Akio Ando

A 22.2 Multichannel Sound System for Ultrahigh-Definition TV (UHDTV)

A 22.2 multichannel sound system was developed for an ultrahigh-definition TV (UHDTV) system. It consists of three layers of loudspeakers: an upper layer with nine channels, a middle layer with ten channels, and a lower layer with three channels and two channels for low-frequency effects (LFE). Subjective evaluations were performed comparing the impression of various spatial attributes using UHDTV contents with pictures in a large room. These evaluations demonstrate that viewers have better impressions of various spatial attributes in a wider listening area with the 22.2 multichannel sound system than with other sound systems. This paper describes the 22.2 multichannel system for UHDTV, and discusses the advantages of 22.2 multichannel sound. It also describes the sound recordings and productions by the 22.2 multichannel sound system.

High-Performance Condenser Microphone With Single-Crystalline Silicon Diaphragm and Backplate

This paper presents a high-performance silicon condenser microphone fabricated with a new process using single-crystalline silicon. This simple fabrication process, which requires only two photolithography steps and two wet-etching steps, is suitable for low-cost mass production. We designed the structure of a high-performance microphone and simulated it with an equivalent acoustic-circuit model. We then fabricated a prototype microphone based on this design, and experimental measurements on the prototype confirmed its excellent acoustic characteristics, such as a high sensitivity of -43.5 dB, a wide frequency range of 30 Hz to 20 kHz, and a high maximum sound pressure level (1 kHz at 1% THD) of 122 dBSPL. The measured equivalent noise (A-weighted) is 30.5 dBASPL. The measured frequency responses showed good agreement with those estimated from the simulation, indicating that the high controllability of the process enabled us to fabricate the prototype as we designed it. These results show that it is feasible to economically mass produce such high-performance microphones for purposes ranging from broadcasting to consumer use

A lightweight push-pull acoustic transducer composed of a pair of dielectric elastomer films

A lightweight push-pull acoustic transducer using dielectric elastomer films was proposed for use in advanced audio systems in homes. The push-pull structure consists of two dielectric elastomer films developed to serve as an electroactive polymer. The transducer utilizes the change in the surface area of the dielectric elastomer film, induced by an electric-field-induced change in the thickness, for sound generation. The resonance frequency of the transducer was derived from modeling the push-pull configuration to estimate the lower limit of the frequency range. Measurement results presented an advantage of push-pull driving in the suppression of harmonic distortion.

Semicylindrical acoustic transducer from a dielectric elastomer film with compliant electrodes

A semicylindrical acoustic transducer was constructed using a dielectric elastomer film with compliant electrodes that is an electroactive polymer composed of a polyurethane elastomer base and polyethylene dioxythiophene/polystyrene sulfonate electrodes. The use of this dielectric elastomer is advantageous because polyurethane is a common material that keeps its shape without any rigid frame. Because the dielectric elastomer films are essentially incompressible, electric-field-induced thickness changes are usually translated into much larger changes of the film area and side length. Here it is proposed that this change in side length can be utilized for sound generation when the film is bent into a semicylindrical shape. Accordingly, a semicylindrical acoustic transducer was fabricated using a film of thickness of 300 μm and its acoustic characteristics were investigated. The transducer can be operated at low applied voltages by reducing the film thickness, as long as the film is thick enough to generate sufficient force to overcome sound radiation impedance. The second harmonic distortion of the transducer was also investigated as a function of the ratio of the direct current bias voltage to the alternating current audio signal amplitude.

A Narrow-Angle Directional Microphone With Suppressed Rear Sensitivity

A novel microphone that enables rear sensitivity to be significantly suppressed has been developed to improve open-air recording quality. Its assembly comprises a line microphone capsule and a second-order pressure gradient directional microphone. In conventional line microphones, residual rear sensitivity causes an influx of unexpected noise, especially at lower frequencies. Our microphone successfully suppresses rear sensitivity by more than 10 dB compared to conventional line microphones in the frequency range below 1 kHz in which major outdoor noise often occurs. Furthermore, it needs no complicated signal processing circuit and can be driven by a normal 48 V phantom power supply. Finally, our microphone was tested in on-the-spot broadcasts. Its rear sensitivity suppression proved to be effective for practical use, and its sound quality was found to be sufficient for use in TV programs. This paper describes the fundamental principle of the microphones rear sensitivity suppression, the measurement results of its acoustic characteristics and field-test results obtained with it in on-the-spot broadcasts.

