue point's triangle in the cue points slot. The default colour may be changed using Colour. |
Position (s) | Sets the time placement for cue point within the file. The position can be set relative to one of the listed locations. Use the - and + toggle to set the relative time before or after the selected location. For example, if "Playback marker" is selected with - and a time of "12.3", then the cue point is placed 12.3 seconds before the Playback marker. |
Description | Any text. This may be left blank. |
Use Split File to divide a large file onto smaller files using the cue points as split positions. For example, after recording one side of an album or tape, set a cue point at the start of each song, then use Split File 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 separate CD Recording software.
Use the Auto Cue to automatically set cue points at silences between songs.
If a cue name contains the text "[-Exclude-]" or is set to "]X[", that cue point and section of the file is excluded from the split files.
Any information entered in File | Information is stored in each split file, if possible.
Destination Folder Settings | |||||||||||||||||
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
Setting | Description | ||||||||||||||||
Use this folder | Specifies a destination folder where all the split files will be stored. | ||||||||||||||||
Use file's current folder | Uses the original file's folder as the destination folder for all the split files. | ||||||||||||||||
Overwrite existing files |
If checked, then files with the same name that already
exist in the destination folder are replaced (overwritten). If not checked, then splitting is aborted if a file with the same name is found. |
||||||||||||||||
Split Filename Settings |
|||||||||||||||||
Setting | Description | ||||||||||||||||
Filename template |
Controls how filenames are generated for the files.
The template may contain any combination of text, number symbols #, or tokens listed below.
The drop down template list contains some commonly used presets.
# is replaced with the sequential split file
number. Entering "Track##",
for example, names the files named Track01, Track02, Track03, etc.
# can be placed anywhere
in the template, so names like "#### - CD1" or "#Track##" would be
valid. The least significant digit is placed in the right-most
# slot, so
the first names would be "0001 - CD1" and "0Track01". Tokens are replaced with
information from the
original file, which has to be set before splitting the file.
Any characters or symbols that are invalid for filenames (such as :, ?, *, etc.) are replaced with spaces.
|
||||||||||||||||
First number | Sets the first number to use when creating the names with # as part of the template. It may also be used to set a track number in the tag/metadata. | ||||||||||||||||
Information Settings | |||||||||||||||||
Setting | Description | ||||||||||||||||
Discard cue point | If checked, then split files will not contain the cue point located at the beginning of each split. If this is not checked, then the cue point's name and description are stored in each file. If you've entered a description for each cue point and want it stored in each split file, make sure this option is not checked. | ||||||||||||||||
Replace track number | If checked, the track number information stored in split files is replaced with the sequential split number starting at First number. Otherwise the track number of the original file is stored. | ||||||||||||||||
Title | The title information for each split file will contain the selected item. | ||||||||||||||||
File Format Settings | |||||||||||||||||
Setting | Description | ||||||||||||||||
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. For accurate, glitch free alignment, you must
resample
the file to 44100Hz before splitting. 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. Choose the Set button to change the format. | ||||||||||||||||
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. |
Use the Effect Chain Editor tool to chain together a number of effects so they are all processed at once. There are many advantages to using chains, such as:
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. Files may be added in several ways:
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 or click on a column header to sort by file name or by file date. Clicking on the same column header a second time reverses the sort order. If more files are added to the list, they will not be sorted automatically and the column header must be clicked again if sorting is required.
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. Select Retain all cue points to keep existing cue points in the files. Otherwise only cue points between merged files will be added.
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 Window section under the Cue Points tool.
Use the Multi-Device Recorder tool to record from several devices simultaneously.
Audio is recorded into currently opened Sound Windows. Use File | New to create one or more Sound Windows for recording.
Initially a single recording device is listed and the currently active Sound Window is selected for output. Use the Add Device button to add additional recording devices. Select a recording device from the drop-down list. Select the output Sound Window and channel to record the audio in. If the recording device you need is not listed, it may not be connected or may be disabled in the Windows Sound settings.
If a recording error is indicated, check the following, then reselect the device:
Meters are live, allowing you do adjust the recording volumes for each device before recording.
Choose the Start button to start recording. Audio is recorded over any audio in the output Sound Windows' selected areas.
Choose Stop to stop recording. The Sound Windows will be updated with the newly recording audio only after recording is stopped.
The Speech Converter tool converts written text to spoken audio (text-to-speech) or spoken audio to text (speech recognition or dictation).
GoldWave uses the speech software in Windows to perform all conversion, so the quality of the voice or the accuracy of the recognition depends entirely on that software. The Speech Converter tool is not supported and will not work in versions of Windows that do not include the speech software. For more information see Microsoft's Speech website. Different voices and recognition engines are available from other vendors.
Use the Speech settings in the Windows Control Panel to configure text-to-speech and train speech recognition to your voice.
The Speech Converter tool consists of a text area with buttons above and below. The buttons along the top open a text file, save the text to a file, and perform basic editing functions on text. Use the Context Menu key (or Shift+F10) to display all the button commands as a menu for easier accessibility.
The buttons below the text area speak the audio, save the speech to an audio file, or take dictation from the microphone or an opened audio file.
