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2009-08-25 17:00:28
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Acez All Audio Converter is a powerful audio converter to convert audio files between MP3, MP2, MP1, MPEG, WAV, OGG, WMA and VOX formats. It includes all the features of Acez MP3 WAV Converter, and has the ability to convert to or from OGG, WMA, VOX audio files. Also Acez All Audio Converter has a build-in audio player to play MP3, MP2, MP1, MPEG, WAV, OGG, WMA, VOX audio files. Acez All Audio Converter is user friendly and easy to use. It works for all Windows platform. |
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Main features of Acez All Audio Converter:
Supported Audio formats of Acez All Audio Converter:
| Format | Description | |
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Supported by Acez All Audio Converter |
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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Supported by Acez All Audio Converter |
Compressed WAV format. DSP Group True
Speech (TM) format.
DSP Group's TrueSpeech is a family of high quality, low bit rate, speech compression algorithms which compress speech down to as little as 1/40th its original size. Several different versions of TrueSpeech at different compression rates are available for licensing, from 8.5 Kbps through 3.9 Kbps. All offer excellent communications over a 14.4 Kbps or better modem. TrueSpeech 8.5 is the 8.5 Kbps member of DSP Group's TrueSpeech family of software products. It is a low complexity speech coder, which is an integral component of Microsoft windows and has also been endorsed by Dialogic for computer telephony products. TrueSpeech 8.5 should be used when compatibility with Microsoft is a prerequisite. Speech information can now be exchanged compatibly between different applications. For example, using TrueSpeech 8.5 for digital simultaneous voice and data applications (DSVD), it may be feasible to utilize the same DSP chip for both speech compression and high speed modem data pump tasks. At the sampling rate of 8 KHz, continuous digital speech is compressed from 128 Kbps to 8.5 Kbps, a 15:1 compression ratio, while maintaining good speech quality. With slightly lower voice quality and lower levels of compression, TrueSpeech 8.5 requires only about half the MIPS and program memory space as TrueSpeech 6.3 and 5.3. [] |
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Supported by Acez All Audio Converter |
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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Supported by Acez All Audio Converter |
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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Supported by Acez All Audio Converter |
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; 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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Supported by Acez All Audio Converter |
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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Supported by Acez All Audio Converter |
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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Supported by Acez All Audio Converter |
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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Supported by Acez All Audio Converter |
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: [] |
Microsoft PCM |
| Microsoft ADPCM | ||
| DSP | ||
| GSM | ||
| VOX | ||
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Supported by Acez All Audio Converter |
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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