Kazuo Ochiai

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.

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.

Topics on speech waveform coding activities in Japan

Current activities on speech waveform coding technique developments in Japan are reviewed. The following topics of the activities on ADPCM and ADM are discussed. (1) An ADPCM code sequence optimization method by Fushikida for an ADPCM with an adaptive quantization and fixed predictor. The method uses a technique reminiscent of dynamic programming and optimizes code sequence. It is shown that about 3 dB SNR improvement is obtainable for speech at 32 kbps (4 bits/sample). (2) Another ADPCM technique by Araseki and Ochiai. This technique is suitable for use at the bit rate less than 32 kbps. Two adaptive predictors connected in a cascaded form are used. One of the predictors has an ordinary small number of prediction taps. However, the other predictor possesses a considerably larger number of predictor taps and is used to predict the speech waveform which is one pitch period apart. (3) Performance improvement techniques and LSI implementation by Tomozawa and Niwa for a discretely variable slope 32 kbps ADM (...