Copy the contents of a website, document, report, or even chapters from a digital book (Sherlock Holmes, for example) and Paste them into the text area to have GoldWave read them to you.
You can save the audio directly to a file to copy it to your iPod or other portable player to listen to while jogging, working out, or doing other activities where reading isn't possible.
Use the Speak button to read the text. Playback is started from the edit cursor or at the beginning of the selection, if there is one. If the edit cursor is at the end of the text and there is no selection, then the entire contents of the text area is read.
Use the Voice Settings button to change the voice, volume, speed, and pitch. Windows usually includes just one voice, but others can be installed.
XML modifiers (such as <pitch middle='5'/>) are supported within the text to change the voice. Search online for "SAPI XML tags" for more information about modifying the voice using tags.
Use the Speak To File button to read the text directly to an audio file. Text is processed much faster than speaking through the sound hardware. Be sure to select all the text, otherwise only the selection or the text after the edit cursor is read to the file.
Audio files will be significantly larger than the original text files, so you must select a file type and attributes that minimize the audio file size while preserving good quality. Fortunately voice files do not require a high sampling rate or bitrate. The bitrate number, given in kilobits per second (kbps), controls the amount of space required per second of audio. Examples are given in the table below, with the amount of storage per minute. Note that CD quality audio requires 10MB per minute, which should be avoided if you intend to copy the audio to a portable player. The MP3 format may be the only one that will play on all portable players.
File Type, Attribues, and Size | ||
---|---|---|
File Type | Attributes | MB/minute |
MPEG Audio | Layer-3, 22050 Hz, 32 kbps, mono | 0.240 |
Windows Media Audio | WMA Voice 9, 20 kbps, 22.05 Hz, mono | 0.150 |
Ogg | Vorbis 22050 Hz, 30kbps (0.1q), mono | 0.225 |
Wave | PCM signed 16 bit, stereo (44100 Hz, CD Quality) | 10.58 |
When attributes do not include a sampling rate, then the rate specified under Voice Settings button is used.
Use Voice Settings to change the voice, volume, speed, and pitch. Not all voice engines support all settings or allow them to be finely adjusted.
Voice Settings | |
---|---|
Setting | Description |
Voice | Sets the voice engine to use. Windows may include just one voice, but others can be installed, such as IVONA. |
Volume | Sets the loudness of the voice. Normally full volume is used. |
Speed | Sets how slow or fast the voice speaks. Choose a setting you find most intelligible. |
Pitch | Sets the tone of the voice. Some voices can be changed up or down by a full octave. |
Rate | Sets the default sampling rate used when saving the speech to a file. This is used only if the selected file attributes do not specify a rate. |
Use the Dictate button to record or convert audio to text. The source audio can be taken directly from the computer's Microphone or from the currently opened file (if present). Use the Dictation Settings to select the source and configure the microphone or train the recognition engine.
GoldWave uses the speech recognition engine that is included with Windows to convert the audio to text. Unfortunately Microsoft still provides an outdated and primitive speech recognition engine that has very poor accuracy across different voices. It works somewhat better after training and only for a single voice. Any background noise or music adversely affects accuracy. Using GoldWave's Noise Reduction effect or other filters may reduce accuracy as well. Recordings must be as clean as possible, without any effects processing.
When processing from a file, a progress bar appears at the bottom of the window. Note that for long sections of speeches without any pauses, there will be a significant delay before any text appears and the progress bar advances.
Using the Microphone source may change your computer's audio settings and GoldWave's recording input. Use Device (and Volume for DirectSound) to reselect the input before starting any new recordings.
Before dictation can begin, the speech recognition software and the audio source must be specified.
Use the Recognizer drop-down list to select a speech recognition engine. Some versions of Windows may not include any speech recognition software. You can install the Microsoft Speech SDK (available from Microsoft's website) to enable speech recognition on some versions of Windows.
The audio Source may be from the microphone or from the current selection of an opened file in GoldWave. Use the Configure and Train buttons to configure the microphone and train the recognition software to understand your voice.
If you are unable to get speech recognition to work in GoldWave, use the Speech settings in Windows Control Panel to configure it.
These command configure and customize GoldWave.
Use Apps Options to enable network communications with apps or change passwords and other settings.
Check Allow network connections to apps to enable apps to interact with GoldWave.
The app's device must be connected to the local wi-fi network. Your computer's firewall must allow the following connections:
If those ports are blocked or GoldWave is not permitted to access the network, apps will not be able to communicate with the program. A connection cannot be made through a cellular data connection.
Use Connection password to set a unique password to prevent unauthorized access. The same password must be set in the app.
If Manually confirm remote connections is checked, a message appears when a remote device attempts to connect to GoldWave so you can allow or block the connection.
Enable or disable apps by checking the corresponding checkbox.
Use Colour Options to change the colour scheme of Sound windows and set channel colours. A preview window shows the current colour scheme.
Use the Item or Channel drop down lists and the Colour buttons to change the colours or select a different scheme from the preset Schemes drop down list.
Clicking the mouse in the preview window selects an item to customize.
