Copyright: Copyright © 1999 - 2002 Softuarium   (http://www.softuarium.com)


Welcome

Thank you for choosing Softuarium YAMP for working with your audio files.
YAMP is a complete, professional, easy and affordable solution for computer users working with audio data. It allows to:

- easily extract tracks from audio CDs
- create samplers of audio CDs
- quickly encode WAV files to MP3 format
- tag MP3 files
- play MP3 and WAV files
- convert MP3 files back to WAV format
- encode WAV files to Ogg Vorbis format
- decode Ogg Vorbis files to WAV format
- shape amplitude envelope of WAV files
- apply filters to WAV files
- import sound files from different file formats
- professionally mix WAV files
- normalize amplitude of WAV files
- record sound from any available source
- resample WAV files

A big ‘thank you’ to registered users of previous versions of the program. Without your continuing support this release would not have happened.


CD ripper

Rippers (also known as CDDA extractors) are programs for extracting sound tracks from audio CDs and saving them to files for further processing. Audio data is extracted digitally, i.e. it is read directly from CD without using sound card.
Note: to access your CD drive YAMP requires that your system has properly installed and configured ASPI layer (set of low level utilities which allow other programs to access your CD drive). Another requirement is that your CD drive supports digital extraction.

To extract tracks from audio CD:

1. Put audio CD in drive and close tray

2. Select source CD drive (this only applies if you have more than one CD-ROM drive)
Program will try to read table of contents) from disk. If successful, list of available tracks will be displayed together with information about track length, approximate size in MB and title (if any).
Please note that Windows can be slow in retrieving information from CDs.

3. Select tracks to be extracted
Multiple selection is possible by holding ‘Ctrl’ or ‘Shift’ keys while clicking on tracks.
Before making a selection you may want to preview (listen to) tracks to make sure that you are selecting the right ones,

4. Select destination folder in which extracted tracks will be saved.
Specify folder by typing it in text box or click ‘Select destination folder’ button
Note: if selected destination folder already contains files with conflicting names YAMP will overwrite them!
Extracted tracks will be saved to files with names ‘Track_1.wav’, ‘Track_2.wav’, etc.

5. Click ‘Rip tracks’ button
Program will begin extracting selected tracks.


CD sampler

Sampler is a special kind of CDDA extractor: it extracts fragments of sound tracks from audio CDs and saves them to files for further processing (for example MP3 encoding). It is a very useful tool for musicians who want to promote their CDs on the Internet by allowing prospective customers to download samples of songs.

Note: to access your CD drive YAMP requires that your system has properly installed and configured ASPI layer (set of low level utilities which allow other programs to access your CD drive). Another requirement is that your CD drive supports digital extraction.

To extract samples of tracks from audio CD:

1. Put audio CD in drive and close tray

2. Select source CD drive (this only applies if you have more than one CD-ROM drive)
Program will try to read table of contents) from disk. If successful, list of available tracks will be displayed together with information about track length and approximate size in megabytes.
Please note that Windows can be slow in retrieving information from CDs.

3. Select tracks to be extracted
Multiple selection is possible by holding ‘Ctrl’ or ‘Shift’ keys while clicking on tracks.
Before making a selection you may want to preview (listen to) tracks to make sure that you are selecting the right ones,

4. Select destination folder in which extracted tracks will be saved.
Specify folder by typing it in text box or click ‘Select destination folder’ button
Note: if selected destination folder already contains files with conflicting names YAMP will overwrite them!
Extracted tracks will be saved to files with names ‘Track_1.wav’, ‘Track_2.wav’, etc.

5. Select sampling options
Here you can decide how much of each track you want to extract (10% - 90%). You can also choose to fade-out endings (volume will be gradually decreased to zero).

6. Click ‘Sample tracks’ button
Program will begin extracting fragments of selected tracks.


MP3 decoder

Sometimes you need to convert MP3 files back to WAV format - for example to record audio CD or for editing. MP3 to WAV converter is just the tool for the job.

To decode MP3 files:

1. Select files to convert
Add files to the list by double clicking on the list box or clicking on ‘Add files’ button.
You can also press ‘Insert’ key when list has focus or drag-n-drop files.
Please note that converter will accept any file with ‘.MP3’ extension. Files are not checked for proper format before conversion.

2. Select destination folder in which WAV files will be saved
Specify folder by typing it in text box or click ‘Select destination folder’ button.
If destination folder already contains files with conflicting names YAMP will ask for permission to overwrite.
Names of output WAV files will be the same as names of input MP3 files (except for extension).

3. Click 'Decode' button
Program will perform format conversion.
Converting MP3 files to WAV is usually faster than WAV to MP3 encoding. If, at any time, you want to cancel conversion in progress click on 'Cancel' button.


MP3 encoder

Bitrate is a parameter determining how accurately an audio file is represented while encoding or playing. It is expressed in kilobits per second (kbps). The higher the bitrate, the better is the audio quality. YAMP supports bitrates from 32 kbps up to 320 kbps.
Bitrate may have constant value. This is known as ‘constant bitrate encoding’ (CBR). This method is fast and produces files that can be played by any MP3 player. However, this is not the most efficient way of encoding. Imagine audio file with several seconds of silence in the middle. Even though there isn’t any data to encode the CBR encoder will still produce a steady stream of information at constant rate.

