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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 replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC. Opus combines the speech-oriented LPC-based SILK algorithm and the lower-latency MDCT-based CELT algorithm, switching between or combining them as needed for maximal efficiency. Bitrate, audio bandwidth, complexity, and algorithm can all be adjusted seamlessly in each frame. Opus has the low algorithmic delay (26.5 ms by default) necessary for use as part of a real-time communication link, network ...
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Libopus 1
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 replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC. Opus combines the speech-oriented LPC-based SILK algorithm and the lower-latency MDCT-based CELT algorithm, switching between or combining them as needed for maximal efficiency. Bitrate, audio bandwidth, complexity, and algorithm can all be adjusted seamlessly in each frame. Opus has the low algorithmic delay (26.5 ms by default) necessary for use as part of a real-time communication link, networked m ...
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Audio File Format
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. The data can be a raw bitstream in an audio coding format, but it is usually embedded in a container format or an audio data format with defined storage layer. Format types It is important to distinguish between the audio coding format, the container containing the raw audio data, and an audio codec. A codec performs the encoding and decoding of the raw audio data while this encoded data is (usually) stored in a container file. Although most audio file formats support only one type of audio coding data (created with an audio coder), a multimedia container format (as Matroska or AVI) may support multiple types of audio and video data. There are three major groups of audio file formats: * Uncompressed audio fo ...
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Opus Logo2
''Opus'' (pl. ''opera'') is a Latin word meaning "work". Italian equivalents are ''opera'' (singular) and ''opere'' (pl.). Opus or OPUS may refer to: Arts and entertainment Music * Opus number, (abbr. Op.) specifying order of (usually) publication, and hence applied to collections as well as individual compositions * OPUS Records, a Slovak record label * Opus 111, a French classical record label bought by Naive Bands * Opus (Austrian band), an Austrian pop-rock group * Opus (Yugoslav band), a former Yugoslav progressive rock group * Opus III (band), an English electronic pop group * Pur (band), a German pop group originally known as "Opus" Albums * ''Opus'' (Opus album), 1987 album by Austrian band Opus * ''Opus'' (Schiller album), 2013 album by German music project Schiller * ''Opus'', a 2014 album by Jane Badler * ''Opus'' (Eric Prydz album), 2016 album by the electronic artist Eric Prydz ** "Opus" (Eric Prydz song), song from the eponymous album. * ''Opus'', a 2007 ...
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Speech Coding
Speech coding is an application of data compression of 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. Some 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. The techniques employed in speech coding are similar to those used in audio data compression and audio coding where knowledge in psychoacoustics is used to transmit only data that is relevant to the human auditory system. For example, in voiceband speech coding, only information in the frequency band 400 to 3500 Hz is transmitted but the reconst ...
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Open Format
An open file format is a file format for storing digital data, defined by an openly published specification usually maintained by a standards organization, and which can be used and implemented by anyone. Open file format is licensed with open license. For example, an open format can be implemented by both proprietary and free and open-source software, using the typical software licenses used by each. In contrast to open file formats, closed file formats are considered trade secrets. However, the actual image used by an open file format may still be copyrighted or trademarked. Depending on the definition, the specification of an open format may require a fee to access or, very rarely, contain other restrictions. The range of meanings is similar to that of the term open standard. Specific definitions Sun Microsystems Sun Microsystems defined the criteria for open formats as follows:
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Lip Sync
Lip sync or lip synch (pronounced , the same as the word ''sink'', short for lip synchronization) is a technical term for matching a speaking or singing person's lip movements with sung or spoken vocals. Audio for lip syncing is generated through the sound reinforcement system in a live performance or via television, computer, cinema speakers, or other forms of audio output. The term can refer to any of a number of different techniques and processes, in the context of live performances and audiovisual recordings. In film production, lip syncing is often part of the post-production phase. Dubbing foreign-language films and making animated characters appear to speak both require elaborate lip syncing. Many video games make extensive use of lip-synced sound files to create an immersive environment in which on-screen characters appear to be speaking. In the music industry, lip syncing is used by singers for music videos, television and film appearances and some types of live perf ...