Colour Settings | |
---|---|
Setting | Description |
Item | Selects the graphical item to change, such as the grid lines, background, etc. Use the adjacent Colour box to select a different colour. |
Channel | Selects the channel to change. Use the adjacent Colour box to select a different colour. |
Set all channels to the same colour | Apply any channel colour change to all channels, so all channels have the same colour. |
Automatically set unselected part to a darker colour | When the colour of a selection channel is changed, the unselected channel colour is automatically set to darker shade of the colour. |
Add gradients to colours | Makes the waveform and background colours gradients rather than plain, solid colours. Gradients are not shown in the preview window. |
Overlap all channels using transparent colours | Draws all channels in a single row so that they overlap. The colours selected are blended so that channel layers can be seen. This is not shown in the preview. Gradients cannot be used if this setting is checked. |
Displays the Control Properties window.
Use File Format Options to:
These options are contained under separate tabs.
Default Save Format Tab
Use this feature to set a default save format for new files when
using the Save
command. Use the
File type drop down list to select the type first, then select the
specific attributes from the list. Use the Filter
controls to lists only certain attributes of interest.
Whenever a file is created and saved, 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.
Select Use this format for... to use the same format for all "save" related commands (Save As, Save Selection As, etc.).
Select Do not allow other file types... to disable the file type and attributes boxes under "save" related commands to prevent them from being changed. It forces the use of the default format.
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 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.
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.
Windows Associations Tab
Use these settings to associate file types with GoldWave so that Windows lists GoldWave as one
of the programs in the "Open With" menu. GoldWave is not set as the default program for
the file types. You must use Windows "Default Programs" settings to change that.
Select the file types to associate with GoldWave, then choose the Apply button. Administrator privileges are required to update the associations in the Windows registry.
Use the Remove All button to remove all GoldWave associations from the registry. Administrator privileges are required.
Use Associate Format settings to assign a specific plug-in module and attributes to a file type or filename extension.
Associate Format Settings | |
---|---|
Setting | Description |
Extension | Sets the file type extension to be association to a format, such as vox, snd, raw. Do not insert a leading period. |
File type | Sets the plug-in module to assign to the file type. |
Attributes | Sets the specific attributes (channels, quality, rate, etc.) to assign to the file type. |
Rate | Sets the default sampling rate for the file type, if not already assigned in Attributes. |
Some plug-in modules allow custom attributes to be set. If so, the Custom button will be enabled and can be used to set those attributes.
Use Keyboard Options to set keyboard assignments (shortcuts) for functions in GoldWave. Current assignments are listed 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 rows and columns. To change a shortcut, move to right column for the function shortcut you want to change and type out the text of the new combination of keystrokes, such as "Ctrl+Shift+Up", "X", "Ctrl+P", "Shift+Enter", etc.
Note that the Left, Right, Up, Down, Page Up, and Page Down keys are used for navigation and scrolling the view in Sound windows and should not be reassigned. The spacebar ("Space") is used to activate the buttons in the Control window when that window has the keyboard focus.
Choose OK to save and use keyboard assignments. 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.
Use the Find box button to search for functions or shortcuts. Enter "Play", for example, to find playback related functions or "Ctrl+X" to find the function currently assigned to that shortcut. Press Ctrl+F to find the next match or Alt+F to change the search text.
The following table lists some of the names of keys that differ from their physical label.
Key Names | |
---|---|
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 usable) |
Scroll Lock | (not usable) |
Pause | Pause (may not work) |
Escape | Esc |
Space Bar | Space |
+ (plus) | Shift+= (US Keyboards) |
@ (at) | Shift+2 (US Keyboards) |
Media Control Buttons* | |
Play/Pause | mxPlayPause |
Stop | mxStop |
Previous track | mxPreviousTrack |
Next track | mxNextTrack |
Browser Back | mxBrowserBack |
Browser Forward | mxBrowserForward |
Browser Refresh | mxBrowserRefresh |
Browser Stop | mxBrowserStop |
Play | mxPlay |
Zoom | mxZoom |
* Note that using the media control buttons may cause the wrong shortcut text to appear in menus. Browser buttons many be assigned to other tasks by the operating system. Also some keyboards do not map keys properly.
Scan codes for keys can be entered directly by entering 0x followed by the hexadecimal scan code, such as 0x24 for the Home key. GoldWave shows key codes and names in the lower right corner of the window when possible. If nothing is shown or it doesn't change, then the key cannot be used.
See Also: Accessibility, Keyboard Commands
Use Language Options to set the language displayed in GoldWave. Choose a language from the drop down Active language list or the Open button to select a custom language file. Choose the Edit button to edit the selected language. Choose the New button to create a new language file. Choose the Download button to download the latest language files. Downloaded languages are added to the bottom of the list.
Use the Language Editor to create a new language file for GoldWave or edit an existing one to improve the translation. On the left side of the window is the list of items that require translation. Use the Search controls to find a specific item or text. Use the Expand and Collapse buttons to fully expand or collapse the selected item in the list.
On the right side are text boxes showing the original English text and translated text. The Original English text box is read-only and cannot be changed. Enter the translated text into the Translation text box. Some items require special formatting. Please read the formatting table below. Choose the Next button to go to the next text item. Choose the Incomplete button to go to the next blank item that has not been translated yet.
If the same English text is found in other items, choose the Replace All... button to replace all items with the exact same text. For example, if the current item contains the English text "Cancel", all other items containing that exact text will be replaced in the translated text. If another item contains the text "Cancel Now", that item won't be replaced because it is not an exact match.