Bitrate may be allowed to vary. This is known as ‘variable bitrate encoding’ (VBR). This is the ‘smart’ way of encoding audio data. If audio file contains a lot of information the encoder will try to use as much of it as possible (that is it will encode it using high bitrate). If the amount of available information changes so does the bitrate. As a result the silence in the middle of the file will be encoded at minimum bitrate. This method typically produces files of better quality than CBR. However, it is slower, and some players do not support VBR-encoded files.

MP3 files can be mono, stereo or joint-stereo.
While ‘mono’ and ‘stereo’ terms are obvious, the joint-stereo (j-stereo) needs some explaining.
At very low and very high frequencies human ear is not able to locate the spatial origin of sounds with full accuracy. The MP3 encoder can therefore record low and high frequencies as monophonic signal while middle frequencies are encoded in stereo. This can reduce file size without any apparent changes in original spatial layout of sound.

To encode files:

1. Select WAV files you want to encode.
Add files to the list by double clicking on the list box or clicking on ‘Add files’ button
You can also press ‘Insert’ key when list has focus or drag-n-drop files.
Please note that the program will accept any file with ‘.WAV’ extension. However, before the encoding begins YAMP verifies that files are in supported format. If a file fails this test appropriate message will be displayed and such a file will be automatically removed from list (only PCM WAV files are supported).

2 Select a destination for MP3 files.
Specify folder by typing it in text box or click ‘Select destination folder’ button
If selected destination folder already contains files with conflicting names YAMP will ask for permission to overwrite. Names of output MP3 files will be the same as names of input WAV files (except for extension).

3. Set average encoder bitrate (CBR bitrate).
The higher the bitrate the bigger a file becomes (and better the quality). On the other hand: the smaller the file size the lower its quality. If you are just about to create your first MP3 file you may want to follow this rough guide:
- Popular music (pop, rock, etc.) sounds good with bitrates 128 - 160 kbps
- Classical music needs higher bitrates (192 - 256 kbps)
- Low quality audio (voice, AM radio, etc.) can be encoded with bitrates as low as 40 - 56 kbps

4. Optionally select variable instantaneous bitrate (VBR encoding).
This will allow you to select desired VBR quality. This quality factor determines how much freedom encoder has in selecting bitrate. Encoder first looks at the average bitrate and then combines it with VBR quality according to following rule:
- if VBR quality is set to ‘highest’ encoder is allowed to use any bitrate it wants (from 32 kbps to 320 kbps) as long as average bitrate equals the value specifies in step 3
- if VBR quality is set to ‘lowest’ encoder does not have any freedom of choice and must encode at the average bitrate specified in step 3
- values in between give the encoder more freedom in bitrate selection as the quality factor changes from ‘lowest’ to ‘highest’

5. Set desired mode (number of channels).
Obviously ‘mono’ means one channel, ‘stereo’ means two channels.
Joint-stereo (j-stereo) is a way of encoding that tricks your ears by regarding low and high frequency sounds as mono and treating mid-range sounds as stereo. As a result files smaller than full stereo can be produced while they still ‘sound like’ stereo when played.

6. Click ‘Encode’ button.


MP3 tagger

Every MP3 file may optionally contain some additional information such as author, title, etc.
This information is placed in so called ID3 tag, which is a small chunk of data (128 bytes) appended to the end of sound file. This utility allows you to read, write and/or delete such tags.

To edit ID3 tags of MP3 files:

1. Choose files to work with
Add files to the list by double clicking on the list box or clicking on ‘Add files’ button
You can also press ‘Insert’ key when list has focus or drag-n-drop files.

2. Select a file for editing
Click on the file you want to edit. If file contains valid tag it will be displayed. Otherwise status bar at the bottom of the window will show ‘Tag invalid or not present’ message.
You can preview (listen to) selected file. This is useful if file names are not very informative.

3. Edit tag
Enter or change information in tag fields.
‘Title’, ‘Artist’, ‘Album’ and ‘Comment’ fields can be up to 30 characters long.
‘Year’ field only allows 4 characters.
‘Genre’ field cannot be edited manually - you can only choose genre from existing list. To see all available genres (genera) just drop down the list and scroll. If you know roughly what genre you are after press the first letter of its name. For example: if your MP3 file is a jazz tune just keep pressing 'J' key and all genres starting with 'J' will be displayed one after another until you find 'Jazz'.

Once you have the tag set up to your satisfaction you can write it to file by clicking ‘Save tag' button. You can also clear tag by clicking 'Clear tag' button or delete tag by clicking 'Remove tag' button.


MP3 / WAV player

YAMP is capable of playing MP3 and WAV files. You can play either single files or entire playlists. Note: before a playlist is loaded it is verified. If some files are not present on your hard disk they are ignored.

During playback audio data is analyzed and frequency spectrum is displayed.

