Question #1 What is Audocity’s default bit recording quality? 32 bit
- Supports 16-bit, 24-bit and 32-bit (floating point) samples (the latter preserves samples in excess of full scale).
- Sample rates and formats are converted using high-quality resampling and dithering.
- Tracks with different sample rates or formats are converted automatically in real time.
Recording in 32 bit quality takes a lot more work than recording 16 bits, and a slower computer may not be able to keep up, with the result being lost samples. If you are recording for immediate export without editing, 32 bit recording may offer no advantage over 16 bit recording if you only have a 16 bit sound device. Most built-in consumer sound devices on computers are only 16 bit (including cheap sound cards)
You can change the default sample format Audacity records at to 24 bit or 16 bit by going to the
Record at 24-bit depth on Windows (using Windows WASAPI host), Mac OS X or Linux (using ALSA or JACK host).
- Record at very low latencies on supported devices on Linux by using Audacity with JACK
- Record at sample rates up to 192,000 Hz (subject to appropriate hardware and host selection). Up to 384,000 Hz is supported for appropriate high-resolution devices on Mac OS X and Linux
Question #2 How can I split a long recording into multiple files or CD tracks?
silences between tracks are automatically detected and labeled using Analyze>Silence Finder (Audacity legacy 1.2.6 Plug-in Pack)
- Click to place the cursor at the start of the first song
- Choose “Add Label at Selection” from the Project menu (or Tracks menu in Audacity Beta). If you wish, you can type the name of the song
- Repeat steps 1 and 2 for each song
- When you are finished, choose “Export Multiple” from the File menu. When you click the “Export” button, Audacity will save each song as a separate file, using the format and location you choose
Question #3 How do you capture a quality voice recording ?
Audacity can record live audio through a microphone or mixer, or digitize recordings from cassette tapes, records or minidiscs
- Record from microphone, line input, USB/Firewire devices and others
- Timer Record and Sound Activated Recording features
- Dub over existing tracks to create multi-track recordings
Dealing with Technical Issues
One feature of recording people speaking is uncertainty of recording level. Speakers vary in volume, and may not be aware of the best microphone techniques so for example may stand in different positions relative to the microphone. In some cases, such as meetings and conference recording, there may also be remote participants who are being heard through a radio or television receiver. The result is wide recording level variation
Rather than record at the final bit depth wanted (let’s say 8 bits), with digital recording one can record at greater bit depth and set the recording level relatively low (say 10 dB to 20 dB below the -0 dB distortion level). This retains plenty of dynamic range but avoids the risk of speakers who are louder than others creating clipping, which will result in unpleasant sound quality
There are also pros and cons about recording at different sample rates. The sample rate of the recording determines the highest frequencies that can be captured. Generally, lower sample rates are acceptable in speech recording where they are not in music, because voices (especially male) have a lower upper range of fundamental frequencies than instruments. Also, by the nature of the different sounds made when speaking and singing, it’s less important for quality reasons to capture the higher overtone frequencies in speech. In any case, the higher the sample rate you do record at, the more CPU time and disc space will be used
Multi Channel Recording
Where speakers don’t stand close to the microphone, multi-channel recording helps to keep all speakers above the room noise level, and clearly audible
For meetings it may be useful to place several microphones around the room, recording each microphone on its own channel. The multiple channels can be mixed down to mono later, selecting for each speaker whichever channel gives the highest ratio of speech to room noise. When post-processing, simply choose one channel for each speaker, mute the others, then mix down
Once again, more channels mean greatly increased CPU use and greater use of disc space. It’s important to test the hardware in multi-channel mode in advance, as running out of CPU capacity could cause recordings to have drop-outs or fail completely
Where simplicity is required, using only two microphones in different positions can still significantly improve the end result
The generally more stable nature of Linux or Unix operating systems may mean a reduced chance of a recording failure if you record with these systems rather than with Windows (other things being equal). Up to date sound card drivers specific to your hardware are more reliable than generic drivers when recording. Be prepared so you can quickly reinstall sound drivers between events if necessary
Shutting down unnecessary applications and processes so that the recording has most of the available CPU to itself is important – especially on slower and older machines with less RAM. Don’t make text notes on the PC that is recording, or do anything but record with it. Such actions are likely to cause recording skips. Consider making a checklist for any important recording. You may want to do a last minute check that you’ve got power settings set to always on, screensaver off, levels set right and so on before you record. Backing up audio files as soon as possible using cloud based storage reduces the chance of data loss
