Zobrazeno 1 - 10
of 48
pro vyhledávání: '"Shigeyuki Unagami"'
Publikováno v:
IEEE Communications Magazine. 28:49-55
Practical echo cancellation techniques, in particular, those used in telecommunications, are reviewed. The various situations in which echoes are generated are examined. Echo path modeling techniques and adaptive algorithms for coefficient control ar
Publikováno v:
ICASSP
This paper describes a newly developed CMOS LSI DSP and its application to a 32 Kbps ADPCM CODEC and a 4,800 bps data MODEM. The paper first analizes the required memory capacities of ROM and RAM as a function of arithmatic operation capability of DS
Publikováno v:
ICASSP
Recently much intensive research of 16kbps Speech coding algorithm has been conducted aiming to reduce the transmission bit rate and yet provides high speech quality. Adaptive predictive coding with adaptive bit allocation (APC-AB)[1] is considered t
Publikováno v:
ICASSP
This paper describes an implementation of a new 16 kbps speech codec using commercially available DSPs and its performance. The coding algorithm chosen here is ADPCM with Multi-Quantizer (ADPCM-MQ) which selects the optimum ADPCM coder frame by frame
Publikováno v:
ICASSP
Adaptive differential PCM (ADPCM) is an effective coding scheme to simplify the hardware and shorten the processing delay to realize a high-efficiency speech codec. The ADPCM with Multi-Quantizer (ADPCM-MQ) coding has been proposed as one of the high
Publikováno v:
ICASSP
The authors introduce a novel approach to narrow- and medium-band speech coding that can dynamically balance the transmission rate between the excitation and the spectral parameters. The coding algorithm, called multimode coding, operates several cod
Publikováno v:
IEEE Global Telecommunications Conference and Exhibition. Communications for the Information Age.
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 t
Publikováno v:
ICASSP
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/inte
Publikováno v:
MILCOM 88, 21st Century Military Communications - What's Possible?'. Conference record. Military Communications Conference.
The authors propose a 4.8-kbps voice-coding algorithm using pitch-synchronous discrete Fourier transform (DFT). This algorithm combines time-scale compression with pitch-synchronous DFT spectrum coding. It also discards the time-compressed signal-pha
Publikováno v:
ICASSP
A multimode source/channel coder which can dynamically control the balance of source and channel coding according to the channel quality is introduced. As an example of such a system, a 4.8-kb/s code excited linear predictive coder with three coding