Choose the Save button to save the translation. Choose the Save As button to save the translation in a different file. The current filename is shown at the top of the window. After saving the language, use the Open button in Language Options to use it in the program.
Formatting | |
---|---|
Character | Format |
Percent (%) |
The percent character is a special character used to insert numbers and text. Examples:
%d inserts a decimal number, %.2lf inserts a floating point number, and
%ls inserts some text such as a filename or preset name.
The translated text must include these exact sequences of
characters and must not add extra percent characters. Failure to do so will result
in program instability.
Note that text in the "Forms" section does not have this restriction. The percent character is used to denote a unit of percent. |
Vertical Bar (|) | The vertical bar character is a special character used to separate text into separate sections in message windows. Be sure to include the same number of vertical bars in the same relative positions in the translated text. |
Newlines |
Text containing newlines should retain the same number of newlines in the same relative positions in
the translated text. Be aware of any text ending in a newline and ensure that newline is retained. Also ensure
that extra newlines are not added to the end of the translated text.
Some items are lists. Each list item is must be on a separate line. The number of lines of text required is give above the Translation text box on the right. |
The Plug-in menu contains a list of submenus for configuring Effect, File, and Visual plug-ins, if any. Plug-ins may be created by different developers. Consult the documentation included with the plug-in for more information. GoldWave Inc. does not provide technical support for plug-ins developed by other companies or individuals.
If the DirectX plug-in is installed, this 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.
If the GWVST32 plug-in is installed, use this to search for and add 32 bit VST plug-ins to GoldWave.
If the Plug-In Over Network plug-in is installed, use this to add, remove, and update PIONs to GoldWave.
The Debug option uses an external web browser to run the plug-ins, allowing you to view the browser console to help in testing. You can also use this option if the embedded browser fails to install. Note that you must have a recent browser set as your default. Internet Explorer and the older version of Edge are not compatible.
To learn more about PION, please see the PION website.
The GoldWave Audio Plug-in Configuration window configures the internal GoldWave file plug-in and controls general file and MP3 handling and 8 bit encoding.
Many file types are handled directly, but some are handled through external (operating system) decoders installed on the system. These system decoders may cause problems or errors for raw file data or non-standard or unusual audio types. Use the system options to control when the decoders should be used and what file types to avoid. DirectShow decoders were widely used in versions of Windows prior to Windows 7. Media Foundation has replaced DirectShow in Windows 7.
GoldWave can use FFmpeg or Libav decoders if installed. Those support a huge range of file types. Use the FFmpeg/Libav decoders box to specify the folders where the avcodec-56.dll and related modules are installed. Please note that GoldWave Inc. does not provide support for FFmpeg/Libav issues. Due to patent and licensing issues, FFmpeg modules cannot be included with GoldWave or even hosted on the website. Anyone installing those modules is responsible for all legal aspects related to their use.
The MP3 encoding settings control how MP3 files are encoded. Usually these do not need to be changed.
Use the dither setting to mask quantization distortion caused by converting high quality audio to low quality 8 bit. A triangular dither masks the distortion with white noise. Keep in mind that 8 bit quality is inherently noisy, so the white noise is quite audible, but usually less distracting than quantization distortion.
Dithering only affects the quality of the audio saved to the file. GoldWave maintains high quality audio internally, so it is never dithered even after saving. You must close and reopen the file from storage to hear the 8 bit audio quality.
Use the ACM_STREAMCONVERTF_END setting to control how the Audio Compression Manager finishes encoding or decoding the last block of audio. If the last block is incomplete, check this option. This setting is necessary because of inconsistencies in ACM codecs.
Use the ACM MP3 decoder setting to enable MP3 decoding through the older ACM codec. The ACM codec incorrectly decodes MP3 files to silence in some cases, so this setting should not be used unless problems occur with the Media Foundation decoder.
Storage Options configure folders, file storage, and undo levels. See the Storage Overview section for additional information.
Storage Settings | |
---|---|
Storage Setting | Description |
Open sound folder |
Sets the initial folder to use when GoldWave starts when opening files.
Open
lists files in this folder.
Select Remember folder used last session to automatically store the last used folder when GoldWave closes. Select Use current file's folder to use the folder of the currently opened and active file. If no files are opened, then the last used folder is used. Select Always start with this folder and use the folder button to select a specific folder to use. |
Clear Recent File List | Removes the list of recently opened files from the File | Recent menu and the drop down menu on the Open toolbar button. |
Save sound folder |
Sets the folder to use when saving a new file or using
Save As or Save Selection As.
Select Use file's current folder or folder last used (for new files) to save the file in the same folder where it currently resides. For new files, the most recently used folder is used. Select Always use this folder and use the folder button to select a specific folder to use. |
Memory, Hard drive |
Sets where GoldWave stores audio, as explained
in the Storage Overview section.
