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Acez All Audio Converter (v3.0.5)

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:

  • 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
Microsoft ADPCM
DSP
GSM
VOX
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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