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MPEG-4 Part 3
MPEG-4 Part 3 or MPEG-4 Audio (formally ISO/IEC 14496-3) is the third part of the ISO/IEC MPEG-4 international standard developed by Moving Picture Experts Group. It specifies audio coding methods. The first version of ISO/IEC 14496-3 was published in 1999. The MPEG-4 Part 3 consists of a variety of audio coding technologies – from lossy speech coding (HVXC, CELP), general audio coding (AAC, TwinVQ, BSAC), lossless audio compression (MPEG-4 SLS, Audio Lossless Coding, MPEG-4 DST), a Text-To-Speech Interface (TTSI), Structured Audio (using SAOL, SASL, MIDI) and many additional audio synthesis and coding techniques. MPEG-4 Audio does not target a single application such as real-time telephony or high-quality audio compression. It applies to every application which requires the use of advanced sound compression, synthesis, manipulation, or playback. MPEG-4 Audio is a new type of audio standard that integrates numerous different types of audio coding: natural sound and synthetic s ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
International Organization For Standardization
The International Organization for Standardization (ISO ) is an international standard development organization composed of representatives from the national standards organizations of member countries. Membership requirements are given in Article 3 of the ISO Statutes. ISO was founded on 23 February 1947, and (as of November 2022) it has published over 24,500 international standards covering almost all aspects of technology and manufacturing. It has 809 Technical committees and sub committees to take care of standards development. The organization develops and publishes standardization in all technical and nontechnical fields other than electrical and electronic engineering, which is handled by the IEC.Editors of Encyclopedia Britannica. 3 June 2021.International Organization for Standardization" ''Encyclopedia Britannica''. Retrieved 2022-04-26. It is headquartered in Geneva, Switzerland, and works in 167 countries . The three official languages of the ISO are English, Fren ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Structured Audio Orchestra Language
Structured Audio Orchestra Language (SAOL) is an imperative, MUSIC-N programming language designed for describing virtual instruments, processing digital audio, and applying sound effects. It was published as subpart 5 of MPEG-4 Part 3 (ISO/IEC 14496-3:1999) in 1999. As part of the MPEG-4 international standard, SAOL is one of the key components of the MPEG-4 Structured Audio toolset, along with: * Structured Audio Score Language (SASL) * Structured Audio Sample Bank Format (SASBF) * The MPEG-4 SA scheduler * MIDI support See also * Csound * MPEG-4 Structured Audio MPEG-4 Structured Audio is an ISO/IEC standard for describing sound. It was published as subpart 5 of MPEG-4 Part 3 (ISO/ IEC 14496-3:1999) in 1999. It allows the transmission of synthetic music and sound effects at very low bit rates (from 0.01 ... References The MPEG-4 Structured Audio Standard External links SAOL.net - MPEG4 structured audio (mp4-sa) Audio programming languages MPEG {{compu-lang-s ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Table-lookup Synthesis
Wavetable synthesis is a sound synthesis technique used to create quasi-periodic waveforms often used in the production of musical tones or notes. Development Wavetable synthesis was invented by Max Mathews in 1958 as part of MUSIC II. MUSIC II “had four-voice polyphony and was capable of generating sixteen wave shapes via the introduction of a wavetable oscillator.” Hal Chamberlin discussed wavetable synthesis in Byte's September 1977 issue. Wolfgang Palm of Palm Products GmbH (PPG) developed his version in the late 1970s and published it in 1979. The technique has since been used as the primary synthesis method in synthesizers built by PPG and Waldorf Music and as an auxiliary synthesis method by Ensoniq and Access. It is currently used in hardware synthesizers from Waldorf Music and in software synthesizers for PCs and tablets, including apps offered by PPG and Waldorf, among others. It was also independently developed by Michael McNabb, who used it in his 1978 co ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
HE-AAC
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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Spectral Band Replication
Spectral band replication (SBR) is a technology to enhance audio or speech codecs, especially at low bit rates and is based on harmonic redundancy in the frequency domain. It can be combined with any audio compression codec: the codec itself transmits the lower and midfrequencies of the spectrum, while SBR replicates higher frequency content by transposing up harmonics from the lower and midfrequencies at the decoder. Some guidance information for reconstruction of the high-frequency spectral envelope is transmitted as side information. When needed, it also reconstructs or adaptively mixes in noise-like information in selected frequency bands in order to faithfully replicate signals that originally contained no or fewer tonal components. The SBR idea is based on the principle that the psychoacoustic part of the human brain tends to analyse higher frequencies with less accuracy; thus harmonic phenomena associated with the spectral band replication process needs only be accurate in ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Long Term Prediction