Select Memory to store files in memory. Select Hard drive to store files on the hard drive. A folder where the temporary files will be saved can be set. The folder should be located on a large, local hard drive with plenty of free space. Changing the folder does not affect opened files already in temporary storage. |
Undo levels | Sets the number of edits/changes that can be undone. 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 hold the previous audio data. When using Memory storage, this should be set to a small value. Otherwise memory may be depleted quickly and system performance will degrade. |
Allow undo after save | Allows changes to be undone after a file is saved. Normally when a file is saved, the undo and redo history are cleared to release temporary storage and clear any ties to other files that may have been used when editing. If this option is checked, then temporary storage is not cleared and all ties remain so undo and redo can occur. Note that if a file is reloaded due to a format change (stereo to mono, for example), undo and redo are not permitted. |
Cue points storage |
Use Automatically import and export separate cue file
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! |
Use Theme Options to change the appearance of GoldWave with a different theme and colour scheme. Choose any theme from the drop-down list to use it. GoldWave should be restarted after selecting a new theme to ensure it is applied properly to all windows.
Two styles of toolbar icons are available: the classic coloured icons and modern monochromatic icons. The monochromatic icons can be tinted with any colour, but be sure to choose a colour that gives good contrast with the theme and colour scheme chosen.
Use Toolbar Options to customize GoldWave's toolbars. Initially the top toolbar 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 toolbars. The order which items appear in the Current list is the same as the order they will appear on the toolbar. You can drag-and-drop items within the Current list to rearrange them.
Use the drop down list to set the size, captions, and visibility of the toolbar.
To arrange toolbars, drag-and-drop them within GoldWave's Main window by clicking on the vertical bumps near the left edge of the toolbar. Right-click on the toolbar to reset it or to allow the order of the toolbars to be changed.
Use Window Options to configure the positions of Sound windows, set axis options, specify preset and initial zoom values, and set miscellaneous options.
Window Options Settings | ||
---|---|---|
Group | Setting | Description |
Sound window layout | Cascade | Allows Windows to arrange Sound windows in layered, cascaded arrangement. |
Maximize | Makes a Sound window occupy the entire area within the Main window. | |
Auto-tile | Resizes all Sound windows whenever a new sound is opened or closed so that all are visible. The arrangement is the same as using Window | Tile Horizontally. | |
Tabs | Makes Sound windows as large as possible, but with a tab for each file along the top for easier selection of opened files. | |
Miscellaneous | Always confirm before saving | Prompts to confirm saving a file whenever Save is used. |
Update "Default" effect preset after each use | Saves the current effect settings under the "Default" preset so those settings will be selected the next time the effect is used. The "Default" preset is updated only if the OK button is used. | |
Draw overview graph | Draws the Overview based on the audio waveform. If unselected, only lines are drawn, which is faster. | |
Hide playback marker | Hides the playback marker whenever playback stops. See Navigation for more information about the playback marker. | |
Hide selection handle { } boxes and playback triangle | Removes the selection handles { } from the start and finish markers and removes the green triangle from the playback marker. Use this setting when doing precise editing so that the waveform is not obscured by the markers. | |
Show playback indicator > in title | Indicates which sound window is currently playing audio by adding > to the beginning of the Sound window's title. | |
Show full filenames in title | Shows the full path and filename of the file in the title bar of the sound window (if there is sufficient space). | |
Retain current view and zoom when using Undo | Does not change the view and zoom to back to the previous level when a modification is undone. The view and zoom level are unaffected by Undo, when possible. Note that the view may have to be changed if the duration of the file has changed and the view is outside the bounds of the file. | |
Hide drag-and-drop selection icons | Removes the icons that appear at the top of the selection area in Sound windows. When these icons are hidden, you can still use Shift to drag the selection range or Ctrl to drag the contents with the mouse. | |
Axes numbering | X |
Sets the format for displaying the horizontal time axis near the
bottom of Sound windows.
|
Y |
Sets the primary units of the vertical axis in Sound windows shown on the left side.
|
|
Y2 | Sets the secondary units of the vertical axis in Sound windows shown on the right side. Grid lines are not shown for this axis. See Y above for the details of each setting. | |
Grid | Grid | Select Grid to display a custom, vertical grid over the waveform, which can be used to align markers. This can be used to show measures and beats in music. Use Insert Silence to add some silence to the beginning of the file to align the grid. |
Spacing | Sets the time between each grid line in seconds. The equivalent beats-per-minute (BPM) is shown. | |
Divisions | Sets the number of divisions between each grid spacing line. These are shown as dotted lines. | |
Snap markers to grid | Ensures that whenever edit markers are moved, they are snapped to the closest grid line. Note that if Edit | Selection | Snap To Zero-Crossing is checked, markers are snapped to the grid first, then the nearest zero-crossing point. | |
Mouse | Simple left and right mouse button selection method | Moves the start marker when the left button is clicked and move the finish marker when the right mouse button is clicked. No context menu appears. You will not be able to use the new right-click-and-drag features. This is the way old versions of GoldWave worked. |
Standard click-and-drag selection method with right-click menu | Sets the selection uses a standard click-and-drag method and displays a context menu when the right mouse button is clicked. Allows the left mouse click to be assigned to a specific function. | |
Single click | Sets the function for perform when the left mouse button is clicked. Available only if the standard method above is selected. | |
Ctrl click | Sets the function for perform when the left mouse button is clicked while the Ctrl key is held down. Available only if the standard method above is selected. | |
View | Time shown for the View Preset command | Sets the amount of time to show in the View when using Preset. |
Initial time shown when sound opened | Sets the amount of time shown in a Sound window when it is first opened. Choose All, to show the entire waveform. Using a value under 1 minute can speed up opening/drawing a large file. | |
Overview height | Sets the default height (in pixels) of the Overview area along the bottom of the Sound window. This setting does not change the height of the Overview in currently opened Sound windows, only new ones. Drag the splitter between the graphs to adjust the height manually. |
Although the evaluation version is fully functional, GoldWave is not free software. The evaluation version will expire after some use. Please see the website for purchasing and pricing details.