Tip: to change display style move mouse cursor over spectrum analyzer and click to display menu.

The player features:

- 10 band graphic equalizer. It allows to produce 12 dB amplification (or attenuation) of frequencies centered at 30 Hz, 60 Hz, 125 Hz, 250 Hz, 1 kHz, 2 kHz, 4 kHz, 8 kHz and 16 kHz.
Note: equalizer works only with 16 bit files. If your audio files are of the 8 bit variety changing equalizer settings will not have any effect.

- Volume controls. Player allows you to control pre-amp, wave out and master volumes.
Pre-amp slider controls the amplitude of output signal signal before it is sent to sound card. It is useful in conjunction with equalizer, because boosting certain frequencies may increase the amplitude to the point where overflow occurs (sound clipping). This is easily detected as unpleasant distortions in played sound. They can be eliminated by lowering pre-amp volume setting. Normally, setting pre-amp volume above the value of 50 is not advised, unless the equalizer is set to attenuate the signal heavily. It is always a good idea to monitor volume indicators. If they show high signal amplitude (green, yellow and red points are all visible) you may want to think about lowering volume setting.
To change overall loudness of sound use wave volume or master volume controls.

- Panning control, which determines how the sound is distributed between channels.

- Pitch control. This slider allows to change playback rate of sound (slower or faster). If sound is played faster it has higher pitch. If it is played slower than normal the pitch is lower.

- Playlist editor. To create/edit playlist click on ‘Playlist editor’ button
The editor window will appear allowing you to add files to playlist, change their order, randomize list and, optionally, save it for later use.


OGG decoder

Sometimes you need to convert Ogg Vorbis - encoded files back to WAV format - for example to record audio CD or for editing. OGG decoder can quickly batch-process your files.

To decode Ogg Vorbis files:

1. Select files to decode
Add files to the list by double clicking on the list box or clicking on ‘Add files’ button.
You can also press ‘Insert’ key when list has focus or drag-n-drop files.

2. Select destination folder in which WAV files will be saved
Specify folder by typing it in text box or click ‘Select destination folder’ button.
If destination folder already contains files with conflicting names YAMP will ask for permission before overwriting.
Names of output WAV files will be the same as names of input OGG files (except for extension).

3. Click 'Decode' button
Program will perform format conversion.
Converting OGG files to WAV is usually much faster than WAV to OGG encoding. If, at any time, you want to cancel conversion in progress click on 'Cancel' button.


OGG encoder

What is Ogg Vorbis? At the risk of repeating what you already know: it is one of the best second-generation compressed audio codecs. This is already true even though the software is in pre-release stage (current version is 1.0 Release Candidate 3). Given a little bit more time to mature it will most likely become the best compression/decompression tool for audio data. It offers a lot of exciting features not available anywhere else (such as bitrate peeling, flexible tagging, support for up to 256 channels, and so on). Best of all, Ogg Vorbis is an open source project and it is patent and royalty free. This implementation of Ogg Vorbis encoder does not include all these exciting features, but already it produces smaller and better sounding files than mp3 encoders.
Unfortunately YAMP is not able to play ogg files yet (we are working on this), but plugins exists that allow ogg files to be played by WinAmp or Windows Media Player. Also, increasing number of hardware players support Ogg Vorbis standard.

To encode files:

1. Select WAV files you want to encode.
Add files to the list by double clicking on the list box or clicking on ‘Add files’ button
You can also press ‘Insert’ key when list has focus or drag-n-drop files.
Currently only 16 bit, stereo PCM WAV files are supported (sampling rates from 8 kHz to 48 kHz).

2 Select a destination for OGG files.
Specify folder by typing it in text box or click ‘Select destination folder’ button
If selected destination folder already contains files with conflicting names YAMP will ask for permission to overwrite. Names of output OGG files will be the same as names of input WAV files (except for extension).

3. Select encoding mode. You can use either quality setting (default and recommended) or bitrate setting.

In default mode the required bitrates are set and managed internally by the encoder to best fit input audio data. You can set the desired output quality by changing slider position from 0 % to 100 %. Normally, settings in the range of 30 - 50 % produce excellent sounding files.

In controlled bitrate mode you are required to set minimum, nominal and maximum bitrates using the sliders. Please note that nominal bitrate must have a value between minimal and maximal bitrates - this is reflected in the behaviour of the sliders.
Generally, using this mode is discouraged, because it does not give the encoder the freedom necessary to produce best quality audio. However, it gives you more control over the final size of output files. Please note that you may take this option to the extreme by setting minimum, nominal and maximum bitrates to be equal. It will produce almost constant bitrate in output files.

4. Click ‘Encode’ button.


WAV envelope editor

Envelope shaping is a process in which amplitude of all samples in a WAV file is modified to fit a particular, user-defined profile. This allows to create effects such as fade-in, fade-out, on/of, etc. It can also be used to uniformly change volume (loudness) of sound file.

Envelope shaping does not alter size, sampling frequency, sample size or number of channels of a sound file. It simply changes the overall loudness of sound as a function of position in file.

