Takashi Araseki

Echo Canceler with Two Echo Path Models

An adaptive echo canceler with two echo path models is proposed to overcome the false adaptation problem for double-talking. The echo canceler possesses two separate echo path models (EPMs), one (background EPM) for adaptively identifying echo path transfer characteristics and the other (foreground EPM) for synthesizing an echo replica to cancel out echo. The parameter values of the foreground EPM are refreshed by those of the background EPM, according to a transfer control logic, when the logic determines that the background EPM is giving a better approximation of echo path transfer characteristics than the foreground EPM. Completely digital hardware implementation is described. Using the hardware, it is shown that virtually complete double-talking protection is actually realizable by the new method.

Echo canceller for attenuating acoustic echo signals on a frequency divisional manner

For use in combination with a loudspeaker and at least one microphone, for example, by attendants in an auditorium, an echo cancelling circuit comprises a self-adaptive echo canceller responsive to a lower frequency component, such as below 1.7 kHz, of a receive-in signal for self-adatively cancelling a corresponding component of a reverberation signal included in a send-in signal during each interval during which an audio signal reaches the circuit from a remote party. For a higher frequency reverberation signal component, an echo suppressor or a voice switch may reduce a weaker one of two signals which are either the higher frequency send-in and receive-in signal components or a combination of a reverberation component cancelled signal with the higher frequency send-in signal component and the whole receive-in signal. Alternatively, a less expensive echo canceller non-adaptively cancels a part of the reverberation signal in response to the receive-in signal. The lower frequency component of the partially reverberation cancelled signal is used by the self-adaptive echo canceller as the lower frequency send-in signal component. An acoustic output may once be reproduced by the loudspeaker in response to the receive-in signal and then converted to an electric signal for supply to the echo cancelling circuit.

A 32 kb/s toll quality ADPCM codec using a single chip signal processor

An ADPCM codec, that can provide toll quality speech at a 32 kb/s transmission rate, has been implemented on a single chip signal processor. Maximum effort has been paid to design a robust adaptation scheme for a quantizer and a predictor to withstand transmission bit errors. The codec employs a simplified robust quantizer and also employs a new backward adaptive predictor. The decoder, including the new adaptive predictor, has a structure having fixed poles and adaptive zeros, attaining both high prediction capability and robustness. The performance of a developed codec, which has analog interface capability through a PCM codec chip, satisfies the standard 64 kb/s PCM performance specification in CCITT recommendation G.712.

A Study on Pulse Search Algorithms for Multipulse Excited Speech Coder Realization

This paper describes and compares several kinds of pulse search methods for multipulse excited speech coder realization. These pulse search methods are derived from minimization criterion for errorpower between original speech and synthetic speech, but their performances and required computation amounts are different. Objective and subjective evaluations are carried out to compare the performances. Further, the relation between speech quality and pulse search method complexity is described. Based on these results, pulse search methods suitable for realizing a high-quality multipulse coder using current VLSI technology are discussed.

Adaptive speech signal detector

An adaptive speech signal detector for use in a 4-wire telephone channel performs an adaptive threshold value setting operation depending on the channel noise level on a transmitter-side channel to detect a speech signal present at the transmitter. The adaptive operation of the speed signal detector is inhibited, however, if the signal level at the related receiver-side channel becomes higher than a preset value. This permits the use of the adaptive speech signal detector with DSI (digital speech interpolation) systems without malfunction due to the operation of an echo suppressor.

Digital signal processing for video

Applications of the digital signal processing of video signals in broadcasting, communication, and consumer electronics are reviewed. These include: digital encoding systems, digital video effect equipment, and the television standards converter for broadcasting; videoconferencing and video telephone equipment; and TV receivers, including those for extended definition and high-definition television (EDTV and HDTV). Performance requirements for video signal processing (VSP) are discussed, and an example of a video signal processor comprising a parallel processor system composed of multiple VSP modules is examined. Future trends in VSP are predicted.<<ETX>>

High quality multi-pulse speech coder with pitch prediction

This paper describes multi-pulse speech coding with pitch prediction. Two realization approaches to the coding, which differ in regard to pitch predictor, are described. Coding performances for the two approaches are evaluated objectively and subjectively. The introduction of pitch prediction to a multipulse coder shows 2-3 dB SNR (Signal to Noise Ratio) improvement, and perceptually annoying noise is reduced over a conventional coder without pitch prediction, at bit rates above 8 kbps.

Linear transformation for low bit rate speech coding

This paper discusses the possibility of an input-dependent orthogonal transform for low bit rate speech coding. An orthogonal transform, based on an autoregressive model of short-time input speech, which is applicable to low bit rate coding, is presented. Its performance is evaluated in comparison with that for conventional transforms, KLT and DCT. The experiments confirm that the input-dependent orthogonal transform improves the average distortion versus average information rate performance over that for the input-independent transform DCT.

A multi-pulse LPC speech codec using digital signal processors

A full-duplex 16 or 9.6 k bits/sec speech codec based on the multi-pulse excited LPC coding technique has been developed. Cross-correlation analysis was used for the pulse search method. The codec was implemented by using five NEC 7720 Digital Signal Processor LSI chips ; four DSPs for the coder and one DSP for the decoder. A general purpose microprocessor was also used for system control. Reconstructed speech was quite natural and almost undiscernible from original speech at 16 k bits/sec and still quite good at 9.6 k bits/sec. Signal to noise power ratio for a sine wave signal was 30 dB at 16 k bits/sec.

An adaptive echo canceller using digital signal processor LSI chips

This paper discusses a compact adaptive echo canceller with digital signal processor (SP) chips on the basis of a cascadable structure. By using this cascadable structure, tap number expansibility and program modification for satisfying various requirements can be easily obtained. Two kinds of echo cancellers were implemented. One for teleconference use, which has 600 taps with 12 SPs, has accomplished more than 20 dB echo cancellation in a conference room. It provides subjective high performance. The other, for telephone line use, with up to 300 taps, attains more than 30 dB echo return loss enhancement (ERLE). Each of these echo cancellers was implemented on a single board.