Choose Enter License to enter an existing license into the program. Or choose Free Trial License to request a free 30 day trial license from the GoldWave website.
Lost LicenseUse Setup Options to change the way GoldWave intializes (such as loading plug-in) and alters the way it runs. These settings should not be changed on most computers.
If certain plug-ins are causing division-by-zero or invalid floating point errors, check the Ignore floating point errors box to ignore the error. Note that this setting may cause distortion in some of the built-in effects in GoldWave, so it is better to contact the plug-in developer to correct the problem rather than ignore it.
Use the Backup Settings button to save all GoldWave settings and presets to a file (except keyboard assignments).
Use the Restore Settings button to load all the settings from the file.
Use the Transfer Settings From v5 button to get all settings and presets from GoldWave v5. Some settings in v5 are not compatible with more recent version of GoldWave. Use this buttom only if you have many custom presets or settings that have to be transferred. If you have only a few custom settings or presets, enter them manually instead. All settings in the current version of GoldWave will be lost. Backup current settings before using this command unless you just updated to the new version from v5.
To save keyboard assignments to a file, use the Save button on the Keyboard Options window.
These commands organize Sound windows and the Control window.
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 old versions of GoldWave.
Horizontal Control places the Control window along the bottom of the Main window and arranges the controls and visuals horizontally.
Vertical Control places the Control window along the right side of the Main window and arranges the controls and visuals vertically.
A list of all currently opened Sound windows is given at the bottom of the Window menu.
Displays an introduction to GoldWave.
Starts the default web browser to display the GoldWave manual.
Use this command to check for updates now.
If the Automatically check for updates box is checked in the Register or About window, GoldWave checks the website about once a month for updates. If a new version is available, a message gives you the option of downloading the new version from the website. Internet firewalls may need to be adjusted to allow GoldWave to make a HTTP connection to the website. This feature does not work if your Internet connection uses a proxy server.
Make sure that all opened instances of GoldWave are closed before installing an update.
Displays GoldWave version and registration information. See Check for Updates. for more information about the Automatically check for updates box.
Command line parameters change the way GoldWave starts when run from the Windows Command Prompt. Use the parameters to open, play, or record files, or to run batch processing. Also use the parameters when manually associating a file type with GoldWave. A parameter must be prefixed with - (minus), as shown.
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 given. Wildcards (*, ?) are supported in filenames. All files are opened or processed.
Command Line Parameters | |
---|---|
Parameter | Purpose |
-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 "C:\Program Files\GoldWave\GoldWave.exe" -process:Mix "My Music.wav" "-clipboard:My Vocals.wav" |
-close | Closes the program after playing the file on the
command line. This parameter must be used with -play. If
the user interacts with the program, it may not be closed.
Example "C:\Program Files\GoldWave\GoldWave.exe" -play -close "C:\My Music\Song.mp3" |
-config | Displays the Setup Options window to configure
core functionality in GoldWave. This is the same as using GoldWave Setup
from the Windows Start menu.
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 found on this window. |
-new(duration,rate,channels):filename | Creates a new file using the default File | New settings
if no other information is given. Use this with
-record to start recording in a new file from the command line.
The duration, sampling rate, channels, and filename may be specified on the command line as shown. The default save format is used when saving the file. When specifying a new filename on the command line, the file is overwritten without warning! Examples "C:\Program Files\GoldWave\GoldWave.exe" -new "C:\Program Files\GoldWave\GoldWave.exe" -new:Sample.wav "C:\Program Files\GoldWave\GoldWave.exe" "-new(60):One Minute Mono.wav" "C:\Program Files\GoldWave\GoldWave.exe" "-new(300, 48000, 2):Five Minutes DAT.wav" |
-nosplash | Suppresses the splash window when starting the program. |
-outfolder | Specifies the destination folder for all processed files for batch processing, overriding the batch preset Folder settings. If the folder name contains spaces, double quotation marks must enclose the entire parameter. If the folder hierarchy is to be preserved, then that option must be selected in the preset. This can be used only with -process. |
-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 or in a new file.
Use with -new to create a new file for recording.
Example "C:\Program Files\GoldWave\GoldWave.exe" -record Example "C:\Program Files\GoldWave\GoldWave.exe" -new(180):newfile.wav -record -close When specifying a filename on the command line, an existing file with that name is overwritten without warning! |
-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 "C:\Program Files\GoldWave\GoldWave.exe" -region:10.0,7.5 music.wav "C:\Program Files\GoldWave\GoldWave.exe" -region:441000,330750 music.wav |
-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. The window of the current instance is brought to the foreground. To prevent that, use -notop. Use the Setup Options window to change this setting. |
-subfolders |
Includes all subfolders when used with batch processing and a patterned
filename is given (such as *.wav).