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Networked Music Performance
A networked music performance or network musical performance is a real-time interaction over a computer network that enables musicians in different locations to perform as if they were in the same room. These interactions can include performances, rehearsals, improvisation or jamming sessions, and situations for learning such as master classes. Participants may be connected by "high fidelity multichannel audio and video links" as well as MIDI data connections and specialized collaborative software tools. While not intended to be a replacement for traditional live stage performance, networked music performance supports musical interaction when co-presence is not possible and allows for novel forms of music expression. Remote audience members and possibly a conductor may also participate. History One of the earliest examples of a networked music performance experiment was the 1951 piece: “ Imaginary Landscape No. 4 for Twelve Radios” by composer John Cage. The piece “used radi ...
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Modified Discrete Cosine Transform
The modified discrete cosine transform (MDCT) is a transform based on the type-IV discrete cosine transform (DCT-IV), with the additional property of being lapped: it is designed to be performed on consecutive blocks of a larger dataset, where subsequent blocks are overlapped so that the last half of one block coincides with the first half of the next block. This overlapping, in addition to the energy-compaction qualities of the DCT, makes the MDCT especially attractive for signal compression applications, since it helps to avoid artifacts stemming from the block boundaries. As a result of these advantages, the MDCT is the most widely used lossy compression technique in audio data compression. It is employed in most modern audio coding standards, including MP3, Dolby Digital (AC-3), Vorbis (Ogg), Windows Media Audio (WMA), ATRAC, Cook, Advanced Audio Coding (AAC), High-Definition Coding (HDC), LDAC, Dolby AC-4, and MPEG-H 3D Audio, as well as speech coding standards such as A ...
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Linear Predictive Coding
Linear predictive coding (LPC) is a method used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model. LPC is the most widely used method in speech coding and speech synthesis. It is a powerful speech analysis technique, and a useful method for encoding good quality speech at a low bit rate. Overview LPC starts with the assumption that a speech signal is produced by a buzzer at the end of a tube (for voiced sounds), with occasional added hissing and popping sounds (for voiceless sounds such as sibilants and plosives). Although apparently crude, this Source–filter model is actually a close approximation of the reality of speech production. The glottis (the space between the vocal folds) produces the buzz, which is characterized by its intensity (loudness) and frequency (pitch). The vocal tract (the throat and mouth) forms the tube, wh ...
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High-Efficiency Advanced Audio Coding
High-Efficiency Advanced Audio Coding (HE-AAC) is an audio coding format for lossy data compression of digital audio defined as an MPEG-4 Audio profile in ISO/ IEC 14496–3. 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. HE-AAC is used in digital radio standards like HD Radio, DAB+ and Digital Radio Mondiale. History The progenitor of HE-AAC was developed by Coding Technologies by combining MPEG-2 AAC-LC with a proprietary mechanism for spectral band replication (SBR), to be used by XM Radio for their satellite radio service. Subsequently, Coding Technologies submitted their SBR mechanism to MPEG as a basis of ...
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Advanced Audio Coding
Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. Designed to be the successor of the MP3 format, AAC generally achieves higher sound quality than MP3 encoders at the same bit rate. AAC has been standardized by ISO and IEC as part of the MPEG-2 and MPEG-4 specifications.ISO (2006ISO/IEC 13818-7:2006 - Information technology -- Generic coding of moving pictures and associated audio information -- Part 7: Advanced Audio Coding (AAC), Retrieved on 2009-08-06ISO (2006, Retrieved on 2009-08-06 Part of AAC, HE-AAC ("AAC+"), is part of MPEG-4 Audio and is adopted into digital radio standards DAB+ and Digital Radio Mondiale, and mobile television standards DVB-H and ATSC-M/H. AAC supports inclusion of 48 full-bandwidth (up to 96 kHz) audio channels in one stream plus 16 low frequency effects ( LFE, limited to 120 Hz) channels, up to 16 "coupling" or dialog channels, and up to 16 data streams. The quality for stereo is satisfactory to ...
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Transparency (data Compression)
In data compression and psychoacoustics, transparency is the result of lossy data compression accurate enough that the compressed result is perceptually indistinguishable from the uncompressed input, i.e. perceptually lossless. A transparency threshold is a given value at which transparency is reached. It is commonly used to describe compressed data bitrates. For example, the transparency threshold for MP3 to linear PCM audio is said to be between 175 and 245 kbit/s, at 44.1 kHz, when encoded as VBR MP3 (corresponding to the -V3 and -V0 settings of the highly popular LAME MP3 encoder). This means that when an MP3 that was encoded at those bitrates is being played back, it is indistinguishable from the original PCM, and the compression is transparent to the listener. The term ''transparent compression'' can also refer to a filesystem feature that allows compressed files to be read and written just like regular ones. In this case, the compressor is typically a general- ...
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