Audacity is a free audio editor. You can record sounds, play sounds, import and export WAV, AIFF, Ogg Vorbis, and MP3 files, and more. Use it to edit your sounds using Cut, Copy and Paste (with unlimited Undo), mix tracks together, or apply effects to your recordings. It also has a built-in amplitude envelope editor, a customizable spectrogram mode and a frequency analysis window for audio analysis applications. Built-in effects include Echo, Change Tempo, and Noise Removal, and it also supports VST and LADSPA plug-in effects.
Audacity is being developed by a team of volunteers under the open-source model. It is written in C and C++, using the wxWidgets cross-platform toolkit. All of the source code to the program is made available under the GNU General Public License, which essentially allows anyone to modify the source code as long as they publicize the changes. The code is hosted on SourceForge.
Questions/comments? Send email to mailto:audacity-help@lists.sourceforge.net
Download Audacity form the sourceforge url above. Follow the online instructions to install.
In addition to Audacity this tutorial also requires the files VST Enabler.dll, lame_enc.dll and VoxengoEssEq.dll. These plugins allow a number of important additional functions not included in the Audacity license. The file lame_enc.dll alows the creation and editing of mp3 files; the file VST Enabler.dll enables VST plugins to be used with Audacity; and the file VoxengoEssEq.dll is a 7 band equalisation VST plugin. Use the links below to download and install.
Although Audacity includes many useful features, the basic setup lacks record level metering. Freeware audio level meters designed by Paul Marshall acn be downloaded using the link below.
There are some specific aspects of working with older open reel tapes that you need to be aware of. There are also some special tools that you will find useful. In recent years the availability of working reel to reel players has diminished and even basic accessories have become hard to come by, but the more of these tools you can locate, the better your results will be. Useful tools would include a splicing block, removable splicing tape, demagnetized scisors, razor blades and Mylar leader tape. For auditioning material a good playback machine would be something like a Technics RS-1500 half-track open reel recorder with quarter and half-track playback. The open loop design on this machine treats older tape very gently, when repeated playback may be necessary. For handling the tape it is advisable to wear soft white cotton gloves. The natural oils in your skin can damage the tape.
The first job in preparing to play an older tape will be to check the leader tape and replace any weak splices. Also watch for any wrinkles on old tape, these may get worse and cause the tape to weaken with repeated playing. To rectify this apply splicing tape to the backside to strenghthen any weak points.
Some older tapes can be very fragile. Until the audio is preserved to some other medium, extreme care must be taken in protection and storage. More recent tapes will be made from Mylar, which will stretch. Older tapes will be made using an acetate base with a low tensile strenghth, which can be prone to breaking. Acetate tape is also susceptible to linear expansion in humid or warm conditions. This is easily recognisable when looking at the reel on its side. If not stored properly, the tape will look like spokes on a wheel with bumps on the tape along its circumference. To reduce this problem, it is necessary to store the tape with what is known as an "archival wind". Play the tape slowly so that air pockets between the layers do not form, so succesive layers will be placed evenly on top of one another. Always use this procedure when storing important tapes.
It is important also to consider your tape deck. This should be thoroughly cleaned prior to playback. All parts of the tape path should be cleaned, including playback heads, guides, capstans and pinch rollers. Some specialists recomend the use of a tape lubricant. It may also be necessary to degausse playback heads from time to time.
Once the tape has been threaded onto the machine and levels have been set, It is good practice to record even this first trial run. In some circumstances this may be the only chance you get to record a very poor quality tape. As the tape plays for the first time watch for any weak splices. Older splicing tapes were acetate based and may have dried out causing the tape to separate as it goes through the guides. Replace the splices with Mylar, which will not creep or bleed around the splice. The contact between tape ends is stromg, yet it can be removed if needed.
After all the splices have been replaced, reverse the reels and play the tape back to the begining. Now you have tape that is evenly wound, with little or no tape left exposed, reducing the risk of edge damage.
One of the major problems you will encounter in working with older tapes is 'tape squeal'. This will manifest itself as a loud and constant squealing noise that can be heard through the speakers and from the tape machine itself. Along with the squeal, the recording will loose all it's high frequencies, sounding dull or unintelligible. This is caused by the tape shedding the oxide coating and leaving deposits on all parts of the deck that contact the tape. The tape will eventually stop in its tracks, not only refusing to play, but unable to fast forward or rewind.
It is possible to prepare tapes to prevent shedding by baking the tape in a convection oven. If you do not have the time or the funds necessary to purchase a convection oven, or your collection is too small to justify buying one the only way to resolve the problem is too clean the tape deck and try playing the tape again. It may be necessary to repeat this process many times with some older and longer recordings. Each playback should be recorded. If a full playback cannot be achieved, then it may be possible to edit these shorter recordings together, to salvage the project.
