Convert AAC to WAV

A proprietary format with lossy compression, which have been created as an alternative for MP3 with a higher quality of coding. Overtones out of human perception are being removed. AAC is two times less than MP3, which means 1/10 of CD digital data.

The format has being widely used since 1997 as the seventh release of MPEG-2 family. As it is supplied by hardware players, the format is highly demanded by music fans. It has been widely spread in Apple production and is being used by default in iTunes.

The format conception is similar to MP3, but it is more advanced. Being able to execute larger compression, it is represented as a separate segment of MPEG-4 standard which is used for creation of compact audio files. AAC is highly efficient with variable and continuous bitrate, with sampling frequency of 8-96 KHz.

The current version complies with the standard of ISO/IEC 14496-3. Compression is performed on the basis of auditory masking effect that provides with data reduction with no deterioration for sound quality.

WAV is one of the most common audio formats and stands for Waveform Audio File Format. It is intended for operation with digitalized audio stream. Although it has also been used for non-audio data due to its sampling rate (1 Hz to 4.3 GHz). It contains musical compositions, voice recordings, and various audio effects. Developed by programmers from Microsoft and IBM, it is a key audio format for placing of uncompressed sound files on Windows PCs. The wav wave files can be played back in QuickTime, Windows Media Player, Wav Player and some other programs. In addition to that, they can be processed in audio editing apps.

When a file is compressed into WAV, the data are not supposed to be lost, and the quality is excellent. However, the format did not have a huge market share, due to its larger size, as compared with MP3. It is required to have enough time and disc space to upload and send such files via the Internet. One of the major advantages of WAV is linked to the use of Linear Pulse Code Modulation (LPCM) for storage of audio stream. Consequently, a copy is just as good as an original, which is highly appraised by experts in music and professional users.

Sound files with this extension are recorded into 8 or 16 bit per sample. A standard option for CD Audio is an audio stream of 16 bit per sample and sampling frequency of 44.1 KHz. One second of sound corresponds to 88 Kb of internal memory. WAV files can store metadata in the INFO chunk, and they also include integrated IFF lists.

In some cases, the standard format may be used for broadcasting. For instance, BBC stereo data of 44 100 Hz and 16 bit are generally accepted within the VCS system.