To shape envelope of files:

1. Select files to be processed
Add files to the list by double clicking on the list box or clicking on ‘Add files’ button.
You can also press ‘Insert’ key when list has focus or drag-n-drop files.
As files are added they are checked for proper format. If a file is not in PCM format it will be rejected.

2. Select destination folder in which processed files will be saved
Specify folder by typing it in text box or click ‘Select destination folder’ button.
If selected destination folder already contains files with conflicting names YAMP will ask for permission to overwrite. Names of processed files will be the same as names of input files.

3. Create or load from file desired envelope curve
Current envelope setting is displayed in 'Current shape' preview area. If you want to use it as it is just click on 'Apply' button and program will process files selected in step 1.
If you want to modify envelope shape please click on 'Edit' button. A new window will appear allowing you to edit envelope points. Working with points is very easy if you remember the following rules:

- create a new point by double-clicking (this causes curve to snap to new point; if you want the new point to snap to curve hold down CTRL button when double-clicking)
- if you want to remove a point double-click on it while holding down CTRL key
- you can drag a point to change its position
- if you draw a rectangle around several points you can drag them as a group

You can also load envelope curves from files and save them to files for later use.
Once you've created a profile you are happy with click 'OK' button. The preview area will be update to reflect current envelope shape.

4. Click on 'Apply' button to start processing


WAV filter

This utility allows you to apply filter to audio files. There are four different filters that you can use: low pass, high pass, band pass and band stop. Filter order (also known as number of taps) and cut-off frequencies can be adjusted so that the filter response precisely matches your requirements. Filter response is displayed graphically (in either dB or % scales).

Filtering audio files can be useful in many situations, the most typical being:

- eliminating high frequency noise (hiss) with low pass filter
- eliminating low frequency noise (mains hum) with high pass filter
- extracting midrange audio data for transmission over narrow channel with band pass filter
- removing particular frequency bands from audio data with band stop filter

To filter WAV files:

1. Select files to be processed
Add files to the list by double clicking on the list box or clicking on ‘Add files’ button.
You can also press ‘Insert’ key when list has focus or drag-n-drop files.

2. Select destination folder for filtered files
Specify folder by typing it in text box or click ‘Select destination folder’ button.
If selected destination folder already contains files with conflicting names YAMP will ask for permission to overwrite. Names of filtered files will be the same as names of input files.

3. Set required filter type (low pass, high pass, band pass or band stop)

4. Set filter order (number of taps) so that filter response matches your requirements.
By default response is shown in %, but you can switch to dB view by clicking on the button in upper right corner of the display.
Please note that while increasing filter order usually improves its response, it also increases the number of calculations required. Typical range of filter taps is from about 25 (simple filter) to about 125 (if really ‘square’ response is needed).

5. Set cut-off frequencies
In case of low pass and high pass filters only one cut-off frequency is required. Both band pass and band stop require two cut-off frequencies specified (low and high). Of course, low frequency must be smaller than high. Note: it is important to remember that low and high cut-off frequencies cannot be too close. Anything less than several hundred Hz will most likely not work. Of course, there is nothing to stop you from experimenting :-) In such case it helps to remember that even very small change of filter order may have dramatic impact on filter response. For example, try following: band pass filter with low cut-off frequency = 9900 Hz, high cut-off frequency = 10 000 Hz. Filter order less than 31 will not produce any response. At 31 taps you’ll see quite acceptable response peak, further increase of number of taps will not help unless you set it to 53, and so on. On the other hand, if you set low cut-off frequency to, say, 9000 Hz, filter response will become much more stable (for wider range of filter order values).

6. Click ‘Filter’ button


WAV import

This utility allows to import to PCM WAV files recorded in other formats. Following formats are supported:

- Microsoft WAV 8, 16, 24 and 32 bit integer PCM.
- Microsoft 32 bit floating point PCM.
- Microsoft IMA/DVI ADPCM WAV format (16 bits per sample compressed to 4 bits per sample); mono and stereo only.
- Microsoft ADPCM WAV format (16 bits per sample compressed to 4 bits per sample); mono and stereo only.
- Microsoft 8 bit A-law an u-law formats (16 bits per sample compressed to 8 bits per sample); single or multi channel.
- Microsoft WAV files using GSM 6.10 encoding; mono only.
- Apple/SGI AIFF and AIFC uncompressed 8, 16, 24 and 32 bit integer PCM.
- Apple Macintosh OSX AIFC files with little endian PCM data (16, 24 and 32 bit).
- Apple/SGI AIFC files containing 32 bit floating point data.
- Sun/NeXT AU/SND format (big endian 8, 16, 24 and 32 bit PCM, 8 bit u-law and 8 bit A-law).
- Dec AU format (little endian 8, 16, 24 and 32 bit PCM, 8 bit u-law and 8 bit A-law).
- G721 and G723 ADPCM encoded AU/SND files (mono only).
- Amiga uncompressed IFF / 8SVX / 16SV PCM files (8 and 16 bit) (mono only).
- Ensoniq PARIS big and little endian, 16 and 24 bit PCM files (.PAF) .
- Sphere NIST 16, 24 and 32 bit PCM files. Big and little endian supported.
- Propellerhead Reason REX files
- IRCAM (Berkeley, CARL) files with 16 and 32 bit PCM, A-law, u-law and float data.

