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Encyclopedia > Speech data compression

Speech coding is the 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. “Source coding” redirects here. ... Digital audio comprises audio signals stored in a digital format. ... This article or section does not cite any references or sources. ... Estimation theory is a branch of statistics and signal processing that deals with estimating the values of parameters based on measured/empirical data. ... This article or section does not cite its references or sources. ...


The two most important applications of speech coding are mobile telephony and Voice over IP. Cellular redirects here. ... An overview of how VoIP works A typical analog telephone adapter for connecting an ordinary phone to a VoIP network Ciscos implementation of VoIP - IP Phone Voice over Internet Protocol, also called VoIP, IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband is the routing of...


The techniques used in speech coding are similar to that 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 narrowband speech coding, only information in the frequency band 400 Hz to 3500 Hz is transmitted but the reconstructed signal is still adequate for intelligibility. Audio compression is a form of data compression designed to reduce the size of audio files. ... Psychoacoustics is the study of subjective human perception of sounds. ... Narrowband (narrow bandwidth) refers to a signal which occupies only a small amount of space on the radio spectrum -- the opposite of broadband or wideband. ...


Speech coding differs from other forms of audio coding in that speech is a much simpler signal than most other audio signals, and that there is a lot more statistical information available about the properties of speech. As a result, some auditory information which is relevant in audio coding can be unnecessary in the speech coding context. In speech coding, the most important criterion is preservation of intelligibility and "pleasantness" of speech, with a constrained amount of transmitted data.


It should be emphasised that the intelligibility of speech includes, besides the actual literal content, also speaker identity, emotions, intonation, timbre etc. that are all important for perfect intelligibility. The more abstract concept of pleasantness of degraded speech is a different property than intelligibility, since it is possible that degraded speech is completely intelligible, but subjectively annoying to the listener. In music, timbre, or sometimes timber, (from Fr. ...


In addition, most speech applications require low coding delay, as long coding delays interfere with speech interaction.


Sample companding viewed as a form of speech coding

From this viewpoint, the A-law and μ-law algorithms used in traditional PCM digital telephony can be seen as a very early precursor of speech encoding, requiring only 8 bits per sample but giving effectively 12 bits of resolution. Although this would generate unacceptable distortion in a music signal, the peaky nature of speech waveforms, combined with the simple frequency structure of speech as a periodic waveform with a single fundamental frequency with occasional added noise bursts, make these very simple instantaneous compression algorithms acceptable for speech. Graph of μ-law & A-law algorithms An a-law algorithm is a standard companding algorithm, used in European digital communications systems to optimize, modify, the dynamic range of an analog signal for digitizing. ... In telecommunication, a mu-law algorithm (μ-law) is a standard analog signal compression or companding algorithm, used in digital communications systems of the North American and Japanese digital hierarchies, to optimize (in other words, modify) the dynamic range of an audio analog signal prior to digitizing. ... PCM is an initialism which can have different meanings: Phase Change Material Pulse-code modulation, a way to digitally encode signals representing sound and their video counterparts Potential Cancer Marker Communist Party of Mexico Plug Compatible Manufacturer Power-train control module, a computer in a car which controls the car... Digital telephony is a technology used in the provision of digital telephone services and systems. ...


A wide variety of other algorithms were tried at the time, mostly variants on delta modulation, but after careful consideration, the A-law/μ-law algorithms were chosen by the designers of the early digital telephony systems. At the time of their design, their 33% bandwidth reduction for a very low complexity made them an excellent engineering compromise. Their audio performance remains acceptable, and there has been no need to replace them. The Delta modulation (DM or Δ-modulation) is an analog-to-digital or digital-to-analog signal conversion. ...


Modern speech compression

Much of the later work in speech compression was motivated by military research into digital communications for secure military radios, where very low data rates were required to allow effective operation in a hostile radio environment. At the same time, far more processing power was available, in the form of VLSI integrated circuits, than was available for earlier compression techniques. As a result, modern speech compression algorithms could use far more complex techniques than were available in the 1960s to achieve far higher compression ratios.


These techniques were available through the open research literature to be used for civilian applications, allowing the creation of digital mobile phone networks with substantially higher channel capacities than the analog systems that preceded them.


The most common speech coding scheme is Code Excited Linear Prediction (CELP) coding, which is used for example in the GSM standard. In CELP, the modelling is divided in two stages, a linear predictive stage that models the spectral envelope and code-book based model of the residual of the linear predictive model. CELP stands for Code Excited Linear Prediction and is a speech coding algorithm originally proposed by M.R. Schroeder and B.S. Atal in 1984. ... CELP stands for Code Excited Linear Prediction and is a speech coding algorithm described by the US Federal Standard FIPS 1016. ... Global System for Mobile communications (GSM: originally from Groupe Spécial Mobile) is the most popular standard for mobile phones in the world. ... Linear prediction is a mathematical operation where future values of a discrete-time signal are estimated as a linear function of previous samples. ...


In addition to the actual speech coding of the signal, it is often necessary to use channel coding for transmission, to avoid losses due to transmission errors. Usually, speech coding and channel coding methods have to be chosen in pairs, with the more important bits in the speech data stream protected by more robust channel coding, in order to get the best overall coding results. In digital telecommunications, channel coding is a pre-transmission mapping applied to a digital signal or data file, usually designed to make error-correction possible. ...


The Speex project is an attempt to create a free software speech coder, unencumbered by patent restrictions. Speex is a free software speech codec that may be used on VoIP applications and podcasts. ... This article is about free software as used in the sociopolitical free software movement; for non-free software distributed without charge, see freeware. ...


Major subfields:

Speech coding is the compression of speech (into a code) for transmission with speech codecs that use audio signal processing and speech processing techniques. ... Adaptive Multi Rate - WideBand or AMR-WB is a speech coding standard developed after the AMR using same technology like ACELP. The codec provides excellent speech quality due to wider speech bandwidth of 50 - 7000 Hz compared to narrowband speech codecs which in general are optimized for POTS wireline quality... W-CDMA (Wideband Code Division Multiple Access), a wideband spread-spectrum 3G mobile telecommuncation air interface that utilizes code division multiple access (or CDMA the general multiplexing scheme, not to be confused with CDMA the standard), is a 3G mobile communications standard allied with the GSM standard. ... VMR-WB is a source-controlled variable-rate multimode codec designed for robust encoding/decoding of wideband/narrowband speech. ... CDMA2000 is a hybrid 2. ... FNBDT is the U.S. Governments new standard for secure voice communication. ... SMV (Selectable Mode Vocoder) is speech coding standard used in CDMA-2000 networks. ... General Information Generically (as a multiplexing scheme), code division multiple access (CDMA) is any use of any form of spread spectrum by multiple transmitters to send to the same receiver on the same frequency channel at the same time without harmful interference. ... Full Rate or FR or GSM-FR was the first digital speech coding standard used in GSM digital mobile phone system. ... Half Rate or HR or GSM-HR is a speech encoding system for GSM developed in the early 1990s. ... Enhanced Full Rate or EFR or GSM-EFR is a speech coding standard that was developed in order to improve the quite poor quality of GSM-Full Rate (FR) codec. ... Adaptive Multi-Rate (AMR) is a Audio data compression scheme optimized for speech coding. ... Global System for Mobile communications (GSM: originally from Groupe Spécial Mobile) is the most popular standard for mobile phones in the world. ...

See also



 

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