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A lossless audio coding format reduces the total data needed to represent a sound but can be de-coded to its original, uncompressed form. A lossy audio coding format additionally reduces the bit resolution of the sound on top of compression, which results in far less data at the cost of irretrievably lost information.
For example, MP3 and AAC dominate the personal audio market in terms of market share, though many other formats are comparably well suited to fill this role from a purely technical standpoint. First public release date is first of either specification publishing or source releasing, or in the case of closed-specification, closed-source codecs ...
Linear pulse-code modulation (LPCM, generally only described as PCM) is the format for uncompressed audio in media files and it is also the standard for CD-DA; note that in computers, LPCM is usually stored in container formats such as WAV, AIFF, or AU, or as raw audio format, although not technically necessary.
Speech coding is an application of data compression to digital audio signals containing speech.Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream.
Hardware audio codecs send and receive digital data using buses such as AC-Link, I²S, SPI, I²C, etc. Most commonly the digital data is linear PCM , and this is the only format that most codecs support, but some legacy codecs support other formats such as G.711 for telephony.
Unified Speech and Audio Coding (USAC) is an audio compression format and codec for both music and speech or any mix of speech and audio using very low bit rates between 12 and 64 kbit/s. [1] It was developed by Moving Picture Experts Group (MPEG) and was published as an international standard ISO / IEC 23003-3 (a.k.a. MPEG-D Part 3) [ 2 ] and ...
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The Adaptive Multi-Rate (AMR, AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding.AMR is a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality [3] speech starting at 7.4 kbit/s.