To import files:

1. Select files to be processed
Add files to the list by double clicking on the list box or clicking on ‘Add files’ button.
You can also press ‘Insert’ key when list has focus or drag-n-drop files.
Files are not verified when they are added to list. However, you can check if a file is in supported format by clicking on it. If program can read it then short file info is displayed (sampling rate, number of channels, sample size and number of samples). If program doesn't know how to read a file all fields say 'unable to read file'.

2. Select destination folder in which imported files will be saved
Specify folder by typing it in text box or click ‘Select destination folder’ button.
If selected destination folder already contains files with conflicting names YAMP will ask for permission before overwriting. Names of processed files will be the same as names of input files except for extension (.wav).

3. Click on 'Import' button to start processing


WAV mixer

Mixer is a tool allowing to create interesting effects by mixing sounds from two sources (files). You can use two different files or produce output from one source played in both inputs simultaneously. The following sound attributes can be changed: volume, panning, pitch and start position. Optionally, you can also loop input files. Mixer can be used to produce real time effects and/or save output to file.

The mixer window consists of three panels: Input 1, Input 2 and Output.

Input 1 and Input 2 panels are identical and feature:

- Activity indicator. The LED is on if input is active, off otherwise.
- Current position indicator. It shows current position in file using standard time format xx : xx : xxx
- Buttons ‘File’ (for loading file you want to play) and ‘Clear’ (removing loaded file)
- Start position edit box. If left as default 00:00:000 it causes input file to be played from the beginning. Setting it to other value determines when playback will start.
Tip: you can produce an interesting effect by loading the same file in Input 1 and 2 and setting start position on one panel a fraction of a second (say, 00:00:100) ahead. Now click ‘Start’ button and listen to your first ‘special effect’ :-)
- Loop checkbox. If you work with short files you may want to ‘loop’ them, i.e. start playing them again immediately after they are finished.
- Volume slider. It controls the amplitude of signal from respective input before it is mixed. If both inputs are active, if input files are normalized to maximum possible amplitude and if input volumes are set to values over 50 most likely the output signal will become distorted due to overflow (clipping). To avoid this it is best to keep input volume settings below 50. You can always normalize the output file after mixing is finished.
- Panning slider, which determines how the sound is distributed between channels.
- Pitch slider. This slider allows to change playback rate of sound (slower or faster). If sound is played faster it has higher pitch. If it is played slower than normal the pitch is lower.

Output panel consists of:

- Activity indicator. The LED is on if mixer is active, off otherwise.
- Current position indicator. It shows mixer active time using standard time format xx : xx : xxx
- Display panel. It shows simplified frequency spectrum of output and volume levels of both channels. It also shows what, if any, file is output being saved to.
- ‘File’ button. Use it to select file to which output will be saved. The output file is in PCM WAV format (44100 Hz, 16 bit, stereo) format. This setting cannot be changed.
- ‘Start’ and ‘Stop’ buttons. Use them to start and stop the mixing process. Note: if you do not use output file, the maximum length of time mixer can be left running is approximately 3.5 hrs. This is due to limitations of operating system. If you save the output to file, the limiting factor may turn out to be the amount of free space available on your hard disk.
- Volume slider. It controls the amplitude (sample values) of output signal. It should not be used to increase/decrease loudness of playback. This can be achieved by adjusting master volume of you sound card , which increases loudness without affecting actual sample values.
It is always a good idea to monitor volume level indicators. If they show high signal amplitude (green, yellow and red points are all visible) you may want to think about lowering volume setting to avoid distortions (clipping).
- Panning slider, which determines how the sound is distributed between channels.


WAV normalizer

Normalizing WAV files means adjusting their amplitudes (loudness). If you have audio files from different sources you’ll most likely find that some of them are louder than others. If you want to use them together (for example to make audio CD) normalizing their amplitudes may be necessary.
There are two ways of adjusting loudness: using peak level values (used by YAMP) or average RMS signal power. While the second method is more closely related to perceived loudness of sound, it also distorts dynamic range of the signal more.
Normalization is a two-pass process. In first pass sound file is scanned to detect maximum instantaneous sample values. Then a gain (amplification or attenuation factor) is calculated and in second pass the gain is applied to all samples.
You should keep in mind that normalization is a process that does not necessarily improve sound quality. It distorts dynamic range of sound file, which becomes ‘stretched’ or ‘squashed’. Also, any noise that audio file may contain will be amplified at the same rate as signal.

To normalize WAV files:

1. Select files to be processed
Add files to the list by double clicking on the list box or clicking on ‘Add files’ button.
You can also press ‘Insert’ key when list has focus or drag-n-drop files.
As files are added they are checked for proper format. If a file is not in PCM format it will be rejected.
When you add a file you’ll see that columns ‘Min’ and ‘Max’ display question marks. This is normal. It means that file has not been scanned for maximum and minimum sample values yet. This takes place when you click ‘Normalize’ button.

