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Format |
Description |
|
ADPCM (MS, IMA) |
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 ratio.
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 format.
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.
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|
CCUIT A-LAW |
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.
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AIFC |
AIFF is Audio Interchange File Format, a
format for storing digital audio samples in a file. This standard format
for sound files was defined by Apple.
AIFC is short for AIFF(C) or AIFF-C, i.e.
the Audio Interchange File Format with optional compression. AIFC is a
newer version of the format that includes the ability to compress the
audio data.
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|
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.
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|
|
CCUIT G721 |
Used for computer telephony. 32 kbit/s
adaptive differential pulse code modulation (ADPCM).
Good for keeping of human speech.
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|
CCUIT G723 |
Used for computer telephony. Extensions of
Recommendation G.721 adaptive differential pulse code modulation to 24
and 40 kbit/s for digital circuit multiplication equipment application.
Good for keeping of human speech.
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|
CCUIT G726 |
Used for computer telephony. 40, 32, 24, 16
kbit/s adaptive differential pulse code modulation (ADPCM). Good for
keeping of human speech.
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|
MP2 (MPEG 1 Layer 2) |
MPEG Layer-2 format. Compression ratio is
1:6...1:8 corresponds to to 256..192 kbps for a stereo signal.
The extensions are *.mp2 or *.mpa.
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|
MP3 (MPEG 1/ 2/ 2.5 Layer
3) |
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;
- quantization;
- 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.
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|
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.
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|
CCIUT u-Law |
Compressed WAV format. u-Law (or CCIUTT
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.
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|
VOX |
Dialogic ADPCM format. The Dialogic ADPCM
format is commonly found in telephony applications, and has been
optimized for low sample rate voice. It will only save mono 16-bit
audio, and like other ADPCM formats, it compresses to 4-bits/sample (for
a 4:1 ratio). This format has no header, so any file format with the
extension .VOX will be assumed to be in this format.
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|
RAW |
Raw format of audio files. Doesn't contain
header of an audio file.
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|
Ogg Vorbis |
Ogg Vorbis format. Ogg Vorbis is an audio
compression format. It is roughly comparable to other formats used to
store and play digital music, such as MP3, VQF, AAC, and other digital
audio formats.
Ogg Vorbis is a fully open,
non-proprietary, patent-and-royalty-free, general-purpose compressed
audio format for mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic)
audio and music at fixed and variable bitrates from 16 to 128
kbps/channel.
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|
WAV |
It is not an audio codec. It is the file
format. This format was created by Microsoft and IBM, and it has
unfortunately become a popular standard. It specifies an arbitrary
sampling rate, number of channels and sample size. It also specifies a
number of application-specific blocks within the file. It has a plethora
of different compression formats.
It is the files with .wav extension. But
this files can be converted by different codecs. NCTAudioStudio2
supports the following types of WAV files:
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Microsoft PCM |
|
Microsoft ADPCM |
|
DSP |
|
GSM |
|
VOX |
|
A-law |
|
U-law |
|
CCUIT G723.1 |
|
CCUIT G721 |
|
CCUIT G723 |
|
CCUIT G726 |
|
CCUIT G729 (A) |
|
WMA |
Windows Media Audio format. A special type
of advanced streaming format file for use with audio content encoded
with the Windows Media Audio codec. The .wma extension indicates a file
format and how the content is encoded.
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