Never use a built-in microphone that comes with a laptop, MP3 recorder or tape deck. Such microphones pick up lots of noise from the device’s drive or from the deck motors or tape
While microphones are usually set on stands for formal events, for meetings of a handful of people held round a table, microphones on the table may be sufficient. Always place the microphone on something soft and squashy so that sounds and vibrations transmitted through the table – of which there are usually many – are not picked up directly. The microphone lead itself should sit on the squashy item before it reaches the table, as some sound and vibration can be passed up a short length of cable. The squashy items should be stable however; sponges fail in this respect! Folded clothing works fine, and the informal appearance helps put speakers a bit more at ease. Alternatively, microphones hung overhead avoid vibration and disturbance
Generally the less attention speakers pay to microphones the better their talk, and a way to minimize awareness is to not even mention the subject. Hidden microphones can put speakers more at ease. They still know the microphone is there, but not being repeatedly reminded of it helps. A basic way to create a hidden microphone is to cover a black-coloured microphone and its visible section of lead with a layer of lightweight open weave clothing
Question #4 How do i improve recording quality?
The built-in sound card that comes with many computers is quite poor. It may be satisfactory for playing sound effects, but not good enough for high-quality recording. Worst offenders are the built-in sound on your motherboard, or any audio device on a laptop. Some tips to reduce noise on your current system:
- Mute playback of devices that you don’t use for recording – such as MIDI Synth, CD Audio, TAD-In, Auxiliary, Microphone, Line In. Only “un-mute” devices to be used.
- Update sound drivers
- Consider shielding your soundcard
- If possible, insert your soundcard into a PCI slot which has a dedicated “Interrupt Request (IRQ) Channel”, as described in your motherboard handbook. Except for dual processor motherboards, there will probably be 4 electronic IRQ channels used to assign IRQs. (This is not the same thing as the 16 virtual IRQs we usually talk about.) For example, on my ASUS CUV4X mobo, Interrupt Request channel “A” is shared by AGP (reportedly noisier than PCI video cards) & PCI-slot1 (leave blank if AGP is in use) & PCI-slot5 (empty). Int-“B” is shared by AGP & PCI-slot2 (NIC – noisy). Int-“C” is a dedicated electronic channel, taking hardwire interrupt pulses generated solely by the device installed in PCI-slot3 (soundcard). Int-“D” is shared by PCI-slot4 (SCSI – noisy) and USB-controller (mouse, keyboard, etc. – very noisy). If I install my soundcard in any slot other than PCI-slot3, the result is a scratching sound, like a loose connection at an input jack. But, it comes from the mouse pulses (slot controlled by Int-“D”) or from video rewrites (slot controlled by either int-“A” or “B”)
- Even if you are using a ‘silent PC’ (one with passive cooling rather than a fan) you will still need sound insulation between it and your mic (a piece of felted board will do) as the hard drives are not silent
- If you are using any outboard (externally powered) audio hardware, make sure all the equipment is plugged into the same power strip. Grounding issues can cause ground loops, which will appear in your recording as a hum
- Cheap sound hardware, anywhere in the analog chain, will result in poor quality recorded sound
Buy a new sound card – especially if you were using your computer’s built-in audio capabilities before. The sound card’s ADC or analog > digital converter is the final step in your analog audio chain. Consider buying a USB audio external interface. The main advantage is that the A/D converter is then outside your computer’s case, which keeps electrical noise to a minimum – only the digital signal gets transmitted back to your computer. Another advantage is that you don’t have to open up your computer to install anything, just plug it in and go (after installing any software drivers required by the device). It’s easier to share it between multiple computers, too. Make sure other USB devices are unplugged if not being used, as especially USB 1.x has a limited bandwidth. Even things like network cards can interfere with USB audio so disable them
Microphones, more than any other single piece of hardware, will impact the quality of your recorded sound
- If you are doing studio work, a condenser microphone (rather than a dynamic) will probably be the most suitable. They have greater accuracy and dynamic and transient response compared to dynamic microphones. For live recordings, professional dynamic microphones may be preferable – they will be less prone to picking up extraneous stage and audience noise
- If you use a professional microphone, you’ll need a good preamp. The “mic” input on a sound card has a preamp behind it, but it’s usually not very good quality and will usually not provide sufficient amplification for the low outputs of pro microphones
- Note also that built-in computer 1/4 inch mic ports are almost always mono and unbalanced. Built-in computer line-in ports are almost always unbalanced. Unbalanced inputs mean you must keep the cable short to prevent interference and muffling, but this increases the interference risk from being too close to the computer. For this reason, many external USB and firewire recording interfaces will provide balanced inputs and outputs
- Don’t forget accessories like microphone stands and cables. Use XLR cables
Question #5 How do you set recording levels?
