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Showing papers on "Adaptive Multi-Rate audio codec published in 1988"


Proceedings ArticleDOI
Peter Vary1, Karl Hellwig1, Rudolf Dipl Ing Hofmann1, R.J. Sluyter, C. Galand, M. Rosso 
11 Apr 1988
TL;DR: The coding scheme which has been selected by the CEPT Groupe-Speciale-Mobile (GSM) as a result of formal subjective listening tests, is based on the regular-pulse excitation LPC technique (RPE-LPC) combined with long-term prediction (LTP).
Abstract: In 1991 a digital mobile radio system will be introduced in Europe The speech codec to be used as the standard is presented The coding scheme which has been selected by the CEPT Groupe-Speciale-Mobile (GSM) as a result of formal subjective listening tests, is based on the regular-pulse excitation LPC technique (RPE-LPC) combined with long-term prediction (LTP) The so-called RPE-LTP codec has a net bit rate of 13 kbit/s The algorithm and the experimental implementations based on different VLSI signal processors are described and demonstrated >

100 citations


Journal ArticleDOI
P. Mermelstein1
TL;DR: A tutorial discussion is provided of the adaptive differential PCM (pulse-code modulation) coding method recommended by the group, which covers the subjective performance tests performed, mode initialization and mode switching, data-speed multiplexing, and communication between narrowband and wideband terminals.
Abstract: CCITT Study Group XVIII recognized the need for a new international coding standard on high-quality audio to allow interconnection of diverse switching, transmission, and terminal equipment and organized an expert group in 1983 to recommend an appropriate coding technique. A tutorial discussion is provided of the adaptive differential PCM (pulse-code modulation) coding method recommended by the group. The discussion covers the subjective performance tests performed, mode initialization and mode switching, data-speed multiplexing, and communication between narrowband and wideband terminals. >

95 citations


Proceedings ArticleDOI
11 Apr 1988
TL;DR: Source coding and channel coding are embedded in a video coding algorithm for low bitrates (64 kb/s) to obtain a complete encoder which is resilient against channel errors.
Abstract: Source coding and channel coding are embedded in a video coding algorithm for low bitrates (64 kb/s) to obtain a complete encoder which is resilient against channel errors. The bits retained for the channel coder are spent for synchronization words, error correction, and update of erroneous parts in the image. The performance of the complete codec is evaluated by simulations in which channels with different characteristics are modeled. >

26 citations


Journal ArticleDOI
TL;DR: A 16 kbit/s speech codec with low complexity and low signal delay is presented which is a special version of the Regular-Pulse Excitation LPC approach (RPE-LPC).

16 citations


Proceedings ArticleDOI
S. Ono1, K. Ozawa1
11 Apr 1988
TL;DR: Subjective evaluation demonstrates that pitch-prediction multiphase speech coding can provide much more natural-sounding synthetic speech than single-pulse-excitation speech coding (as with a vocoder) or pitch-interpolation multipulse speech coding at 2.4-kb/s.
Abstract: 2.4-kb/s speech coding based on pitch-prediction multipulse speech coding is described. An adaptive segmentation procedure to effectively control the renewal rate for synthesis model parameters, a linear time-varying pitch-synthesis filter, a model for multipulse excitation during one pitch period, and a vector quantization for linear predictive coding parameters are introduced. Subjective evaluation demonstrates that pitch-prediction multiphase speech coding can provide much more natural-sounding synthetic speech than single-pulse-excitation speech coding (as with a vocoder) or pitch-interpolation multipulse speech coding at 2.4-kb/s. >

11 citations


Journal ArticleDOI
TL;DR: High-quality speech codec modules operating at 16 and 8 kb/s have been developed using an adaptive predictive coding with adaptive bit allocation (APC-AB) scheme, which achieves high speech quality, close to that of a 7-bit mu -law PCM.
Abstract: High-quality speech codec modules operating at 16 and 8 kb/s have been developed using an adaptive predictive coding with adaptive bit allocation (APC-AB) scheme. An optimized APC-AB algorithm is studied that reduces processing complexity while maintaining speech quality. The coding algorithm is implemented in two digital signal processors (DSPs). The DSP chips, a framing LSI circuit, a PCM codec, and some peripheral ICs are integrated in each of two compact packages, i.e. codec modules, operating at 16 or 8 kb/s. The codec module size is as small as 80 mm*50 mm*12 mm, and its typical power consumption is 500 mW using 2- mu m CMOS LSI technology. At 16 kb/s this APC-AB codec achieves high speech quality, close to that of a 7-bit mu -law PCM. The codec modules are expected to be used for various applications such as customer premises multiplexers for digital leased lines, digital mobile radio, and stored-and-forward-message systems (voice-mail systems). >

6 citations


01 Jan 1988
TL;DR: A comparison ofgregated and integrated communications networ, a comparison of local and wide area networks, and an overview of the contents of the thesis.
Abstract: CHAPTER 1 : INTRODUCTION 1.1 Background to the thesis 1.1.1 Segregated and integrated communications networ 1.1.2 Local and wide area networks 1.1.3 Problems associated with the addition of voice data LAN 1.1.4 The need for a special speech codec 1.2 Aims of the thesis 1.3 An overview of the thesis contents 1.4 Original contributions made by the thesis 1.5 Publications by the author related to the thesis CHAPTER 2 : THE NETWORK AND WORKSTATIONS 2.

