What audio formats are supported by
AVS Audio Tools?
Uncompressed WAV PCM
PCM, Pulse Code Modulation (developed in 1939), is a standard method for
digitizing analog audio signals. This format is an uncompressed, raw bit
stream, linear (transmitted in a linear series meaning that the stream
of the signal is sequential rather than random and the amplitude of both the of
the input signal and output signal remain at a fixed ratio and a sinusoidal
wave input signal will result in a sinusoidal wave output signal at the same
frequency), signed two's complement, fixed point encoded data file. A PCM
encoder (ADC) may sample analog sound from 8,000 times per second (8 KHz) and
use 8-bits to represent each sample, is usually utilized at 44.1 KHz and 16-bit
resolution to match CD Audio standards, and can encode at 96 KHz and 24-bit
(approximate). The procedure uses only two alternating pulse values (1 and 0)
duplicating binary code. This codec creates a raw (RAW) data file (no header or
footer information describing sample rate, sample resolution, monaural or
stereo) and gives us only 256 possible amplitude values (based on 8-bit binary
numbering). When the codec is set to sample at the Audio CD level at 44.1 KHZ
sample rate (44,100 samples per second), with 16-bit resolution per sample
(65,536 possible amplitude levels), this results in a file that requires 1,411
Kbps (kilobits per second, or 1.4MB) bit rate for representation / playback of
one second of stereo music.
Compressed WAV
ADPCM, Adaptive Differential Pulse Code Modulation developed by
Microsoft and IBM, produces a high-quality sound than found in the WAV format
and a compression ratio of 4 to 1. This is a conversion of a PCM bit format
that, similar to DPCM, attempts to predict the value of each successive sample
during the encoding process. It also utilizes a variable bit rate procedure to
reduce the difference between sample amplitude levels. Thus, it very efficient
and reduces file size by encoding the difference between successive samples
rather than expending all encoding bit resources on reproducing the sample.
There is both a Microsoft ADPCM and an IMA (Interactive Multimedia Association)
ADPCM (used in Sony MiniDisc in tandem with ATRAC). One can use IMA ADPCM in
the WAV, AIFF and SND formats.
U-law, A-law Compression is a common lossy compression scheme, similar
to ADPCM, which can be used on AU, AIFF and WAV files. It is an international
standard for compressing voice quality audio. It has a compression ration of
2:1. The G.721, G.723-24 and G.723-40 ADPCM formats are CCITT standards for
compression of 8000-Hz 14-bit samples into a 32-, 24- or 40-kbps data stream.
These compressed formats have extremely slow decompression rates. Because it is
optimized for speech, in the United States it is a standard compression
technique for telephone systems (in Europe, a-law is used). On the Internet it
is used for ".au" file formats, alternately know as "Sun audio" or "NeXT"
format.
GSM is the international standard digital mobile telephony encoding
format. It uses linear predictive coding to substantially compress the data by
predicting the likely shape of the sound wave and recording the differences
between the actual sound and the prediction. Compression and decompression are
slow and the quality is not great, but the algorithm is freely available
resulting in widespread use in products.
MPEG 2 Layer2 (MP2), MPEG 2 Layer3 (MP3)
MP3 was introduced as a part of the official MPEG-1 standard in 1992 and until
today it is the most successful audio-standard since WAV. The German Fraunhofer
Gesellschaft (FhG), which has developed this audio-compression still
holds the key patents the MP3-techology inherits. The development started back
in 1987 at the Fraunhofer Institut Integrierte Schaltungen as project
EUREKA EU147. The final compression algorithm became later known as MP3. In
April 1989 Fraunhofer applied patent on MP3 in Germany and it became part of
the MPEG-1 standard in 1992. It was in January 1995 when Fraunhofer applied
patent on MP3 in America as well and it was granted in November 1996. Using
MP3-compression PC-users were able to compress an ordinary music-CD to one
tenth of it's original size - thus 12 hours of music could be stored on a
recordable CD that on the other hand could be played by a MP3-CD-player or an
ordinary PC. What made MP3 that popular in the end was the online peer-to-peer
program named Napster. Millions of songs were exchanged every day via the
popular program. That was solely possible by MP3, because conventional formats
such as WAV or AU were way to big in size with similar quality. MP3 also
offered like WMA later the big advantage of being streamable (not all of the
file has to be downloaded to listen to it).
WMA
Microsoft's respond to MP3, the Windows Media Audio-standard. As it is part of
the Windows Media package, Windows Media Audio 8 was presented in early
December 2000 and it is until now the best Windows Media product.
Windows Media Audio among other things is firmly integrated in Microsoft's
Windows Media Player.
Microsoft promises with this version almost CD-quality with just a third of the
source-file's size.
Above all WMA offers the advantage that copyright-protected songs cannot be
published any further (Digital Rights Management). That's not the only reason
why many music- and movie-corporations meanwhile decided in favour of WMA
instead of MP3. Like MP3 WMA is almost predestined for the internet by offering
streaming capabilities (see MP3 for details) both with WMA and WMV (Windows
Media Video).
Ogg Vorbis
The development of the OGG standard began in 1993, then known as "Squish". OGG
was right from the start an open source project and hence is free of any
patents. It was designed as a substitute for MP3 and WMA and by now it is
almost as popular and well known as MP3. Above all, the algorithm is still
being developed what is mainly due to its flexibility. Although the
sound-quality gets better with every further development the files are
backwards compatible and can be played with older players as well. Like MP3 OGG
offers encoding at variable bitrates. Using this compression parts of the song
are encoded with a higher compression than others what depends on the source.
Most times, this compression goes along with squishy noises or even small
interruptions. OGG is also one of the very few formats that support
multi-channel compression. Surround-files could theoretically be compressed
with more than two channels. OGG is, like it's predecessors, streamable and
although the used player has to support this feature, it's one of many good
reasons for OGG.
VOX (Dialogic PCM)
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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