Separation of Sound Sources Propagated in the Same Direction

This paper describes a method for separating a target sound from other noise arriving in a single direction when the target cannot, therefore, be separated by directivity control. Microphones are arranged in a line toward the sources to form null sensitivity points at given distances from the microphones. The null points exclude non-target sound sources on the basis of weighting coefficients for microphone outputs determined by blind source separation. The separation problem is thereby simplified to instantaneous separation by adjustment of the time-delays for microphone outputs. The system uses a direct (i.e. non-iterative) algorithm for blind separation based on second-order statistics, assuming that all sources are non-stationary signals. Simulations show that the 2-microphone system can separate a target sound with separability of more than 40 dB for the 2-source problem, and 25 dB for the 3-source problem when the other sources are adjacent.

Progressive early decision of speech recognition results by comparing most likely word sequences

The most likely word sequence determined at the end of an utterance constitutes an optimal recognition result in continuous speech recognition for the entire utterance. However, depending on the application, the delay from the utterance to the determination of the recognition result may pose a practical problem, and progressive early decision of recognition results during an utterance becomes necessary. Although in the case of a one-pass search algorithm, progressive early decision of the recognition result by detecting past sole paths during search is possible, an effective early decision scheme is not available for the case of multiple passes. Thus, a scheme for progressive early decision of recognition results by successively comparing the most likely word sequences during an utterance with the past most likely word sequences is proposed and is applied to a one-pass decoder and a two-pass decoder. The proposed scheme attempts to shorten the delays associated with word decisions while limiting the degradation of the recognition rate by controlling the word decision margin and the interval for obtaining the most likely word sequences. In speech recognition experiments of broadcast news, the proposed scheme achieved an average word decision delay equal to that of the past sole path detection method in a one-pass decoder without significantly degrading the word recognition accuracy, and was able to progressively decide recognition results with an average word decision delay time of about 0.5 second in a two-pass decoder.

Comparison of Tied-Mixture and State-Clustered HMMs with Respect to Recognition Performance and Training Method

Tied-mixture HMMs have been proposed as the acoustic model for large-vocabulary continuous speech recognition and have yielded promising results. They share base-distribution and provide more flexibility in choosing the degree of tying than state-clustered HMMs. However, it is unclear which acoustic models to superior to the other under the same training data. Moreover, LBG algorithm and EM algorithm, which are the usual training methods for HMMs, have not been compared. Therefore in this paper, the recognition performance of the respective HMMs and the respective training methods are compared under the same condition. It was found that the number of parameters and the word error rate for both HMMs are equivalent when the number of codebooks is sufficiently large. It was also found that training method using the LBG algorithm achieves a 90% reduction in training time compared to training method using the EM algorithm, without degradation of recognition accuracy.

Introduction to the Issue on Spatial Audio

The papers in this special issue focus on spatial audio technology and applications for its use. Spatial audio is an area that has gained in popularity in the recent years. Audio reproduction setups have evolved from the traditional two-channel stereophonic setup towards multi-channel loudspeaker setups. Advances in acoustic signal processing even made it possible to create surround sound listening experiences as well as attempts to create binaural real life listening experiences using traditional stereo speakers and headphones. Finally, there has been an increased interest in creating different sound zones in the same acoustic space. At the same time, the computational capacity provided by mobile audio playback devices has increased significantly. These developments enable new possibilities for advanced audio signal processing, such that in the future we can record, transmit and reproduce spatial audio in ways that have not been possible before. In addition, there have been fundamental advances in our understanding of 3D audio.

Control of frame loudspeaker array by minimizing fluctuations of frequency response and synthesized wave front

In sound reproduction with accompanying pictures, the localization of sound on or behind the flat-panel display is problematic because a loudspeaker cannot be placed in such directions. To gain a stable localization, the use of a loudspeaker array set on the frame of the display may be a solution. In general, the loudspeaker array can generate the spherical wave surface from the virtual sound source located at the center of the wave surface. However, the frequency response and the shape of the wave surface reproduced by the array sometimes deteriorate, particularly when setting the virtual sound source at a certain distance from the back of the display In this study, new parameters are introduced to scale the deterioration of the frequency response and the shape of the wave surface. A new method is then proposed, and the weighting coefficients over the array elements are generated by minimizing these parameters. The experimental result showed that the frequency response approached the flat response, altho...