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The quality of an image depends on several
factors:
• Number of points (pixels) that compose it;
the larger the number of points used in the
image, the better quality it has, since each
point will be smaller and the image will be
shown with more detail.
• Colour depth: refers to the number of
colours that has the image, and the larger it
is, the more shades it has.
• Compression of information: when saving
a bitmap image on a file, the information of
each of their points or pixels is stored,
consequently the file size is very big. To
avoid it, the most of the graphical formats
use some compression method, which
implies a loss of quality to a greater or lesser
extent.
A sound wave is expressed as a sine function with the following parameters:
• Amplitude: indicates the intensity of sound and is measured in decibels (dB). Noise
levels above 85 dB can be dangerous its continuous exposure causes hearing
damage.
• Wavelength: is the separation between the crests of the wave and indicates the true
distance that the sound travels over a time period.
• Frequency: is the number of oscillations per second. The more frequency, the lower
wavelength. High frequencies are translated into high-pitched sounds and low
frequencies into bass sounds. Frequency is measured in Hertz (Hz).
• Spectrum: represents the intensity of sound for each one of the frequencies
The digitalisation of sound consists of the conversion of the analogic wave into digital
values that a computer system can interpret. To make this process, a microphone that
converts wave sounds into electrical signals is used. Then, the sound card performs the
analogue/digital conversion. In order to listen the digital sound the inverse process
has to be made, that is to say, the digital/analogue conversion. This conversion is
made by the sound card, amplifying the sound to the levels needed for vibrating the
speaker cone that will again generate the sound waves which can perceive the human
ear. The digitalisation process takes place in three phases:
Sampling consists of taking samples periodically from the original sound at regular
intervals. The number of samples that are taken in about a second is called sampling
frequency and is expressed in Hertz (Hz).
Phase 2; quantification of the analogue signal: Quantification consists of assigning to each sample a numeric value. The smaller the
interval between the possible values, the more approximate the measurement will be. If
the value of the sample remains
phase 3: binary coding
Uncompressed formats
MIDI (Music Instrument Digital Interface). Communication protocol between the
computer and other electronic music devices , being the most used in musical composition. CD-A or CDDA (Compact Disc Digital Audio). Uses as support the optical disc where the recording is carried out through a laser
beam. Is a high quality non-compressed format WAV (Waveform Audio File). Microsoft´s format that is normally used in Windows
for storing sound in different resolutions and sampling rates. Is not very popular on
the Internet because the files are very large.
Lossless compression formats
FLAC (Free Lossless Audio Codec). Is an free-licence open format. Allows
compressing an audio file up to half of the original size and decompressing it into a
file identical to the original one.
AIFF (Audio Interchange File Format). Developed by Apple, can be distinguished by
its fast signal processing,
Lossy compression formats
MP3 . One of the most popular formats due to its
sound quality and small size. .
Ogg Vorbis. Free open format, based on the public GNU licence. Can
multiplex independent channels for audio, video, text and metadata. WMA (Windows Media Audio). Audio compression technology,.
Real Audio. Widely used around Internet because of its real time playback
without generating an audio file, for instance, on an online radio station. This
format is often associated with the RealPlayer audio player.
AAC (Advanced Audio Coding).