Modified discrete cosine transform in the context of "CELT"

Opus is a free and open source lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed for efficient low-latency encoding of both speech and general audio. Due to its lower latency relative to other standard codecs, Opus finds specific use cases in real-time interactive communication for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications.

Opus combines the speech-oriented LPC-based SILK algorithm and the lower-latency MDCT-based CELT algorithm, switching between or combining them as needed. Bitrate, audio bandwidth, complexity, and algorithm choice can be adjusted for each individual frame. Opus has low algorithmic delay (26.5 ms by default) ideal for use as part of a real-time communication link, networked music performances, and live lip sync; by trading off quality or bitrate, the delay can be further reduced down to 5 ms. Its delay thus is significantly lower compared to competing codecs, which require well over 100 ms. Opus remains competitive with these formats in terms of quality per bitrate.

Dolby Digital, originally synonymous with Dolby AC-3 (see below), is the name for a family of audio compression technologies developed by Dolby Laboratories. Called Dolby Stereo Digital until 1995, it uses lossy compression (except for Dolby TrueHD). The first use of Dolby Digital was to provide digital sound in cinemas from 35 mm film prints. It has since also been used for TV broadcast, radio broadcast via satellite, digital video streaming, DVDs, Blu-ray discs and game consoles.

Dolby AC-3 was the original version of the Dolby Digital codec. The basis of the Dolby AC-3 multi-channel audio coding standard is the modified discrete cosine transform (MDCT), a lossy audio compression algorithm. It is a modification of the discrete cosine transform (DCT) algorithm, which was proposed by Nasir Ahmed in 1972 for image compression. The DCT was adapted into the MDCT by J.P. Princen, A.W. Johnson and Alan B. Bradley at the University of Surrey in 1987.

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. 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) and also as an MPEG-4 Audio Object Type in ISO/IEC 14496-3:2009/Amd 3 in 2012.

It uses time-domain linear prediction and residual coding tools (ACELP-like techniques) for speech signal segments and transform coding tools (MDCT-based techniques) for music signal segments and it is able to switch between the tool sets dynamically in a signal-responsive manner. It is being developed with the aim of a single, unified coder with performance that equals or surpasses that of dedicated speech coders and dedicated music coders over a broad range of bitrates. Enhanced variations of the MPEG-4 Spectral Band Replication (SBR) and MPEG-D MPEG Surround parametric coding tools are integrated into the USAC codec.

A music download is the digital transfer of music via the Internet into a device capable of decoding and playing it, such as a personal computer, portable media player, MP3 player or smartphone. This term encompasses both legal downloads and downloads of copyrighted material without permission or legal payment. Music downloads are typically encoded with the MP3 audio coding format. or using the modified discrete cosine transform (MDCT) audio data compression, particularly the Advanced Audio Coding (AAC) format used by iTunes.

Since the advent of streaming, downloads as a mode of music distribution has seen a steady decline from its peak in the early 2010s. According to a Nielsen report, downloadable music accounted for 55.9 percent of all music sales in the US in 2012. By the beginning of 2011, Apple's iTunes Store alone made US$1.1 billion of revenue in the first quarter of its fiscal year. According to the RIAA, music downloads peaked at 43% of industry revenue in the US in 2012, and has since fallen to 3% in 2022.

An audio codec is a device or computer program capable of encoding or decoding a digital data stream (a codec) that encodes or decodes audio. In software, an audio codec is a computer program implementing an algorithm that compresses and decompresses digital audio data according to a given audio file or streaming media audio coding format. The objective of the algorithm is to represent the high-fidelity audio signal with a minimum number of bits while retaining quality. This can effectively reduce the storage space and the bandwidth required for transmission of the stored audio file. Most software codecs are implemented as libraries which interface to one or more multimedia players. Most modern audio compression algorithms are based on modified discrete cosine transform (MDCT) coding and linear predictive coding (LPC).

In hardware, audio codec refers to a single device that encodes analog audio as digital signals and decodes digital back into analog. In other words, it contains both an analog-to-digital converter (ADC) and digital-to-analog converter (DAC) running off the same clock signal. This is used in sound cards that support both audio in and out, for instance. Hardware audio codecs send and receive digital data using buses such as AC'97, SoundWire, 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.

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.

Common applications of speech coding are mobile telephony and voice over IP (VoIP). The most widely used speech coding technique in mobile telephony is linear predictive coding (LPC), while the most widely used in VoIP applications are the LPC and modified discrete cosine transform (MDCT) techniques.

High-Efficiency Advanced Audio Coding (HE-AAC) is an audio coding format for lossy data compression of digital audio as part of the MPEG-4 standards. It is an extension of Low Complexity AAC (AAC-LC) optimized for low-bitrate applications such as streaming audio.

The usage profile HE-AAC v1 uses spectral band replication (SBR) to enhance the modified discrete cosine transform (MDCT) compression efficiency in the frequency domain. The usage profile HE-AAC v2 couples SBR with Parametric Stereo (PS) to further enhance the compression efficiency of stereo signals.

The MPEG-4 Low Delay Audio Coder (a.k.a. AAC Low Delay, or AAC-LD) is audio compression standard designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It is closely derived from the MPEG-2 Advanced Audio Coding (AAC) standard. It was published in MPEG-4 Audio Version 2 (ISO/IEC 14496-3:1999/Amd 1:2000) and in its later revisions.

AAC-LD uses a version of the modified discrete cosine transform (MDCT) audio coding technique called the LD-MDCT. AAC-LD is widely used by Apple as the voice-over-IP (VoIP) speech codec in FaceTime.