After you have played the tape and transferred its contents to the new medium, it is important to consider the future storage of that tape. A sound storage environment is vital. An ideal situation would be to store the tape in a vertical psoition in a plastic container on a solid wood shelf. The ideal temperature to store tape when not in use is between 55 and 75 degrees F, with humidity maintained betwwen 30% and 40%.
If you look on the back of your computer you will find the soudcard I/O panel. This will look something like the diagram below. Look carefuly at the sockets and you will see that they are named. You will probably have something like 'line in', 'Mic in' and 'spkr/line in' written next to them. (Names may vary according to manufacturer and model, but this set up is fairly standard.) Most soundcards only have mini-jack inputs, so you will have to make sure that you have the right cables to hook up your input device.
soudcard I/O panel
Your input device needs to be connected to the line in socket. Any analogue device can be fed in to this input. Cassette decks and Reel-to-reel tape recorders need no other devices, but record decks will need some sort of pre-amp. If you are using more than one source you might consider using some sort of mixing device so that you don't have keep changing the connections. Another solution might be to use a hi-fi amplifier with multiple I/O connections.
diagram showing analogue to digital recording setup In this simple setup the tape recorder (analogue source) is connected directly to the soundcard line input. Computer monitors are used for audio playback.
To hear your audio you will need to connect some speakers. If these are not powered you will need to add an amplifier as well. Look on the diagram to see how to set this up.
In this more complex setup the tape recorder (analogue source) is connected to a mixing desk which is connected to the soundcard line input. Audio playback is achieved via an external amplifier with additional speakers.
With regard to speakers, it is important to remember that home stero speakers are built to sound good, not to sound accurate. They tend to boost the lows and the highs. Desktop multimedia speakers are even worse. To get aclear idea waht you've recorded, you'll need to invest in a pair of studio-quality speakers and a matching amplifier.
In this setup the tape recorder (analogue source) is connected to a hi-fi amplifier, with multiple I/O, which is connected to the soundcard line input. Audio playback is achieved via the same amplifier, with additional speakers.
The room in which you listen will add its own coloration to the recording. When a professional is evaluating a mix, he or she often makes a copy to CD, DAT or anologue cassette and listens to it in various environments. One room may swallow the bass while another swallows the high mid range. If you only plan to listen to your recordings on your own system, in your own room, then by all means mix so that it sounds best in thet environment. If you plan to distribute your recordings, don't make unnecessary assumptions about the quality of your monitoring system.
Digitization of audio is an automatic process carried out by the analogue-to-digital convertor (ADC) component of your computer soundcard.
Two main factors determine an ADC's performance - sample rate (the number of times per second at which an audio signal is sampled) and resolution (the number of levels used to describe its amplitude).
The sample rate determines the range of frequencies an ADC can adequately handle, and it is expressed in samples per second, or kHz - '44.1kHz' mans 44,100 samples per second. An ADC's sample rate should be at least double the highest frequency it is required to convert, otherwise it will 'alias' - ie. produce whistling tones that are harmonically related to the sample frequency. (The man who made this discovery was called Nyquist, and the upper limit of frequencies that can be properly recorded - around 20kHz in the case of CD standard digital audio - is therefore refered to as the Nyquist frequency.)
Quantisation resolution can be exprssed either in bits, or as a number of discrete levels. As a bit can only have two values (0 or 1), an 8-bit ADC can describe 2 to the power 0f 8 (= 256) levels. A 16-bit system would yield 65,536 levels (2 to the power of 16). The greater the resolution of a convertor, the more accurate will be the representation of a sound.
The diagram shows a sine wave being sampled with a resolution of 16 levels - a 4 bit convertor. The input sine wave has a frequency of 1kHz, therefore its period (1/1kHz) is 1ms, during one period, 20 values are sampled, and the sample rate is therefore 20kHz (one sample is taken every 50 microsconds.)
The major standard for digital audio is 16-bit quantisation at 44.1kHz or 48kHz, which allows recording of the full range of frequencies that the average humen ear can detect, and offers a maximum 96db signal to quantisation noise ratio.
To digitise archive audio using Audacity the following steps must be followed.
Input volume must be adjusted. To do this launch the Audio Level Meter using either the program menu or from desktop icon. The meter will give a visual indication of the changing levels of audio input.
To make make changes to this level you will need to use the Record level volume control this can be found from the program menu:
Start/Programs/Accessories/Multimedia/Volume Control
or by Doubleclicking the 
icon in the system tray.
volume control
volume control - properties
Once this is opened you need to find the recording conrol. From the options menu select properties and then select recording. When the page opens select line in. With your audio source playing, watch the meter and adjust the levels until the average signal is just touching the red area at the top. Don't let the volume go beyond this level. If it does the box at the top will be lit and the level must be adjusted. If the recording levels are set too high poor quality recordings will result. If the level is set too low this will give a poor signal-to-noise ratio and result in similarly disapointing recordings.
recording control
Now you must open Audacity
Before you start recording it is important that you check and set the software preferences.