5 citations


Journal ArticleDOI
TL;DR: The modification of a proprietary constraint length 7, 1/2-rate codec to operate with punctured coding at rates up to 7/8 is discussed and it is concluded that the method is very suitable to provide variable coding rates between 1/1 and 7/7 but suffers from increasingly severe practical disadvantages at coding rates above 7/ 8.
Abstract: It is attractive to be able to alter the coding rate, and hence the error correcting power, of a forward error correcting coder/decoder (codec) to provide a codec whose error correction capability can be matched to the requirements of the data communication system in which it is used. Punctured coding applied to convolutional encoding/Viterbi decoding offers a method of achieving this. Intelsat have specified punctured coding to provide flexibility in selecting either 1/2 or 3/4 rate coding in their Open Network Intermediate Data Rate system. This paper discusses the modification of a proprietary constraint length 7, 1/2-rate codec to operate with punctured coding at rates up to 7/8. Following a brief discussion of the requirements for variable rate coding in communication systems the practical implementation of a prototype codec and considerations relating to performance characterization are discussed. Experimental results at coding rates of 1/2, 2/3, 3/4, and 7/8 are given, together with simulation of performance at 15/16 coding rate to demonstrate the sensitivity of performance to path history length at higher coding rates. It is concluded that the method is very suitable to provide variable coding rates between 1/2 and 7/8 but suffers from increasingly severe practical disadvantages at coding rates above 7/8.

4 citations


Proceedings ArticleDOI
28 Nov 1988
TL;DR: The authors propose a method for the combined use of constant SNR (signal/noise ratio) and constant noise to determine optimum bit/rate control of the VRC.
Abstract: Discusses an implementation of an experimental variable bit rate codec (VRC) based on adaptive differential pulse code modulation with a multiquantizer (ADPCM-MQ) that processes speech signals at 16 kb/s to 48 kb/s The authors propose a method for the combined use of constant SNR (signal/noise ratio) and constant noise to determine optimum bit/rate control of the VRC This algorithm was implemented using six digital signal processor chips The average bit rate of the prototype codec for the telephone voice channel was about 23 kb/s, including supplementary information, and a quality better than that specified in G721 (32 kb/s ADPCM) was observed >

4 citations


01 Nov 1988
TL;DR: The performance of a jointly optimised sub-band codec (SBC), channel codec, and post processing speech enhancement system for binary phase shift keying (BPSK) transmissions over Rayleigh fading channels is presented.
Abstract: The performance of a jointly optimised sub-band codec (SBC), channel codec, and post processing speech enhancement system for binary phase shift keying (BPSK) transmissions over Rayleigh fading channels is presented. Forced up-dates of the SBC quantizer step sizes, and step-size leakage algorithms were examined and it wc2 found that for a channel BER of 10 the segmental SNR increased by approximately lOdB compared to the basic SBC. At high BERs forced up-dating was preferable. Systematic and non-systematic Reed Solomon (RS) coding and convolutional coding were employed. The RS codec was also embedded into the SBC, and speech enhancement based on template matching deployed. By using an, embedded RS codec with speech enhancement the gain in segmental SNR over the robust SBC for BERs in excess of was 7 dB. i The sub-band codec (SBC) produces Kear toll quality speech at 16 kbit/s and can be implemented using a DSP chip. It is therefore an appropriate codec for the transmission of speech signals in mobile radio environments, with the proviso that suitable channel coding and speech enhancement methods are employed.

3 citations


Journal ArticleDOI
R. B. Hanes1, P. M. Attkfins1
TL;DR: The 16 kbit/s speech codec developed by British Telecom Research Laboratories and selected as the UK candidate to the GSM Pan-European study on digital cellular land mobile radio offers several important features including low delay, low computational complexity and a good tolerance to transmission errors.

Proceedings ArticleDOI
11 Apr 1988
TL;DR: A new 8 kbps speech coding algorithm called TC-MQ (time domain compression ADPCM-MQ) is proposed, based on time scale modification and sub-band coding with the aid ofADPCM with a multiquantizer that was confirmed by computer simulations.
Abstract: A new 8 kbps speech coding algorithm called TC-MQ (time domain compression ADPCM-MQ) is proposed. It is based on time scale modification and sub-band coding with the aid of ADPCM with a multiquantizer. For time scale modification, the decimation/interpolation technique is introduced for the unvoiced period. Furthermore, in order to get computational accuracy of the pitch extraction for the voiced period, a method using the normalized autocovariance function is proposed. The algorithm was confirmed by computer simulations. The segmental SNR was about 13-16 dB for Japanese short sentences. A good mean opinion score value was also obtained by means of a subjective evaluation test. >

