Most common audio formats are supported (some of them directly such as
Ogg Vorbis, WAV (Windows wave), FLAC (Free Lossless Audio Codec)
or AIFF (Macinthosh Audio Interchange File Format) ... and others such
as MP3 (by means of LAME) or AAC (by means of FFMPEG)).
Sample precission can be 16, 24 or 32 bits/sample. Therefore, it
cannot capture 8 bits/sample and therefore, use logarithmic quantization.
However, it can export these formats throught libsndfile.
Gain: A measure of how much a signal is amplified. 0 dB means not
amplification nor attenuation.
Pan: Controls the spread of a stereo sound into the stereo channel.
Sync-Locked Track Groups: lets you keep existing audio or labels
synchronized with each other even when carrying out actions like inserting,
deleting or changing speed or tempo.
Stereo track: a 2-channel track where each channel is a independent
audio signal.
Mono track: a 2-channel track where there is only one audio signal (the
same for each channel).
Left/Right track: a 1-channel track with data for, only, the left/right
channel.
Label track: Used to reference points or regions in the project’s audio
tracks.
Time track: It is used in conjunction with one or more audio tracks
to progressively increase or decrease playback speed (and pitch) over the
length of the audio project (only one Time traks per project is allowed).
Playback Region: a track region associated with a shaded selection
region in the waveform which is indicated in the timeline by a thin
horizontal gray bar with arrowheads on each end.
Overdubbing: is the process of record sound while the existing tracks
are played.
Speed: The number of samples/second played.
Tempo: It refeers to the playing time but not changing the pitch.
Pitch: The fundamental (first one) frequency of a sound (usually, a music
note).
For each tone you wish to generate, enter numbers from 0 to 9, lower case
letters from a to z, and the * and # characters. You can also enter the
four “priority” tones used by the US Military (upper case A, B, C and D).
Use the slider to select the ratio between the length of each tone in the
series and the length of the silences between them.
Produces a sound based on work by the composer of electronic music, Jean
Claude Risset.
Chapter 3 Recording and playing
3.1 Sound driver selection
By default, in Linux, Audacity will use ALSA which is the best choice for
avoiding latency issues during capturing audio.
3.2 Input (recording) and output (playing) selection
Depending on your audio hardware, a number of input devices are
available. Again, in Linux systems, ALSA native “hw” devices should
minimize latency problems.
The Transport Toolbar is the easiest way to control Audacity playback and
recording. From left to right we found:
The Pause Button: press once to pause playback or recording then
once to resume.
The Play Button: press once to start playback from the beggining.
The Stop Button: interrupts the playing/recording.
The Skip to Start Button: moves the time cursor to the start of the
project.
The Skip to End Button: moves the time cursor to the end of the
project.
The Record Button: start a new audio capture (you must push Stop
to stop this process).
3.4 Using keyboard shorcuts
P: Pause.
SPACE: Plays or stop.
SHIFT+SPACE: Plays in a loop.
HOME: Skip to Start.
END: Skip to End.
R: Record.
SHIFT+R or SHIFT+Record: Append-Record. Record starting from
the end of the selected (previously existing) track(s).
SHIFT+A: Stop and set cursor. When stop a playback or a recording,
the cursor or start of the selection is set to the position where
playback/recording was stopped.
Latency is an issue that depends mainly on your hardware (sound device
and computer) and your audio buffer size
(Edit -> Preferences -> Recording -> Audio to buffer).
In order to minimize the latency, you can improve your hardware and/or
reduce the audio buffer size. However, reducing the audio buffer size can
produce underruns in the capturing process if the CPU is not able to
manage this configuration requirement.
When overdubbing
(see Section 3.9), you must measure and set your current lantency in
Edit -> Preferences -> Recording -> Latency Correction,
otherwise those audio tracks that have been previously captured will be
the synchronized with the new ones.
To find out your latency you must capture a sound produced by your computer
and compute the delay between the time the sound has been produced and the
time the sound has been captured. To do this:
In Selection Toolbar make sure that "Snap To" is set to "Off".
Above the second group of numbers, make sure that “Length” is
selected.
Click on one of the downward-pointing arrows in the digits boxes to
the right of “Snap To” and select “hh:mm:ss + milliseconds”.
Generate 2 bars of click track (Generate -> Click Track), and
choose the default “Ping” sound.
Click “OK” to generate the click track.
Now click the Record Button in the Transport Toolbar.
You will get a new track. The top track is the original click track,
the bottom track is the looped-back recording.
Zoom in so you can see one of the clicks in the top track and its
delayed version on the bottom track.
You can now read the latency directly from the second panel of
numbers.
Write in the Edit -> Preferences -> Recording tab -> Latency Correction
Entry the negative of this number.
Repeat this experiment to check if the latency when overdubbing has
been “hidden”.
3.6 Timer recording
The Transport -> Timer Record Dialog allows to configure a
temporized recording (starting date and time and duration).
Can be used to compare the audio level of two different parts of a
mono track. Suppossing that one of the parts is the foreground (that
obviously also has a background noise) and the other is the backgroud
(with only background noise), this analysis can be useful to know if there
is enoung signal (20 dB or more) in the foreground section compared to
the background section.
Procedure:
Select a region containing the signal (speech, for example). This is
the "foreground" selection.
Click the “Foreground”’s “Measure selection” Button.
Select a region containing only the background sound. This is the
background selection.
Click the “Background”’s “Measure selection” Button.
Displays runs of clipped samples in a Label Track.
You can control the length of the clipped runs (default 3 samples) and
the number of unclipped samples (default also 3) that must occur before
a sun of clipped samples will be determined.
Graphs the spectral density function of the sound.
You can choose between:
Spectral analysis: A genuine Fourier analysis.
Autocorrelation: A measure of the redundancy of the signal.
Cepstrum: The power cepstrum
(of a signal) is the squared magnitude of the Fourier transform of the
logarithm of the squared magnitude of the Fourier transform of a signal.
Mathematically:
Physically, the cepstum resents the rate of change in the different
spectrum bands. It’s particularly useful for properties of vocal
tracks and is used, for example, to identify speakers by their voice
characteristics.
When you reproduce (push the Play Button) a project, all tracks are mixed
together in order to play them, but you don’t have a track that can be
written into a stereo audio file with the result of our project.
The Mix and Render Option of the Tracks Menu explicitly mixes down
all selected tracks to a single mono or stereo track. The resulting track
(called “Mix”) replaces the selected tracks and is placed underneath any
tracks that were not mixed and rendered.
It is possible also to mix and render to a new track by using
CTRL+SHIFT+M.
A simple, combined compressor and limiter effect for reducing the dynamic
range of audio. It reduces the difference between loud and soft, making
the audio easier to hear in noisy environments or on small loudspeakers.
Reduce constant background sounds such as “hum” (zumbido), “whistle”
(silvido), “whine” (gemido) or “buzz” (rumor), and moderate amounts of
“hiss” (siseo).
Removes a very short region (up to 128 samples) of damaged or destroyed
audio, replacing it with an estimated region of audio based on what is
happening either side of the region.
Flips the audio samples upside-down, reversing their polarity. Notice that
if you play the original and the inverted audio together, both tracks will
be substracted.