2. Select destination folder in which processed files will be saved
Specify folder by typing it in text box or click ‘Select destination folder’ button.
If selected destination folder already contains files with conflicting names YAMP will ask for permission to overwrite. Names of normalized files will be the same as names of input files.

3. Set normalization parameters
You can change sound amplitude from 1% to 100%.
If you want to produce files that are as loud as possible (without clipping) select 100% of ‘Maximum allowed amplitude’.
If you want to decrease loudness select value smaller than 100% of ‘Current amplitude’.
Of course, there is nothing to stop you from setting 100% of ‘Current amplitude’, but this will not result in any changes to loudness.

4. Click ‘Normalize’ button
Program will begin processing files.
In first pass it will detect maximum and minimum sample values and display them in ‘Min’ and ‘Max’ columns (these values are in range [-32768..32767] for 16-bit audio and [-128..127] for 8-bit audio). Program will then calculate gain based on parameters set in step 3 and detected sample values. In second pass this gain will be applied to sound file.


WAV / MP3 player

YAMP is capable of playing MP3 and WAV files. You can play either single files or entire playlists. Note: before a playlist is loaded it is verified. If some files are not present on your hard disk they are ignored.

During playback audio data is analyzed and frequency spectrum is displayed.

Tip: to change display style move mouse cursor over spectrum analyzer and click to display menu.

The player features:

- 10 band graphic equalizer. It allows to produce 12 dB amplification (or attenuation) of frequencies centered at 30 Hz, 60 Hz, 125 Hz, 250 Hz, 1 kHz, 2 kHz, 4 kHz, 8 kHz and 16 kHz.
Note: equalizer works only with 16 bit files. If your audio files are of the 8 bit variety changing equalizer settings will not have any effect.

- Volume controls. Player allows you to control pre-amp, wave out and master volumes.
Pre-amp slider controls the amplitude of output signal signal before it is sent to sound card. It is useful in conjunction with equalizer, because boosting certain frequencies may increase the amplitude to the point where overflow occurs (sound clipping). This is easily detected as unpleasant distortions in played sound. They can be eliminated by lowering pre-amp volume setting. Normally, setting pre-amp volume above the value of 50 is not advised, unless the equalizer is set to attenuate the signal heavily. It is always a good idea to monitor volume indicators. If they show high signal amplitude (green, yellow and red points are all visible) you may want to think about lowering volume setting.
To change overall loudness of sound use wave volume or master volume controls.

- Panning control, which determines how the sound is distributed between channels.

- Pitch control. This slider allows to change playback rate of sound (slower or faster). If sound is played faster it has higher pitch. If it is played slower than normal the pitch is lower.

- Playlist editor. To create/edit playlist click on ‘Playlist editor’ button
The editor window will appear allowing you to add files to playlist, change their order, randomize list and, optionally, save it for later use.


WAV recorder

Recorder is a utility allowing you to create audio files from any audio input device (such as microphone) connected to your computer. It also can be used to ‘capture’ sounds played by your computer. For example, there are many Internet web pages that play sound when you visit them. If you wish to convert this audio data to a file on your hard disk recorder is just the tool for the job.
Recorder can also be set to work in unattended mode. You can program it to start recording at any time for up to one year from current date.

To manually record sound file:

1. Select file to which sounds will be saved (‘File’ button)
Recorder does not have any limit on length of file, but obviously it is limited by available free space on hard disk.

2. Click ‘Record’ button.
Recorder will start saving data to previously selected file. Current length of file will be continually updated and displayed.
At any time you can pause/resume recording. When finished recording click ‘Stop’ button. Your file will now be ready for playback or further processing.

To set recorder's timer:

1. Select file to which sounds will be saved (‘File’ button)

2. Click ‘Timer’ button.
A new window will appear allowing you to see current date and time and set necessary timer options.
You will need to:

- Set date on which you want to start recording. You can program recorder for up to one year from current date.
- Set time at which you want to start recording.
- Set desired length of recording. You can set it to a maximum of 24 hours if your hard disk has enough free space.
As you change this parameter you will see required amount of space and current free space available displayed. Please note that recording length must be set to value that leaves at least 10% of currently available disk space.
- Once these steps are completed click on 'Set timer' button. The recorder will now be ready. It will start recording at the date and time set in previous step and will stop recording once the length of the recording matches the set value.

Very important: once you set the timer for recording you must remember not to use any other YAMP tools and not to exit YAMP. This will reset timer to its default inactive state. In other words: set the timer and leave the program alone until it finishes recording.

Optionally, before recording you might want to:

A. Configure your sound card for recording (‘Volume’ button)
When ‘Volume Control’ window is visible do the following:

To display specific devices
- On the Options menu, click Properties
- Click Recording and make sure that the check box next to the device you want is selected

To adjust the recording volume
- On the Options menu, click Properties, and then click Recording
- Make sure that the device you want to adjust the volume for is selected, and then click OK
- Drag the Volume slider up to raise the volume, or down to lower the volume
- To change the balance between the left and right speakers, drag the Balance slider

B. Select sound format (‘Format’ button)
By default YAMP records sounds as PCM WAV files (22 050 Hz, 16 bit, stereo). You may want to change this to some other format. The list of available formats depends on what audio codecs are installed on your computer.


