|Tutorials > AVI Settings
|Video compression methods
|The rate of the frames representation, in frames per second.
By Default value is: 23.97 frame rate
|PCM - Standard Windows WAV format for non-compressed
audio files. Pulse Code Modulation (PCM) is the standard method
of digitally encoding audio. It is the basic uncompressed data
format used in file types such as Windows .wav.
|ALAW - Compressed WAV format. A-Law (or CCITT
standard G.711) is an audio compression scheme common in telephony
applications. It is a slight variation of the u-Law compression
format, and is found in European systems. This encoding format
compresses original 16-bit audio down to 8 bits (for a 2:1 compression
ratio) with a dynamic range of about 13-bits. Thus, a-law encoded
waveforms have a higher s/n ratio than 8-bit PCM, but at the
price of a bit more distortion than the original 16-bit audio.
The quality is higher than you would get with 4-bit ADPCM formats.
Encoding and decoding is rather fast and generally, widely supported.
|MP3 - MPEG Layer-3 format. Very popular format
for keeping of music.
The mp3 algorithm development started in 1987, with a joint
cooperation of Fraunhofer iis-a and the university of erlangen.
it is standardized as iso-mpeg audio layer 3. it soon became
the de facto standard for lossy audio encoding, due to the high
compression rates (1/12 of the original size, still remaining
considerable quality), the high availability of decoders and
the low cpu requirements for playback. (486 dx2-66 is enough
for real-time decoding). it supports multichannel files (although
there's no implementation yet), sampling frequencies from 16khz
to 24khz (mpeg2 layer 3) and 32khz to 48khz (mpeg1 layer 3).
formal and informal listening tests have shown that mp3 at the
192-256 kbps range provide encoded results undistinguishable
from the original materials in most of the cases.
mp3 uses the following for compression:
|- huffman coding;
- m/s matrixing;
- intensity stereo;
- channel coupling;
- modified discrete cosine transform (mdct);
- polyphase filter bank.
|Compression ratio is 1:10...1:12 corresponds to 128..112 kbps
for a stereo signal.
|MPEG Version 2.5 was added lately to the
MPEG 2 standard. It is an extension used for very low bitrate
files, allowing the use of lower sampling frequencies. If your
decoder does not support this extension, it is recommended for
you to use 12 bits for synchronization instead of 11 bits.
|ULAW - Compressed WAV format. u-Law (or CCITT
standard G.711) is an audio compression scheme and international
standard in telephony applications. u-Law is very similar to
A-Law, a variation of u-Law found in European systems. This
encoding format compresses original 16-bit audio down to 8 bits
(for a 2:1 compression ratio) with a dynamic range of about
13-bits. Thus, u-Law encoded waveforms have a higher s/n ratio
than 8-bit PCM, but at the price of a bit more distortion than
the original 16-bit audio. The quality is higher than you would
get with 4-bit ADPCM formats. Encoding and decoding is rather
fast and generally, widely supported.
|ADPCM - Compressed WAV format. ADPCM (Adaptive
Differential Pulse Code Modulation) is an audio compression
scheme which compresses from 16-bit to 4-bit for a 4:1 compression
|ADPCM stands for Adaptive Differential Pulse
Code Modulation. ADPCM is a lossy compression mechanism. There
are various flavors of ADPCM. This particular algorithm was
suggested by Microsoft; its quality is similar to IMA (Interactive
Multimedia Association) ADPCM. MS ADPCM compresses data recorded
at various sampling rates. Sound is encoded as a succession
of 4-bit nibbles. Each nibble represents the difference between
the current sampled signal value and the previous value. The
compression ratio obtained is relatively modest: 16-bit data
samples encoded as 4-bit differences result in 4:1 compression
|Microsoft ADPCM is directly supported on
most Windows implementations as a native format. Although the
quality of IMA ADPCM voice files is not great, the files are
portable. There is a real advantage in having compact files
that can be played on most Windows PCs.
|GSM - Compressed WAV format. Good for keeping
of human speech. It is lossy speech compression that allow to
get telephone quality speech with 13 kbit/s. It is a standard
used for telephone sound compression in European countries and
its gaining popularity because of its quality.
|GSM 06.10 stands for Global System for Mobile
Communications and is a variant of LPC called RPE-LPC (Regular
Pulse Excited - Linear Predictive Coder) and is a European standard
originally for use in encoding speech for satellite distribution
to mobile phones. It can be found in use in various telephony
products such as voice mail applications.
It compresses 160 13-bit samples into 260 bits (or 33 bytes),
i.e. 1650 bytes/sec (at 8000 samples/sec). It results in very
good compression with good quality output but is very costly
in terms of performance.
|Number of sampling per second
|Mono - Mono data uses one channel.
|Stereo - Stereo data uses two channels