Proceedings ArticleDOI
11 May 1988
TL;DR: The design of a codec based on the (127,99) four-error-correcting BCH code is described, which has a short implementation cycle, requires a very small number of integrated circuit chips, and yields a codec that can operate up to about 5 Mb/s.
Abstract: The design of a codec based on the (127,99) four-error-correcting BCH code is described. This code was chosen as a compromise between overall performance and implementation complexity for a frequency-hopped spread-spectrum system operating under worst-case partial band noise jamming and worst-case multitone jamming. The codec is designed for implementation with application-specific integrated circuits. The approach has a short implementation cycle, requires a very small number of integrated circuit chips, and yields a codec that can operate up to about 5 Mb/s. >

Proceedings ArticleDOI
G.S. de Brito1
13 Jun 1988
TL;DR: The methodology used by the European Conference of Post and Telecommunication Administration to select and to adapt a low bit-rate speech coding scheme for the GSM system, a pan-European mobile telecommunications system, as well as the major characteristics of the selected algorithm are presented.
Abstract: The methodology used by the European Conference of Post and Telecommunication Administration (CEPT) to select and to adapt a low bit-rate speech coding scheme for the GSM system, a pan-European mobile telecommunications system, as well as the major characteristics of the selected algorithm, is presented. An overview of the radio transmission system is included. >

Proceedings ArticleDOI
07 Jun 1988
TL;DR: An experimental 64-kb/s video codec that transmits still and motion pictures in several coding modes and the computer simulation used to check the proposed coding, scheme's efficiency at 64 kb/s is detailed.
Abstract: The authors discuss an experimental 64-kb/s video codec that transmits still and motion pictures in several coding modes. In previous work (1986), they proposed a coding scheme, tailored to 384-kb/s transmission, that used motion-compensated prediction and hybrid quantization and a 64-kb/s video codec design concept having several coding modes. Here, they detail the computer simulation used to check the proposed coding, scheme's efficiency at 64 kb/s. They then describe the hardware architecture that implements the design concept. The performance of the video codec is then demonstrated in different coding modes. >

Proceedings ArticleDOI
24 Jun 1988
TL;DR: An overview is given of low bit rate (m*64 kb/s, m=1.6) video coding using a generalized video codec that uses a hybrid coding algorithm combining differential pulse-code modulation and transform coding or vector quantization in the spatial domain to code sequences of spatially and temporally subsampled images.
Abstract: An overview is given of low bit rate (m*64 kb/s, m=1.6) video coding using a generalized video codec. This codec uses a hybrid coding algorithm combining differential pulse-code modulation (DPCM) in the temporal domain and transform coding or vector quantization (VQ) in the spatial domain to code sequences of spatially and temporally subsampled images. The coder uses motion detection and estimation for more efficient prediction; the resulting motion vectors are also used in the decoder for motion-compensated interpolation of the images left out in the subsampling process. All the elements of the codec are discussed and some of the options available for each element are noted. >

Proceedings ArticleDOI
V. Lazzari1, M. Quacchia1, D. Sereno1, E. Turco
12 Jun 1988
TL;DR: The implementation of a real-time 16-kb/s speech codec is described, which has been tailored to the specific needs of digital mobile radio.
Abstract: The implementation of a real-time 16-kb/s speech codec is described, which has been tailored to the specific needs of digital mobile radio. The source coder belongs to the class of split-band adaptive predictive coders (SB-APC) and uses vector quantization (VQ) and dynamic bit-allocation. The transmitter and receiver ends have been implemented on a single board by using one TMS32020 digital signal processor for protection against errors, has been ensured by using a combination of a Golay code and a Hamming code as a forward-error-correcting code (FEC). The FEC algorithm has been implemented on an additional board with the same kind of digital signal processor as the speech codec. >

Proceedings ArticleDOI
28 Nov 1988
TL;DR: A method for the excitation sequence search is proposed in which the pulse amplitudes are computed one at a time, rather than simultaneously in block, by solving a linear system of reduced dimension.
Abstract: The regular-pulse-excitation (RPE) technique for speech coding gives good speech quality, but requires a heavy computational effort. A method for the excitation sequence search is proposed in which the pulse amplitudes are computed one at a time, rather than simultaneously in block, by solving a linear system of reduced dimension. The algorithm also provides an appreciable improvement in the performance, because the noise energy is minimized taking into account both quantization errors and the effect of pulses in the next block. The algorithm is implemented in a 8-kb/s codec, using vector quantization for the vocal tract all-pole model parameters. Computer simulations over a large database of male and female speech signals (about 80s of speech) gave good results in terms of segmental signal-to-noise ratio, even in the absence of a patch predictor. >

Proceedings ArticleDOI
Schdbinger1, Zehner, Matthiesen, Totzek, Hartl, Reimann, Tielert 
01 Jan 1988
TL;DR: A DPCM video codec for two-dimensional prediction with adaptive quantizer is presented and correct operation has been verified up to 26 MHz.