The level at which you record your audio is very important. If the level is set too low, your audio will have background noise when you turn the volume up to hear it properly. If the level is too high, you will hear distortion. The process of testing the recording level without actually recording is called monitoring . To do this in Audacity, you need to use the Meter Toolbar:
or in legacy 1.2.x go to the and check
In the image above, the left-hand VU Meter with the green bars measures the playback level, and the right-hand meter with the red bars measures the recording level. Assuming you are , the upper bar stating “L” refers to the left-hand channel, and the lower bar “R” refers to the right-hand channel. The values on the meter are negative values below the distortion level, where the distortion level has a value of zero (0). Hence the smaller the negative values become, and the closer the meter reads to the right-hand edge of the scale, the closer you are to the maximum possible level without distortion
To start monitoring, look at the right-hand recording meter just to the right of the recording symbol, and click the downward-pointing arrow:
This reveals a dropdown menu:
The menu has several options. “Vertical Stereo” will rotate the meter so that the zero level is at the top, and “Linear” will change the meter scale so that the values read from zero to 1.0 where the distortion level has a value of 1.0
Now you can start singing into your microphone, playing your guitar (or your record or tape), and you will see the red recording bands move in real time with the loudness of the input signal. If you can’t hear what you are recording, this doesn’t matter for the purposes of the level test because the meters will indicate the level accurately
For most purposes, an optimal recording level is such that when your input is at its loudest, the maximum peak on the meters is around –6.0 dB (or 0.5 if you have your meters set to linear rather than dB). This will give you a good level of signal compared to the inherent noise in any recording, but without creating distortion. Distortion is often referred to as clipping , because at this point there are not enough bits available to represent the sound digitally, so they are cut off above this point
To adjust the input level itself, use the right-hand slider (by the microphone symbol) on the Mixer Toolbar:
Move the slider left or right until the recording level settles at about -6 dB. If the meter bars drift so far to right of -6 dB that they touch 0, the red indicator will appear to right of the meter bar, as in this image where the left hand channel has at some stage peaked beyond the distortion level:
As soon as you see the red indicator you’ll know you have increased the input level too far, so will need to move the input volume slider back leftwards. Note that the achieved recording level is a combination of both the input level you record at and the output level of the source. If you find you achieve near maximum levels on the recording meter with only a very low setting on the input slider, this may lead to the recording sounding excessively close and un-natural. In this case you may want to cut back the output level somewhat if you can. Similarly if you can’t get close to maximum levels on the meter even when the input slider is on maximum, try turning up the output level
If you find the meters don’t respond at all to the input slider, double-check that you are recording from the correct input source as selected in the dropdown box to right of the input slider. If you still have problems, try setting the levels in the system mixer instead. On Windows machines this is done through the Control Panel, and on Mac OS X systems in Apple Audio-MIDI Setup
There is no reason you can’t use a standalone recording meter if you prefer
Monitoring the audio using playthrough
The simplest way if you wish to hear what the monitored input sounds like is to go to the, and enable . Don’t enable this option if you are recording sounds the computer is playing with the sound device’s “Stereo Mix” or similar option, because this will lead to echoes or even failure of the recording
Question #6 How do you reduce noise?
