MP3 (MPEG 1/ 2/ 2.5 Layer 3)
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;
- 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.
MP2 (MPEG 1 Layer 2)
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.
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.
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.
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. Softdiv MP3 to WAV Converter supports the PCM type of WAV.
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.
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.