This can be used only with -process. Before using this parameter,
folder settings for the
preset must be set appropriately under the
Destination Tab of Batch Processing.
Example "C:\Program Files\GoldWave\GoldWave.exe" "-process:iTunes to MP3" "C:\My Music\*.m4a" -subfolders |
GoldWave ("the package") includes the following software and documentation:
GoldWave.exe GoldWave application file GWSpeed64.dll Accelerated multithreaded code GoldWave.html GoldWave manual layout GoldWaveManual.html GoldWave manual Index.html GoldWave manual index Style.css GoldWave manual style Images (*.png;*.jpg) All images associated with this manual GWPreset.xml Presets and shapes ReadMe.txt Important information WhatsNew.txt Revision history FLACFile.pig FLAC file plug-in en.lang English language File OggFile.pig Ogg Vorbis file plug-in OpusFile.pig Opus file plug-in QTFile.pig QuickTime file plug-in WMAFile.pig Windows Media Audio file plug-in GWVST32.pig VST effect wrapper GWVSTBridge.pig VST 64 bit to 32 bit bridge GWVST.html GWVST manual
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 © 2023 by GoldWave ® Inc. All rights reserved. This document and associated images may not be shown on a public website.
Copies of the evaluation version of GoldWave may be given to anyone. Magazine and book publishers may include the evaluation version on cover and companion CDs and DVDs. Only the original GoldWave self-installing evaluation file may be distributed. This ensures everyone receives a complete and working copy. Distributing modified or incomplete copies is a violation of the copyright. Please note that GoldWave is not free software and should not be described as such.
GoldWave is a registered 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:
Postal Address
GoldWave Inc.
P.O. Box 51
St. John's, NF
CANADA A1C 5H5
In the digital world, everything is reduced to an on or off state so that it can be stored in computer memory as a single bit of information: 1 or 0. Complex real world things, like images and audio, cannot be directly represented in such a simple manner. An image is rarely composed of black and white dots and audio is rarely just on or off.
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.
Standard Sampling Rates | ||
---|---|---|
Attributes | Quality and Usage |
MB/Minute (16 bit, mono) |
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 information for two speakers, 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. Audio containing more than 2 channels are referred to as multichannel. GoldWave currently supports up to 8 channels in a 7.1 surround sound format as shown in the table.
Channel Layout | |||||
---|---|---|---|---|---|
Front left (FL) | Center (C) | Front right (FR) | 3.1 | 5.1 | 7.1 |
Low frequency effects (LFE) | |||||
Back (5.1)/Side (7.1) left (SL) | Back (5.1)/Side (7.1) right (SR) | ||||
Back (7.1) left (BL) | Back (7.1) right (BR) |
Due to the internal structure of Wave files and the Windows default channel ordering, the "back" channels are stored before the "side" channels in 7.1 surround sound. Therefore GoldWave refers to the 5.1 surround channels as "back" channels and displays the 7.1 "side" channels below the "back" channels.
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 on visuals 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 memory (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 absolute value of the amplitude scale converted to a percentage by multiplying by one hundred and adding a percent sign. Sometimes a negative percentage is used, such as in the Flanger effect, to invert the waveform.
The decibel scale (specifically dBFS) is unusual in that it is 0dB at the maximum peak level 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) and such values may cause clipping. When changing the volume, positive values increase the level and negative values decrease the level.
Levels shown in most visuals are dBFS and not dBSPL.
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 user for thresholds, such as in Silence Reduction, or in rare cases were the absolute level is set directly, 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, Noise Reduction, and Spectrum Filter.
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, Spectrum Filter, 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).
File compression (not to be confused with amplitude compression) makes files smaller using a variety of algorithms. 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 AAC, 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.
When using constant bitrate, each section of audio compresses to exactly the same size, regardless of the content. If the audio contains silence, then the data may be padded to fill the required bitrate. If the audio contains complex music, then quality may be decreased until it fits within the bitrate.
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 compression uses the smallest size possible for each section of audio. If the audio contains silence, then the bitrate will be very low. If the audio contains complex music, the bitrate will be at its maximum.
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.