Noise can be reduced during post-production, by use of various plugins. Typically, they are fed a sample of the noise alone and then subtract that noise from the rest of the recording. To facilitate this, be sure to record a second or two of “silence” before you start the actual performance. This gives you a clean sample of the noise. This works extremely well with low-level background sound like air conditioning
PREVENTION IS BETTER THAN CURE
- Avoid noise in the first place instead of trying to remove it afterwards
- Use a sound card with balanced inputs, and use shielded cable, sufficiently long so that you can move the microphone right away from the computer
- An inexpensive external USB sound card should (other things being equal) be much quieter than the motherboard sound device that came with the computer
- Place the microphone on a floor stand and ground it separately from the computer, or use a ceiling-suspended microphone
- Keep all microphone cables away from mains electricity cables
To prevent noise entering your microphone recording:
- Set the correct input level of your sound sources. Set it as high as possible to increase the dynamic distance of sound and noise, but as low as necessary to prevent clipping
- Use balanced audio connections and shielded cable
- If you have it at hand use a hardware limiter as long as Audacity cannot process sound in real-time. Personally, when recording a band I use a simple foot pedal limiter on the vocals, because these seem to be the most dynamic sound source. The input level of instruments can be adjusted quite easily
- When you have found the optimal levels, decrease them about a dB or two when recording music just to be sure. When actually performing, musicians tend to be a little bit louder than in rehearsal mode
- Shut every non-used sound channel and sound source. Mute non-used channels on your mixing board, switch off non-used amps, keyboards, and don’t forget shut the door and the window
- Especially in a home-recording environment, avoid switching lights or electric machines on or off during recording, because a spark can cause a knack on the track
- Avoid fluorescent lighting and keep cell-phones a good distance away from any equipment
Noise tends to stay on the same low level and cannot be controlled in general, but the record source is often dynamic and may change and can be controlled. So you may get aware of your potentials in noise control here. Initially non-audible noise often comes to attention when a low signal is amplified and/or normalized because normalizing and amplifying increases both the wanted signal and the unwanted noise. Therefore, the measures and proceedings described prevent some of the typical noise problems later
- Cheap PC mics, besides having rotten sound quality, are nearly omnidirectional. Use a directional microphone. If the business end is not pointing at the noise source, it won’t pick up the noise. It may still pick up ambient noise, including any sound originating elsewhere in the room and bouncing off the walls. That will be reduced if you put the microphone right on top of the sound source. When recording vocals, the performer’s lips should almost be touching the mic, and sing straight into it (not across the top). Ambient noise will be blocked by the singer’s head
- Use a noise-blocking stand and a long enough cable to distance the mic from the computer. Often times the vibrations from a computer’s fans and drives will vibrate the computer desk and the surrounding area. If the microphone and its stand are on the computer’s desk, the microphone often will pick up the vibrations and produce a noise on the audio track (often referred to as a “warble” sound: a soft, repeating hum). To help prevent this, use a ceiling-suspended microphone stand or a full-size floor stand that can have its height adjusted. If these (pricey) options are not available, an alternative is to support a desk stand using a sound-insulating lift, such as a flimsy cardboard shoe box or a number of newspapers. These things insulate the noise rather well, making it difficult for any vibration noises to flow through to the microphone. Almost any lift made of non-rigid, flexible material will do
- Direct connection. If recording instruments like keyboards and electric guitars, feed their signal directly into your sound card’s line input, or to a sound board and then into your PC. Guitars will need preamps. If you’re recording acoustic instruments, use directional microphones placed close to the instrument, or use a pickup with preamp and connect direct
- Get the desired signal as loud as possible (without clipping) into the microphone. This allows you to reduce the gain, which will also reduce the low-level noise. The further a microphone is away from the source, the more you have to amplify the mic’s input signal to get to a usable level. But, boosting the gain amplifies everything, including background sounds and even the internal electrical noise of the amplifier. Ideally, the microphone should be right on top of the source, with the gain no higher than necessary to get peaks around -3dB. If you are doing multitrack recording, record each individual track as loudly as possible. Set the final volume of each track during post-production mixdown