WAV resampler

Resampler is a utility that lets you change sampling rate and sample size of existing sound files. When changing to a higher sample rate, extra samples are interpolated and the file size increases. When changing to a lower sample rate, some samples are removed and the file size decreases.

Important thing to remember is that resampling, while sometimes necessary, tends to decrease audio quality. If you are prepared to trade quality for smaller file size then down-sampling is fine. For best results try to use new sampling rate that is an even multiple of old one (such as resampling from 44.1 kHz to 22.05 kHz). Up-sampling will not improve sound quality. It will however, increase file size.

Resampler supports sample sizes from 8 to 32 bits (8, 16, 24, 32 on input and 8, 16, 24 on output), but please keep in mind that Windows is able to play only 8 and 16 bit WAV files.

To resample WAV files:

1. Select files to resample
Add files to the list by double clicking on the list box or clicking on ‘Add files’ button.
You can also press ‘Insert’ key when list has focus or drag-n-drop files.
As files are added they are checked for proper format. If a file is not in PCM format it will be rejected.
When you add a file you’ll see its current sampling rate and sample size displayed in columns ‘Rate’ and ‘Size’ , respectively.

2. Select destination folder in which resampled files will be saved
Specify folder by typing it in text box or click ‘Select destination folder’ button.
If selected destination folder already contains files with conflicting names YAMP will ask for permission to overwrite.
Names of resampled files will be the same as names of input files.

3. Set new sampling rate, sample size and required dithering mode

4. Optionally, you may also choose to
    a) use two-pass resampling to avoid clipping distortions
    b) normalize amplitude while resampling (this will force a two-pass run)
    c) use fast resampling (approximately twice as fast as default setting, but resulting in lower quality)

5. Click ‘Resample’ button


License information

Before using Softuarium YAMP please read carefully terms and conditions of this agreement. By using this software you are consenting to be bound by and are becoming a party to this agreement. If you do not agree to all terms of this agreement do not use the software and destroy all copies of the software from your computer.

1. Softuarium (the company) grants you (the customer) a worldwide, non-exclusive, personal license to install and use the accompanying YAMP program (the software) in the manner provided herein.

2. The software is licensed, not sold. The software is protected by copyright law, international copyright treaty provisions and other intellectual property laws and treaties. Customer must not allow copies of the software or any part of it to be made or distributed to anyone else. Customer may make backup copies of the software for archival purposes only.

3. The software is licensed for use on limited number of computers, depending on the type of license purchased. Customer may delete the software from one computer and reinstall it on another, but customer may not install the software on more computers than the license allows for.

4. The software may not be rented, leased, nor in any other manner commercially exploited without the company's prior written permission.

5. The company makes no representations or warranties about the suitability of this software for any purpose. The software is provided 'as is' without express or implied warranties, including warranties of merchantability and fitness for a particular purpose or non-infringement. The company may not issue updates to this software.

6. All audio content accessed through the software is the property of the applicable content owner and may be protected by applicable copyright law. This license gives customer no rights to such content, and company disclaims any liability for misuse of content.

7. Title, ownership rights, and intellectual property rights in and to the software shall remain with the company. Customer agrees not to modify the software or attempt to decipher, decompile, disassemble or reverse engineer the software. Reproduction and/or redistribution of any portion of the software is prohibited without the company's prior written permission.

8. The company may terminate this license at any time by delivering notice to customer and customer may terminate this license at any time by destroying or erasing customer's copy of the software. Upon termination of this license, customer agrees to destroy all copies of the software. In case of termination sections 5 through 10 of this agreement shall indefinitely survive its termination. This license is the complete statement of the agreement between the company and customer and merges and supersedes all other or prior understandings, purchase orders, agreements and arrangements.

9. The company or its suppliers shall not be liable for direct, incidental, consequential, special or indirect damages of any sort whatsoever. This includes loss of business profits, revenues, information, or business interruption resulting from customer's use of the software or inability to use the software, even if the company has been informed of the possibility of such damages.

10. This software is not fault-tolerant. It is not designed, manufactured or intended for use in hazardous environments requiring fail-safe performance. The company and its suppliers specifically disclaim any express or implied warranty of fitness for high-risk activities.