Keyboard Commands | |
---|---|
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. |
Ctrl+G | Displays a window to set the playback marker's location to a specific time. |
H | Starts playback relative to the mouse's horizontal position in the waveform. |
Ctrl+H | Moves (hops) the selection to the next section between adjacent cue points. |
J, K, L | Rewinds, plays, and fast forwards respectively from the current playback marker'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. |
Q | Drops a new cue point at the start marker position. Cue naming is controlled by the Auto Cue settings. |
Shift+Q | Drops a new cue point at the finish marker position. Cue naming is controlled by the Auto Cue settings. |
Ctrl+Q | Drops a new cue point at the current recording, playback, or start marker position. Cue naming is controlled by the Auto Cue settings. |
Ctrl+Shift+Q | Drops a new cue point at the current recording, playback, 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+Backspace | Displays the Trim Silence window. Removes leading and trailing silences at the beginning and end 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 | Toggles the selection the left channel. |
Ctrl+Shift+R | Toggles the selection right channel. |
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 Selection 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) | Moves the start marker to the current playback position. |
] (right bracket) | Moves the finish marker to the current playback position. |
Shift+[ | Plays some audio up to the start marker. Marker preview controls the amount of audio played or three seconds is played if that value is set to zero. |
Shift+] | Plays three seconds of audio up to the finish marker. Marker preview controls the amount of audio played or three seconds is played if that value is set to zero. |
Ctrl+[ | Plays from the start marker to the finish marker. |
Ctrl+] | Plays from the finish marker to the end. |
Shift+\ | Moves the selection to its previous range. |
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 a play 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 play 3 button settings. Use Ctrl for play 1 and Shift for play 2. |
F2, F3, F4, F5, F6, F7, F8 | Plays 1, plays 2, plays 3, rewinds, fast forwards, pauses, and stops respectively. |
F9 | Starts recording in a new file. |
Ctrl+F9 | Starts recording in the selection of the active file. |
Ctrl+F8, Ctrl+F7 | 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. Note that if the source and destination are the same, then the source changes during processing.
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 memory storage (see Options | Storage).
Some 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:
Also be aware that if the source and destination are the same, the source is modified during evaluation. An expression such as wave(n)+wave(n-3000)/3 generates a recursive reverb effect because wave(n-3000) gives the stored, modified values instead of the original source values. This allows for feedback processing. Use separate source and destination sounds to avoid modifying the source.
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 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.
The following table summarizes evaluator operators and functions.
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 wave(n)*(wave(n)>0.001) replace all samples below 0.001 with silence (zero). |
^ | 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 |
db, idb | Converts a number to dB or converts a dB to a number. db(0.5) give -6.02.... idb(-6) gives 0.5.... |
exp | Exponential base e |
step | Unit step ( 0 for t < 0, 1 for t >= 0 ) |
int | Integer value |
rand(n) | Generates a uniform 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.
sin(2*pi*261.7*t)
Use the Play button to preview the expression before processing the entire file.
Expressions | ||
---|---|---|
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
This tutorial explains the steps required to record from a turntable, restore the audio, and split the recording into tracks so it can be burned to a blank CD.
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" mini-plug stereo Y-adapter 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. A "line" or "input" recording
device should be selected. Make sure that a microphone
recording device is not selected.
Step 3
Select the Record tab on the Control Properties
window, then check the
Monitor input on visuals box. This will activate
the visuals on the Control window so you can adjust volume levels
without having to record. If you cannot hear the input, turn on
recording monitoring in Windows.
Step 4
Adjust the recording volume on the
Device tab
(see Step 2 above) and go to the next step.
If you are using DirectSound mode, use the Volume tab to select and adjust the source volume. Make sure the correct Volume device is selected. It should be the same as the recording device on the Device tab. 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 (see the
"Target" area on the horizontal meters in the figure below). 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. The small rectangular boxes
on the far right of the Level Visual should not light during recording,
otherwise some clipping distortion may have occurred.
Note that the volume fader on the Control window changes the playback volume only. You must adjust the recording volume in the Control Properties window explained above.
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
Set the duration of recording.
To have more control over the channels and sampling rate used for recording
and to stop recording after a certain amount of time,
use File | New to
create a new file with the channels, sampling rate, and duration required, then use
the Record Selection
button instead in the next step. Note that the recording device may not
support the sampling rate or channels you've selected.
Step 7
Click the red Record New
button or press F9 to start recording in
GoldWave. Start playing the thoroughly cleaned, dust free album on
the turntable. Press the red Stop Recording
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. Close the file to discard the recording.
If you created a new file and used Record Selection, use Edit | Undo to undo the recording and start recording again.
When 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 | Trim | Silence 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 time line 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 | Both 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 | Cue Point | Add Cue Point 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 to write the track files to a blank CD as audio tracks.
You can import the files into iTunes or Windows Media Player to use those
programs to burn a CD. Note that you cannot
just copy the files to the CD using Windows Explorer. That will
create a data track. Each song must be written as a separate
audio track.
End of Tutorial
GoldWave does not include CD burning functionality. Use separate CD burning software such as Windows Media Player, iTunes, etc.
Many CD Recording programs can write any type of audio file, but some require a certain file type and attributes. Standard audio CDs contain PCM signed 16 bit, stereo audio at a sampling rate of 44100Hz. To save a file in that format in GoldWave, first use Resample to set the rate to 44100Hz, then use Save As to choose the "Wave" type and use the Attributes button to select "PCM signed 16 bit, stereo" attributes.
To divide a long file into separate tracks, split it into smaller files as explained above.
Troubleshooting | |
---|---|
Problem | Cause/Solution |
Cannot open large files |
|
Cannot play sounds |
|
Cannot record sounds |
|
System seems slow or visuals do not update smoothly |
|
System freezes or crashes or an exception occurs |
|
After a crash, there is less free space on the hard drive |
|
Visuals are out of synchronization |
|
Pops, clicks, or stuttering during playback |
|
Expression Evaluator slow |
|
Editing seems to be getting slower and disk activity is increasing |
|
Distortion in recording |
|
Gaps in recording and playback |
|
Sound won't play for more than a few seconds. |
|