Note: placing the microphone “right on top of the sound source” might not be ideal when recording certain sources (such as bowed instruments like violins and cellos). Instead of placing the mic right on top of your sound source without regard to factors other than noise, you should experiment with different kinds of microphone placement until you find one that provides the best sound. If the “optimal” placement is too noisy you can look for other ways to reduce the noise. In the end, nothing beats an ideal recording environment
- Don’t forget the possibilities of non-technical noise reduction:
- Turn off your refrigerator and furnace / air conditioner during the recording session
- Watch out for telephones, cell phones, pagers, ticking clocks, lawnmowers, and the like
- Avoid locating the recording session near airports, train tracks, and fire stations
- Hang blankets on the walls, to dampen a live room. Or record in a room with wall-to-wall carpeting
- If you can, record in a basement, to help isolate your session from outside noise. You’ll probably need to use the blanket-on-the-wall trick here, since concrete walls make good sound reflectors
- Record late at night to reduce traffic noise leaking in from outside
60/50 Hz hum and/or crackle
- A common problem. Make sure all your recording equipment is connected to the same ground. This is easiest to accomplish by plugging everything into the same power strip. Then ground the computer separately from the recording equipment
- Keep microphone cables well away from mains electricity cables (including those behind walls)
- Try to use incandescent light bulbs (including halogen lamps); avoid using fluorescent lamps near a signal path (cables and equipment), especially for low-power signal lines such as microphone cables. Fluorescent lamps often generate a significant amount of high-frequency RF noise, which may then be captured by the cable or the equipment. Lamps on the ceiling do not usually induce buzzes (because they are far away), but if used in a group of 4 or more, they may introduce buzzes into the power line, which may affect other equipment on the same power circuit. Power conditioners may be used to alleviate this problem
- If all else fails, get rid of the hum during post-production by using a de-noise plugin or an extremely narrow notch filter. Crackle will be much more difficult to remove
On virtually any recording you can find noise. It is not always necessary to get rid of it completely. First-of-all, it is often audible only in very low-volume passages. Second, the average not-too-picky listener will accommodate to the noise level of your recording. In this regard it is comparable to the odor of a room: When you enter it you become aware of it, but once you stay in there for a period of time you’ll probably cannot smell it anymore. Third, he/she may listen to your program in the car or while washing dishes so he/she may be not in the position at all to hear the noise
Sometimes, a completely silent passage, e.g. between sequential parts of a program, can irritate a listener more than a constant low-level noise throughout the mix. This is so because complete silence may disrupt the ambiance of the material. There are situations where you actually want to add noise (e.g. in film production between a cuts of the same scene)
So, you may want to change your attitude towards noise here: It’s not just dirt that needs to be removed, but it’s a natural part of all listening experiences that has to be dealt with appropriately. In general, we need to accomplish two things: The noise just has to stay on a roughly similar level throughout the material and it should not be too obvious to let it be ignored
Removing hi-band noise appears to be an option you can choose to take or to leave out. If you apply it, do it on the most basic level you can reach, on unprocessed tracks (for example, before adding reverb), on single tracks or even on single passages
On multiple tracks:
- Before doing anything else, increase the tracks to a working level with the Audacity Normalize function. Leave a little bit of headroom when amplifying
- Mute all passages where there is nothing to be used in the mix. You may use the envelope tool to make silent passages of a track really silent
- Fade-ins and fade-outs are much better than sharp edges cut-and-pasted that very likely will cause clicks
On a single noisy track, you may want to use the Noise Removal feature of Audacity. You accomplish this task in two steps:
1) You pick a “noise-only” part of the track’s signal. This part should not be longer than approx. half a second. This is a sample of the noise that will be used to compute the necessary changes to the track to remove just the noise (though this idea is always merely theoretical)
