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Acez All Audio Converter
(v3.0.5)
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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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[ view screenshot ] |
Download Free Trial Version
of Acez All Audio Converter Main
features of Acez All Audio Converter:
- Convert audio files between 5 popular audio
formats:
MP3, MP2, MP1, MPEG -> MP3, WAV, OGG, WMA, VOX
WAV -> MP3, WAV, OGG, WMA, VOX
OGG -> MP3, WAV, OGG, WMA, VOX
WMA -> MP3, WAV, OGG, WMA, VOX
VOX -> MP3, WAV, OGG, WMA, VOX
Acez All Audio Converter can convert audio files from one type to the
same type but with different frequency, channel, bitrate, quality and so
on.
- Acez All Audio Converter supportes 4 wav
formats: Microsoft PCM, Microsoft ADPCM, DSP Group TrueSpeech, GSM.
Select the frequency, channel, bits per sample of the output wave file.
- Acez All Audio Converter uses MP3 Lame Encoder;
Support selecting frequency, channel, bitrate, encoder quality; HighPass
Filter, LowPass Filter; CBR, ABR, VBR encoding modes available. 16 kinds
of presets MP3 formats.
- Acez All Audio Converter supports selecting OGG
frequency, channel, nominal bitrate; VBR Enabled; Quality Enabled.
- Acez All Audio Converter has a list of
supported WMA formats.
- Acez All Audio Converter supports selecting
frequency of the output VOX files.
- Build-in audio converter: Play audio files of
MP3, MP2, MP1, MPEG, WAV, OGG, WMA, VOX formats.
- Auto rename, skip, or overwrite options for
existing file.
- Easy to use and user friendly.
Supported Audio formats of Acez All Audio
Converter:
| Format |
Description |
|
ADPCM (MS, IMA)
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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DSP
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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GSM
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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MP2 (MPEG 1 Layer
2) 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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MP3 (MPEG 1/ 2/ 2.5
Layer 3) 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;
- 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
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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VOX
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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Ogg Vorbis
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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WAV
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:
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Microsoft PCM |
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Microsoft ADPCM |
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DSP |
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GSM |
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VOX |
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WMA
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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