Various kind of audio file formats and its utility?
An audio file format is a file format for storing digital audio data on a computer system. The bit layout of the audio data (excluding metadata) is called the audio coding format and can be uncompressed, or compressed to reduce the file size, often using lossy compression. An audio recording produced from original analog or digital audio formats that have been encoded using linear Pulse Code Modulation (PCM). For audio files, it is important to distinguish between a file format and a codec. A codec performs the encoding and decoding of the raw audio data while the data itself is stored in a specific audio file format. Sampling frequency, bit-depth, and monophonic or stereo, for example, are important characteristics of audio files. There are a handful of audio file types you should be familiar with if you are planning to copy music off the Internet or even copy a CD. In this article, we will discuss the types of audio file formats and its utility. So, let's check out types of audio file -
Waveform Audio File Format is a Microsoft and IBM audio file format standard for storing an audio bitstream on PCs. It is an application of the Resource Interchange File Format (RIFF) bitstream format method for storing data in "chunks", and thus is also close to the 8SVX and the AIFF format used on Amiga and Macintosh computers, respectively. It is the main format used on Microsoft Windows systems for raw and typically uncompressed audio. The usual bitstream encoding is the linear pulse-code modulation (LPCM) format. WAVs are compatible with Microsoft Windows, Macintosh, and Linux operating systems. The format takes into account some differences of the Intel CPU such as little-endian byte order.
Audio Interchange File Format (AIFF) is an audio file format standard used for storing sound data for personal computers and other electronic audio devices. The format was developed by Apple Inc. in 1988 based on Electronic Arts' Interchange File Format (IFF, widely used on Amiga systems) and is most commonly used on Apple Macintosh computer systems. The file extension for the standard AIFF format is .aiff or .aif. For the compressed variants it is supposed to be .aifc, but .aiff or .aif are accepted as well by audio applications supporting the format.
The Au file format is a simple audio file format introduced by Sun Microsystems. The format was common on NeXT systems and on early Web pages. Originally it was headerless, being simply 8-bit µ-law-encoded data at an 8000 Hz sample rate. Newer files have a header that consists of six unsigned 32-bit words, an optional information chunk and then the data (in big-endian format). The format now supports many audio encoding formats, it remains associated with the µ-law logarithmic encoding.
RAW Audio format or just RAW Audio is an audio file format for storing uncompressed audio in raw form. Comparable to WAV or AIFF in size, the RAW Audio file does not include any header information. Raw files can have a wide range of file extensions, common ones being .raw, .pcm, or .sam.
WavPack is a free and open-source lossless audio compression format. WavPack compression (.WV files) can compress and restore 8-, 16-, 24-, and 32-bit fixed-point, and 32-bit floating point audio files in the .WAV file format. It also supports surround sound streams and high-frequency sampling rates. Like other lossless compression schemes, the data reduction rate varies with the source, but it is generally between 30% and 70% for typical popular music and somewhat better than that for classical music and other sources with greater dynamic range.
True Audio (TTA) is a lossless compressor for multichannel 8, 16 and 24 bits audio data. .tta is the extension to filenames of audio files created by the True Audio codec. TTA performs lossless compression on multichannel 8, 16 and 24-bit data of uncompressed WAV input files. The term "lossless" refers to the fact that such compression results in no data or quality loss; when decompressed, the audio file data are bit-identical to those of their originals. Compression ratios achieved by the TTA codec vary, depending on music type, but range from 30% to 70% of the original.
Apple Lossless, also known as Apple Lossless Audio Codec (ALAC), or Apple Lossless Encoder (ALE), is an audio coding format, and its reference audio codec implementation, developed by Apple Inc. for lossless data compression of digital music. After initially keeping it proprietary from its inception in 2004, in late 2011 Apple made the codec available open source and royalty-free. Traditionally, Apple has referred to the codec as Apple Lossless, though more recently it has begun to use the abbreviated term ALAC when referring to the codec. Apple Lossless supports up to 8 channels of audio at 16, 20, 24 and 32-bit depth with a maximum sample rate of 384kHz. Apple Lossless data is frequently stored within an MP4 container with the filename extension .m4a. This extension is also used by Apple for lossy AAC audio data in an MP4 container.
MPEG-4 is an extension to the MPEG-4 Part 3 (MPEG-4 Audio) standard to allow lossless audio compression scalable to lossy MPEG-4 General Audio coding methods (e.g., variations of AAC). It was developed jointly by the Institute for Infocomm Research (I2R) and Fraunhofer, which commercializes its implementation of a limited subset of the standard under the name of HD-AAC. MPEG-4 allows having both a lossy layer and a lossless correction layer similar to Wavpack Hybrid, OptimFROG DualStream, and DTS-HD Master Audio, providing backward compatibility to MPEG AAC-compliant bitstreams.
Windows Media Audio (WMA) is the name of a series of audio codecs and their corresponding audio coding formats developed by Microsoft. It is a proprietary technology that forms part of the Windows Media framework. WMA consists of four distinct codecs. The original WMA codec, known simply as WMA, was conceived as a competitor to the popular MP3 and RealAudio codecs. Microsoft has also developed a digital container format called Advanced Systems Format to store audio encoded by WMA.
Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors. Opus supports constant and variable bitrate encoding from 6 kbit/s to 510 kbit/s, frame sizes from 2.5 ms to 120 ms, and five sampling rates from 8 kHz (with 4 kHz bandwidth) to 48 kHz. An Opus stream can support up to 255 audio channels, and it allows channel coupling between channels in groups of two using mid-side coding.
MP3 is an audio coding format for digital audio. MP3 (or mp3) as a file format commonly designates files containing an elementary stream of MPEG-1 audio and video encoded data, without other complexities of the MP3 standard. MP3 was designed by the Moving Picture Experts Group (MPEG) as part of its MPEG-1, and later MPEG-2, standards. This allows a large reduction in file size when compared to uncompressed audio.
Musepack or MPC is an open source lossy audio codec, specifically optimized for transparent compression of stereo audio at bitrates of 160-180 (manual set allows bit rates up to 320) kbit/s. It was formerly known as MPEG plus, MPEG+ or MP+. Development of MPC was initiated in 1997 by Andree Buschmann and later assumed by Frank Klemm, and as of 2004 is maintained by the Musepack Development Team (MDT) with assistance from Buschmann and Klemm. Encoders and decoders are available for Microsoft Windows, Linux and Mac OS X, and plugins for several third-party media players available from the Musepack website, licensed under the GNU Lesser General Public License (LGPL) or BSD licenses, and an extensive list of programs supporting the format.
Advanced Audio Coding (AAC) is a proprietary audio coding standard for lossy digital audio compression. Designed to be the successor of the MP3 format, AAC generally achieves better sound quality than MP3 at the same bit rate. The confusingly named AAC+ (HE-AAC) do so[clarification needed] only at low bit rates and less so at high ones. AAC has been standardized by ISO and IEC, as part of the MPEG-2 and MPEG-4 specifications. Part of AAC, HE-AAC (AAC+), is part of MPEG-4 Audio and also adopted into digital radio standards DAB+ and Digital Radio Mondiale, as well as mobile television standards DVB-H and ATSC-M/H. AAC supports the inclusion of 48 full-bandwidth (up to 96 kHz) audio channels in one stream plus 16 low-frequency effects (LFE, limited to 120Hz) channels, up to 16 "coupling" or dialog channels, and up to 16 data streams. The quality for stereo is satisfactory to modest requirements at 96 kbit/s in joint stereo mode; however, hi-fi transparency demands data rates of at least 128 kbit/s (VBR).