You have to be careful in selecting your sample. If you pick a sample that contains not only noise but also a slight part of – let’s say – a reverb tail, you’ll remove that, too. To give you another example, the sound of breathing can be quite similar to noise but it provides a lot of the vitality of a vocal track. Now, select a small portion of noise, call “Noise Removal” and Select “Get Noise Profile”. If you are unhappy with your selection, you can repeat this step. Every time the last sample will be overwritten with the new one
2) Now, select the portion of the track that needs noise reduction (in most cases this will be the complete track). Call the “Noise Removal” function a 2nd time. You can click on “Preview” to listen to the first seconds of your selection or on “Remove Noise” to execute the noise filtering. The less/more slider is quite obvious, I think, and can be changed for testing and applying the effect
If you’re unhappy with the result you can Undo all changes on your track and try again. Noise Removal often helps to reduce hi-band noise such as hissing and (to a certain extent) crackling
While removing hi-band noise might be considered optional, removing subsonic noise seems mandatory to the writer. In contrast to hi-band noise you can apply subsonic reduction as a step in the mastering process, on the mixed and processed material, before a final Normalize
Subsonic or low-band noise can enter your recorded material in many ways, such as through physical vibrations during the recording or noise from the tape machine (if you still use one of those)
Everything below 20 Hz is called “subsonic” because the human ear is unable to perceive it as recognizable sound. You can recognize subsonic noise by eye when the zoomed in waveform in Audacity is not symmetrical along the time axis. If you have already applied normalization in Audacity as recommended above, this should have removed any DC offset in the recording. The reasons for removing subsonics are the same as with DC offset – they will reduce the headroom available on the recording by taking up dynamic range, and can introduce clicks when editing
To remove subsonics from your track you may filter it with:
- Audacity’s built in Equalizer under the Effect menu
- Audacity’s built-in High Pass Filter under the same menu – set the cutoff frequency to around 25 Hz. You can repeat this same effect a couple of times if a sharper cutoff slope is desired
After removing subsonic noise you can generally re-normalize your track, and it will appear louder yet much more defined in the bass
Question #7 How do you repair “popping” vocals?
Manually fixing ‘breath sounds’ on a vocal recording can take an inordinate amount of time – so if possible avoid them in the first place. A ‘pop-shield’ between speaker and microphone can help, and can be made cheaply. Should you still have popping or percussive vocals, here’s how to repair them, though it will never be as perfect as a good original recording:
1) Make sure the recording’s DC offset is zeroed. (This in itself will eliminate one possible cause of clicks generated by subsequent edits and silences and should be done before you do any editing). Do this by selecting the whole track and choosing theeffect. In the resulting box make sure you’ve only selected “Remove DC offset”
2) Zoom in on the percussive sound. They’re easy to pick up. They look like a big single waveform just before the rest of the sound
3) Select this waveform and then choose theeffect. This will soften the percussive and hopefully solve your problem
4) Since these percussive sounds are mostly very low in frequency, some users have reported great success using the ‘high pass’ effect instead of the ‘fade in’ effect as suggested in step 3), above. Note that the ‘high pass’ effect can be repeated multiple times on the same selection. This approach has an additional advantage of not interfering with or reducing the level of higher frequency sounds, an advantage when the vocal percussive sound was recorded along with other instruments or sounds
Question #8 What file types may be imported/exported?
- Audacity does not care what the extension of the file is. If the file is well-formed, libsndfile will correctly detect the format and import the file appropriately. Audacity fully supports 24-bit and 32-bit samples and almost unlimited large sample rates.
- Import and export WAV, AIFF, AU, FLAC and Ogg Vorbis files
- Fast “On-Demand” import of WAV or AIFF files (letting you start work with the files almost immediately) if read directly from source
- Import and export all formats supported by libsndfile such as GSM 6.10, 32-bit and 64-bit float WAV and U/A-Law
- Import MPEG audio (including MP2 and MP3 files) using libmad
- Import raw (headerless) audio files using the “Import Raw” command
- Create WAV or AIFF files suitable for burning to audio CD
- Export MP3 files with the optional LAME encoder library
- Import and export AC3, M4A/M4R (AAC) and WMA with the optional FFmpeg library (this also supports import of audio from video files).
- Audacity also supports virtually any uncompressed format using the Import Raw function. With this function you can also import SoundDesigner-II Files (used in the Mac-World).
