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CELP
Code-excited linear prediction (CELP) is a linear predictive speech coding algorithm originally proposed by Manfred R. Schroeder and Bishnu S. Atal in 1985. At the time, it provided significantly better quality than existing low bit-rate algorithms, such as residual-excited linear prediction (RELP) and linear predictive coding (LPC) vocoders (e.g., FS-1015). Along with its variants, such as algebraic CELP, relaxed CELP, low-delay CELP and vector sum excited linear prediction, it is currently the most widely used speech coding algorithm. It is also used in MPEG-4 Audio speech coding. CELP is commonly used as a generic term for a class of algorithms and not for a particular codec. Background The CELP algorithm is based on four main ideas: * Using the source-filter model of speech production through linear prediction (LP) (see the textbook "speech coding algorithm"); * Using an adaptive and a fixed codebook as the input (excitation) of the LP model; * Performing a search in close ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Celp Decoder
Code-excited linear prediction (CELP) is a linear predictive speech coding algorithm originally proposed by Manfred R. Schroeder and Bishnu S. Atal in 1985. At the time, it provided significantly better quality than existing low bit-rate algorithms, such as residual-excited linear prediction (RELP) and linear predictive coding (LPC) vocoders (e.g., FS-1015). Along with its variants, such as algebraic CELP, relaxed CELP, low-delay CELP and vector sum excited linear prediction, it is currently the most widely used speech coding algorithm. It is also used in MPEG-4 Audio speech coding. CELP is commonly used as a generic term for a class of algorithms and not for a particular codec. Background The CELP algorithm is based on four main ideas: * Using the source-filter model of speech production through linear prediction (LP) (see the textbook "speech coding algorithm"); * Using an adaptive and a fixed codebook as the input (excitation) of the LP model; * Performing a search in closed- ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Comparison Of Audio Coding Formats
The following tables compare general and technical information for a variety of audio coding formats. For listening tests comparing the perceived audio quality of audio formats and codecs, see the article Codec listening test. General information Notes # The 'Music' category is merely a guideline on commercialized uses of a particular format, not a technical assessment of its capabilities. (For example, in terms of marketshare, MP3 and AAC dominate the personal audio market, 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, is the date of first binary releasing. Many developing codecs have pre-releases consisting of pre-1.0 versions and perhaps 1.0 release candidates (RCs), although 1.0 may not necessarily be the release version. # Latest stable version is that of ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Algebraic CELP
Algebraic code-excited linear prediction (ACELP) is a speech coding algorithm in which a limited set of pulses is distributed as excitation to a linear prediction filter. It is a linear predictive coding (LPC) algorithm that is based on the code-excited linear prediction (CELP) method and has an algebraic structure. ACELP was developed in 1989 by the researchers at the Université de Sherbrooke in Canada. The ACELP method is widely employed in current speech coding standards such as AMR, EFR, AMR-WB (G.722.2), VMR-WB, EVRC, EVRC-B, SMV, TETRA, PCS 1900, MPEG-4 CELP and ITU-T G-series standards G.729, G.729.1 (first coding stage) and G.723.1. The ACELP algorithm is also used in the proprietary ACELP.net codec. Audible Inc. use a modified version for their speaking books. It is also used in conference-calling software, speech compression tools and has become one of the 3GPP formats. The ACELP patent expired in 2018 and is now royalty-free. Features The main advantage of ACE ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
ACELP
Algebraic code-excited linear prediction (ACELP) is a speech coding algorithm in which a limited set of pulses is distributed as excitation to a linear prediction filter. It is a linear predictive coding (LPC) algorithm that is based on the code-excited linear prediction (CELP) method and has an algebraic structure. ACELP was developed in 1989 by the researchers at the Université de Sherbrooke in Canada. The ACELP method is widely employed in current speech coding standards such as AMR, EFR, AMR-WB (G.722.2), VMR-WB, EVRC, EVRC-B, SMV, TETRA, PCS 1900, MPEG-4 CELP and ITU-T G-series standards G.729, G.729.1 (first coding stage) and G.723.1. The ACELP algorithm is also used in the proprietary ACELP.net codec. Audible Inc. use a modified version for their speaking books. It is also used in conference-calling software, speech compression tools and has become one of the 3GPP formats. The ACELP patent expired in 2018 and is now royalty-free. Features The main advantage of ACE ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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] |