From the file menu select preferences. From the Audio I/O tab make sure that the selection for channels reads - 2(stereo); from the Quality tab make sure that the selection for Default Sample Rate reads - 44100Hz; from the File Formats tab make sure that the selection for Uncompressed Export Format reads - WAV (Microsoft 16bit PCM). It is also worth having a look at the keyboard and mouse commands page to learn some of the software shortcuts. As you become more familiar with the software you may want to change these, or set up shortcuts of your own.
preparing to record with Audacity
recording with Audacity
To start recording select 'line in' from the drop down menu and click the red button on screen or press keyboard R to record. Next start your audio source. To stop recording hit spacebar or keboard S. Watch the levels as you record. If the meter shows that the levels have gone over the top reduce the level on the line-in recording control and record again. When recoring is complete save file choosing Export_as_WAV from the file menu. Playback the file using spacebar for start/stop or keyboard S to stop.
With your audio source digitised and saved, it is time to edit the recording in preparation for archiving and/or delivery.
To start the editing process open Audacity 1.2.0. From the project menu select Import audio and open your file. Keyboard command Ctrl+Igives the same option. To adjust the screen view, from the view menu select Fit in window (Ctrl+F).
The first job is to trim the file. Your recording will probably include some dead time at the begining and end, and may include other unwanted elements. Select the section you want to keep by dragging the mouse then save this selection as a new file, choosing Export_selection_as_WAV from the file menu. This is a useful technique if you want to break a long file into several smaller sections. Alternatively, with section selected by dragging the mouse choose trim from the edit menu (ctrl+T) and save the file choosing export_as_WAV from the file menu.
selecting a small section of a file
For smaller files, or for more accurate work select unwanted areas at either end by dragging mouse and delete using delete button, edit/delete or ctrl+K. You can use ctrl+1 or view/zoom in to view a larger version of the selected area.
In addition to the basic file trimming described digital audio files can be enhanced in many other ways. These could include any, or all of the following:
DC offset is low-frequency, inaudible noise that results from equipment grounding problems. If you don't remove it, it can skew the results of subsequent sound editing. Dc Offsett can be adjusted using Audacity 1.2.0 by selecting the whole file (ctrl+A) and choosing normalize form the effect menu, with remove DC offset and normalize to -3dB selected.
Normalization maximizes the volume level of the audio file's loudest sections. Consequently, quiet sections may not encode as well. Dynamics compression evens out input levels by attenuating (turning down) the input when it rises above a specified threshold. Audacity 1.2.0 includes a compression utility, which can be found on the effects menu. You can control attenuation by specifying a compression ratio. This turns down the loudest sections, and you can readjust input levels accordingly.
Tip: For multipurpose dynamics compression, set the threshold to -10dB, the ratio to 4:1, and the attack and release times to 100ms. Adjust the input level to get approximately 3dB of compression and an output level of about 0dB.
Equalization (EQ) changes the tone of the incoming signal by "boosting" (turning up) or "cutting" (turning down) certain frequencies. Using EQ, you can emphasize certain frequencies and cut others that contain noise or unwanted sound. The plugin VoxengoEssEq which can be found at the bottoom of the effects menu provides seven band eq control.
Tip: For voice-only content, you can make the file more intelligible by cutting frequencies below 100 Hz and carefully boosting frequencies in the 1 to 4 kHz range.
Your edited sound file should have the highest possible gain without clipping. Digital audio files that do not use their full amplitude range produce lower quality playback. If the amplitude range is too low, you can use Audacity 1.2.0 to adjust the range and increase the amplitude.
Tip: the Amplify function found on the effects menu maximizes levels automatically. Because some systems have trouble with files normalized to 100 percent, normalize to 95 percent of maximum with 'the 'new peak amplitude' level set to -0.5dB.
The editing functions described elsewhere on this CDrom will produce digital audio files of good quality suited to most archive functions. However, better quality still can be achieved with care and experience.
One the more advanced techniques that may be applied is that of Pre-Dynamic EQ. This relates to whether or not you are going to use some type of limiter (as per the Hard Limiter) or mild compression on your final mix. If you plan on doing so, you may consider tweaking EVER SO SLIGHTLY some key frequencies that may be accentuated in the compression process. Key frequencies are 60hz-80hz (bass, 45hz and below for sub-bottom) 250hz (muddiness/low mid) 1.25k-2.5k (vocal fundamental) 5.8k/6.3k (sibilance) and 8k-14k (presence and shimmer). By boosting or cutting (at very small increments) you can really tailor your sound like a pro, using a variety of narrow and broadband Q. Don't boost/cut more than 3dB in any one band, except perhaps in the 250hz range if your sound is very muddy. The VoxengoEssEq 7 band equalisation VST plugin, included with this installation, allows some control over Pre-Dynamic EQ. The more advanced user may wish to eperiment with other VST plugins or additional tools.
Original text © Mike Hirst 2004. This text © Mike Hirst 2006.