- You can import multiple files at once by shift-clicking or control-clicking on multiple files in the Open or Import dialog boxes. Alternatively, drag multiple files to your Audacity window. (On Windows, drag the files to the Audacity project window, and on Mac OS X, drag the files to Audacity’s icon in the Finder or in the Dock)
- Hint: Audacity may not realize the file is an MPEG file unless it has an appropriate extension. To be sure, try renaming it so that it ends in “.mp3”, and then if libmad can open it, it definitely will.
- Audacity imports ID3 tags from MP3 files, which give the Artist, Title, Album, and other song info, using libid3tag. You can see these tags by selecting “Edit ID3 Tags…” from the Project menu. Audacity will let you save these tags if you export an MP3 file. You can write either ID3V1 tags or ID3V2.3 tags.
- Audacity projects consist of a project file (.aup) and a corresponding data directory. Audacity project files are just XML, so you can read them using any text editor or XML reader. The file format is intended to be specific to Audacity, but open so that others can work with it if they’re interested. Audacity projects store everything that you see in the Audacity window. They open and save very quickly, so you can continue your work where you left off.
Audacity project files are not intended to be used as a portable format, or as the primary way to store your audio. Export as a common supported format like WAV, AIFF, or MP3
- When you create a new project, Audacity writes data to a temporary directory. You can set the location of this directory (folder) in the Preferences dialog.
- To save time, Audacity doesn’t make a copy of files when you import them – instead, it saves a reference to the original file in the project. If you prefer Audacity projects to be self contained, you can choose to always make copies in the Preferences dialog on the File Formats tab.
Question #9 What type of editing tools are available?
- Unlimited Undo lets you revert actions all the way back to when you first opened the project.
- Undo History window lets you see all of the changes you’ve made, and quickly jump back to a previous point.
- Audacity splits tracks into small blocks internally, so large cut and paste operations are quick because they don’t require rewriting the entire track each time a change is made. This is different than the Edit Decision List system used by many other editors, but the effect is similar: editing is quick, and it’s easy to Undo.
- Audacity displays the current cursor position or selection bounds in a status bar in the bottom of the project window. You can change the units using the “Set Selection Format” option in the View menu.
- Lots of basic editing operations:Save/restore selection
- Trim (delete everything except selection)
- Find Zero Crossings
- Modify cursor and selection using arrow keys (modify with Shift and Control)
- Envelope editor lets you adjust the relative volume of tracks over time. Just select the envelope tool (the one with the two white diamonds pointing towards a center control point) and click on a track
- Drawing tool has three options to edit individual samples (zoom in first): Multi-mode tool lets you select, modify envelopes, edit individual samples, and zoom, all from one tool. Which tool is active is based on the exact location of the mouse:
- Click: change samples
- Alt-click: Smooth
- Ctrl-click (and drag): change just one sample
Mixing, panning, and warping
- Audacity automatically mixes when you have more than one track open. It automatically resamples as necessary.
- Each track is designated as either Left, Right, or Mono. When you see a stereo track (two tracks joined together), the top one is the Left Channel, and the bottom one is the Right Channel. To change this, use the track pop-down menu.
- Each track has a gain control that you can use to adjust its volume.
- Each track also has a panning control that lets you give it relatively more volume in the left or the right channel.
- Adding a Time Track lets you warp the speed of playback over time.
Question #10….What built-in special effects are available?
- Audacity has many built-in effects and also supports plug-in effects in the LADSPA, VST, and Nyquist formats.
- Change the pitch without altering the tempo (or vice-versa)
- Remove static, hiss, hum or other constant background noises.
- Alter frequencies with Equalization, Bass and Treble, High/Low Pass and Notch Filter effects.
- Adjust volume with Compressor, Amplify, Normalize, Fade In/Fade Out and Adjustable Fade effects.
- Remove Vocals from suitable stereo tracks.
- Create voice-overs for podcasts or DJ sets using Auto Duck effect.
- Other built-in effects include: Run “Chains” of effects on a project or multiple files in Batch Processing mode.
- Paulstretch (extreme stretch)
- Truncate Silence