In GSM, a Regular Pulse Excitation-Long Term Prediction (RPE-LTP) scheme is employed in order to reduce the amount of data sent between the mobile station (MS) and base transceiver station (BTS). In essence, when a voltage level of a particular speech sample is quantified, the mobile station's internal logic predicts the voltage level for the next sample. When the next sample is quantified, the packet sent by the MS to the BTS contains only the error (the signed difference between the actual and predicted level of the sample). See also * GSM * Quantization (signal processing) Quantization, in mathematics and digital signal processing, is the process of mapping input values from a large set (often a continuous set) to output values in a (countable) smaller set, often with a finite number of elements. Rounding and ... GSM standard {{mobile-tech-stub ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
AAC-LC
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 mo ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Scalable Lossless Coding
MPEG-4 SLS, or MPEG-4 Scalable to Lossless as per ISO/IEC 14496-3:2005/Amd 3:2006 (Scalable Lossless Coding), 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. Standardization of the HD-AAC profile for MPEG-4 Audio is under development (as of September 2009). MPEG-4 SLS allows having both a lossy layer and a lossless correction layer similar to Wavpack Hybrid, OptimFROG DualStream and DTS-HD Master Audio, providing backwards compatibility to MPEG AAC-compliant bitstreams. MPEG-4 SLS can also work without a lossy layer (a.k.a. "SLS Non-Core"), in which case it will not be backwards compatible, Lossy compression of files is necessary for files that need to be streamed to ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
MPEG-2
MPEG-2 (a.k.a. H.222/H.262 as was defined by the ITU) is a standard for "the generic video coding format, coding of moving pictures and associated audio information". It describes a combination of Lossy compression, lossy video compression and Lossy compression, lossy audio data compression methods, which permit storage and transmission of movies using currently available storage media and transmission bandwidth. While MPEG-2 is not as efficient as newer standards such as H.264/AVC and HEVC, H.265/HEVC, backwards compatibility with existing hardware and software means it is still widely used, for example in over-the-air digital television broadcasting and in the DVD-Video standard. Main characteristics MPEG-2 is widely used as the format of digital television signals that are broadcast by terrestrial television, terrestrial (over-the-air), Cable television, cable, and direct broadcast satellite Television, TV systems. It also specifies the format of movies and other programs th ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
MPEG-1
MPEG-1 is a standard for lossy compression of video and audio. It is designed to compress VHS-quality raw digital video and CD audio down to about 1.5 Mbit/s (26:1 and 6:1 compression ratios respectively) without excessive quality loss, making video CDs, digital cable/satellite TV and digital audio broadcasting (DAB) practical. Today, MPEG-1 has become the most widely compatible lossy audio/video format in the world, and is used in a large number of products and technologies. Perhaps the best-known part of the MPEG-1 standard is the first version of the MP3 audio format it introduced. The MPEG-1 standard is published as ISO/IEC 11172 – Information technology—Coding of moving pictures and associated audio for digital storage media at up to about 1.5 Mbit/s. The standard consists of the following five ''Parts'': #Systems (storage and synchronization of video, audio, and other data together) #Video (compressed video content) #Audio (compressed audio content) #Conformance tes ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Parametric Stereo
Parametric Stereo (abbreviated as PS) is an audio compression algorithm used as an audio coding format for digital audio. It is considered an Audio Object Type of MPEG-4 Part 3 (MPEG-4 Audio) that serves to enhance the coding efficiency of low bandwidth stereo audio media. Parametric Stereo digitally codes a stereo audio signal by storing the audio as monaural alongside a small amount of extra information. This extra information (defined as "parametric overhead") describes how the monaural signal will behave across both stereo channels, which allows for the signal to exist in true stereo upon playback. History Background Advanced Audio Coding Low Complexity (AAC LC) combined with Spectral Band Replication (SBR) and Parametric Stereo (PS) was defined as HE-AAC v2. An HE-AAC v1 decoder will only give mono sound when decoding an HE-AAC v2 bitstream. Parametric Stereo performs sparse coding in the spatial domain, somewhat similar to what SBR does in the frequency domain. An AAC HE ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Harmonic And Individual Lines And Noise
{{Unreferenced, date=December 2017 Harmonic and Individual Lines and Noise (HILN) is a parametric codec for audio. The basic premise of the encoder is that most audio, and particularly speech, can be synthesized from only sinusoids and noise. The encoder describes individual sinusoids with amplitude and frequency, harmonic tones by fundamental frequency, amplitude and the spectral envelope of the partials, and the noise by amplitude and spectral envelope. This type of encoder is capable of encoding audio to between 6 and 16 kilobits per second for a typical audio bandwidth of 8 kHz. The framelength of this encoder is 32 ms. A typical codec extracts sinusoid information from the samples by applying a short-time Fourier transform to the samples and using that to find the important harmonic content of a single frame. By matching sinusoids across frames, the encoder is capable of grouping them into harmonic lines and individual sinusoids. The matching can take amplitude, frequency a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
