First released in 1995 and developed by the Fraunhofer Society, the mp3 audio format is arguably one of the most recognizable audio file format that ever existed. This digital audio encoding format hails from the lossy data compression algorithm family. The format enjoys widespread support and compatibility with most handheld music players, smartphones, computers, and in car stereos most of which have native support for the format. It is also a popular format used for distribution of audio files over computer networks such as the internet. Free mp3 encoders such as LAME and audacity give authors the tools necessary to create .mp3 files.
The Waveform Audio file format associated with the .wav filename extension was a format developed through collaboration between Microsoft and IBM. It is an extension of the RIFF and was first released in 1991. It is one of the earliest standards used for encoding audio bit streams on personal computers. It is typically used for storing uncompressed raw audio files on the Microsoft Windows platform, however it enjoys cross platform support on Macintosh and Linux and does have support for compressed audio. Because of the relatively large file sizes of uncompressed .wav files, the WAVE format in unpopular for file distribution over limited bandwidth computer networks including the internet.
FreeFileConvert uses tuned encoding for MP3 to WAV conversions, preserving clarity while trimming file size. Finished audio streams instantly across phones, tablets, desktops, and modern browsers without extra tweaks.
Upload MP3 files from desktop, tablet, or cloud storage, queue multiple jobs, and let the converter finish autonomously. Return whenever convenient to download synchronized WAV results on any device you rely on.
Process up to 5 files sized 1000 MB per batch without splitting queues manually. Mixed-format uploads convert together, producing consistent WAV audio with dependable progress tracking.
The MP3 format compression algorithm works through perceptual coding. A technique in which audio artifacts outside the auditory range of the average human, are reduced or 'cancelled out'. The result is a near perfect replication of the original audio sample at a compression ratio of 11 to 1 at 128 kilobits per second. Audio quality can be controlled by such factors as the bit rate, sample rate, among other features. But bits rates being the main determinant of audio quality can be adjusted between the typical range from 128 kb/s to as high as 320kb/s.
Bit stream encoding in the WAVE file format is achieved using the linear pulse code modulation format. It has three main data blocks and one to many number of wave chunks identified as the chunk ID, chunk size, wave ID, and finally the format information and the sampled data. Data storage is based on the little endian byte order.
Upload your audio file in the MP3 format from your device, Dropbox, or Google Drive.
Select WAV as the output format and click Convert. Adjust optional settings if needed.
Download the converted audio file. Each file stays available for up to 5 downloads.