MPEG-4 Audio
MPEG-4 Part 3 or MPEG-4 Audio (formally International Organization for Standardization, ISO/International Electrotechnical Commission, IEC 14496-3) is the third part of the International Organization for Standardization, ISO/International Electrotechnical Commission, 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 compression, lossy speech coding (Harmonic Vector Excitation Coding, HVXC, CELP), general audio coding (Advanced Audio Coding, AAC, TwinVQ, BSAC), lossless compression, lossless audio compression (MPEG-4 SLS, Audio Lossless Coding, Direct Stream Transfer#DST, MPEG-4 DST), a Text to speech, Text-To-Speech Interface (TTSI), MPEG-4 Structured Audio, Structured Audio (using Structured Audio Orchestra Language, SAOL, SASL, MIDI) and many additional audio synthesis and codin ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Speex
Speex is an audio compression codec specifically tuned for the reproduction of human speech and also a free software speech codec that may be used on VoIP applications and podcasts. It is based on the CELP speech coding algorithm.Xiph.OrIntroduction to CELP Coding Retrieved 2009-09-01 Speex claims to be free of any patent restrictions and is licensed under the revised (3-clause) BSD license. It may be used with the Ogg container format or directly transmitted over UDP/ RTP. It may also be used with the FLV container format. The Speex designers see their project as complementary to the Vorbis general-purpose audio compression project. Speex is a lossy format, ''i.e.'' quality is permanently degraded to reduce file size. The Speex project was created on February 13, 2002. The first development versions of Speex were released under LGPL license, but as of version 1.0 beta 1, Speex is released under Xiph's version of the (revised) BSD license. Speex 1.0 was announced on March ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Low-delay CELP
G.728 is an ITU-T standard for speech coding operating at 16 kbit/s. It is officially described as ''Coding of speech at 16 kbit/s using low-delay code excited linear prediction''. Technology used is LD-CELP, low-delay code excited linear prediction. Delay of the codec is only 5 samples (0.625 ms). The linear prediction is calculated backwards with a 50th order linear predictive coding filter. The excitation is generated with gain scaled VQ. The standard was finished in 1992 in the form of algorithm exact floating point code. In 1994 a bit exact fixed point codec was released. G.728 passes low bit rate modem signals up to 2400 bit/s. Also network signaling goes through. The complexity of the codec is 30 MIPS. 2 kilobytes of RAM is needed for codebooks. Mean opinion score for G.728 is 3.61. The essence of CELP techniques, which is an analysis-by-synthesis approach to codebook search, is retained in LD-CELP. The LD-CELP however, uses backward adaptation of predictors an ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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, ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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, ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Residual-excited Linear Prediction
Residual-excited linear prediction (RELP) is an obsolete speech coding algorithm. It was originally proposed in the 1970s and can be seen as an ancestor of code-excited linear prediction (CELP). Unlike CELP however, RELP directly transmits the residual signal. To achieve lower rates, that residual signal is usually down-sampled (e.g. to 1–2 kHz). The algorithm is hardly used anymore in audio transmission. It is still used in some text-to-speech voices, such as the diphone databases found in the Festival A festival is an event ordinarily celebrated by a community and centering on some characteristic aspect or aspects of that community and its religion or cultures. It is often marked as a local or national holiday, mela, or eid. A festival c ... and Flite speech synthesizers. References External links * https://harvest.usask.ca/bitstream/handle/10388/etd-12202003-142739/0105Thesis.pdf?sequence=1&isAllowed=y (5.8 MB) Speech codecs {{tech-stub ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