Credits

This program contains portions of code owned by SwiftSoft Inc. (http://www.swiftsoft.de)

This program uses libraries from v1.0 RC3 distribution of Vorbis SDK Copyright (c) 2001, Xiphophorus
The SDK is available from http://www.xiph.org/ogg/vorbis

This program uses, or contains code based on following software released under the LGPL (GNU Lesser General Public License):

- GOGO-no-coda v. 2.33 (Copyright 1999 PEN@MarineCat, Herr Mie)
  Source code is available from http://homepage1.nifty.com/herumi/soft/gogo2/src

- Libsndfile v0.27 written by Erik de Castro Lopo (erikd@zip.com.au)
  Original source code is available from http://www.zip.com.au/~erikd/libsndfile
  Modified source code is included in Yamp's installation folder (sub-directory 'Misc')

- SSRC v1.29 written by Naoki Shibata (shibatch@users.sourceforge.net)
  Original source file is available from http://shibatch.sourceforge.net
  Modified source code is included in Yamp's installation folder (sub-directory 'Misc')
  SSRC, in turn, uses freeware FFT and Bessel functions written and copyrighted (1996) by Takuya Ooura (ooura@mmm.t.u-tokyo.ac.jp)

Full text of GNU Lesser General Public License is included in Yamp's installation folder (sub-directory 'Misc', file 'lgpl.txt').
Full text of Vorbis SDK redistribution and use conditions is included in Yamp's installation folder (sub-directory 'Misc', file 'vorbis.txt').


Contact us

Your primary source of information about the program is this help file. It should teach you how to use the program and answer most of your questions. If solutions to your problems are not found here please contact support@softuarium.com. We are always happy to help, but can do it most efficiently if you:

- make sure that help file or FAQ does not have the information you request
- tell us which version of the program you use
- tell us something about your computer (version of operating system, multimedia hardware)
- tell us what you are trying to do (step by step) and what results are produced (listing error messages, if any). Just saying 'It doesn't work !' is not enough for us to diagnose a problem.

We encourage you to join our mailing list (if you are a registered user your e-mail address is added to mailing list automatically). This is the best way to stay informed about new releases, bug fixes and other relevant topics. The list has low traffic and there are no out of topic messages. Your e-mail address is kept confidential at all times.
To join the list please send message to mailinglist@softuarium.com (put ‘subscribe’ in message body).

With any questions related to licensing, ordering and pricing please contact our sales department: sales@softuarium.com

Of course we are always happy to get feedback from our customers. Tell us what you find useful and good, what is missing or badly designed. Above all, tell us what you would like to see in next versions. We have plans of our own, but are open to suggestions.
If you have an idea how to make this program better please send a message to feedback@softuarium.com


Pricing and ordering

Current version of YAMP is sold for $30 (single user license).
If you are interested in purchasing many licenses please contact us for information on bulk discounts.
The address is sales@softuarium.com

You can securely buy your retail version of YAMP online (please click here).
We accept all major credit cards.

If you prefer more conventional methods such as mail, fax and phone orders please contact us at sales@softuarium.com for more details.

Our sales department can also answer your questions about becoming an affiliate or OEM partner.


FAQ

I can't get the CD rip function to work with home made audio CDs (it works fine with original audio CDs).
I have a big collection of MP3 files. Can I convert them to Ogg Vorbis format ?
What are the limitations of demo version?
If I buy the program, may I load it on to 2 computers or do I have to pay twice?

I can't get the CD rip function to work with home made audio CDs (it works fine with original audio CDs). Most obvious cause of such behavior would be improperly written table of content (TOC) on disk. However, this is not very likely - your CD-recording software would have alerted you about unsuccessful completion of burn process.

It might be caused by trying to rip disk that was written in TAO (track at once) mode instead of DAO (disk at once) mode typical for audio CDs.

Most likely, however, the problem is this: CD-ROM readers do not perform error correction of the data when doing raw sectors reads (as in digital audio extraction). As a result, ripping software crashes even though CD can be played perfectly well (because then error correction feature is switched on).
When a program does "raw" sector reads, it gets the entire 2352-byte block, which includes the CD-ROM error correction data (ECC) for the sector. Instead of applying the ECC to the sector data, the drive just hands back the entire block, including any errors that might have been written to disk - and errors are present very often (most commonly they are caused by dropped-sample interpolation). Normally this not enough to hear the difference when playing such a disk, but enough to cause problems for reading software.
To make the story short: if you rip home made audio CDs you do it at your own risk. Some will work, some will not.

I have a big collection of MP3 files. Can I convert them to Ogg Vorbis format ? Technically, yes - it is very easy. Just decode MP3 to WAV and then encode WAV to OGG. However, transcoding is a very bad idea because audio quality is lost in the process. Remember that both MP3 and OGG are lossy compression algorithms. When source file was encoded as MP3 some of the original information was lost/discarded. While decoding MP3 to WAV is lossless, the missing information is not re-created. Then, is you encode such a WAV file again, more information will be lost/discarded and quality will be lower. To use a simple analogy: imagine that you want to duplicate an audio tape. First copy will not be too bad. However, if you make a copy of a copy it will not sound so good any more.
To summarize: transcoding using lossy algorithms decreases quality. This applies to all possible conversions between formats such as MP3, OGG, VQF, WMA, and so on.

What are the limitations of demo version? The only limitation of demo version is that it stops working after 100 executions. It does not 'cripple' files in any way.

If I buy the program, may I load it on to 2 computers or do I have to pay twice? That depends:
You only need one license if you have many computers but you are the only user, or if there are many users but program is installed on only one computer.
If there are many users and computers (in other words multiple copies of program are used at the same time) then you need appropriate multi-user license.