AES 東京コンベンション 2009: 技術発表詳細

A1 - 1 OFDM技術を用いたデジタル（多チャンネル）ワイヤレスマイクシステムの開発

Nowadays, regulation of B type and A type wireless microphone was standardized, most of wireless mike are in the direction of digital format type.Because the demand for good and high sound quality efficient in frequency-use multichannel wireless mike is large in the hall and the theater and other various places, so the development and supplement for new digital microphone system and also reasonable price product is expected to be introduced.

Authors have already reported the principle and the feasibility of synchronous averaging method in the frequency domain which can acquire good response characteristics with a high S/N ratio in an asynchronous sound system between the sides of sound source and measurement. This report proposes a TSP (Time Stretched Pulse) method for practical use, in that a precise impulse response could be obtained without distortion even if the clock frequency of measured sound differs from that of the sound source, and also introduces some applications of this method.

A1 - 3 External latency-optimized soundcard synchronization for applications in wide-area networks

Soundcards can be clocked with a set of fixed sample frequencies. The reference for these frequencies is generated by the card's internal quartz, which feeds an attached PLL (phase locked loop). The PLL output's quare signal refers to the term wordclock, which triggers the respective conversion processes. Two soundcards, however, inherently suffer from a slight wordclock drift. This can be eliminated by feeding one card's external wordclock input with the other card's clock. Since it is due to numerous reasons not possible to transmit the clock in wide area networks (WAN), exact synchronization requires a direct cable connection. Hence, this paper investigates a new solution, which provides precise remote soundcard synchronization via a novel frequency comparison and adjustment method.

A1 - 4 Fast Measurement of Motor and Suspension Nonlinearities in Loudspeaker Manufacturing

Automatic testing at the end of the assembling line is performed to find loudspeaker defects. Distortion measurements can reveal symptoms of driver nonlinearities but they are difficult to interpret and it is hardly possible to distinguish nonlinearities of the electro-dynamical motor from nonlinearities in the mechanical suspension. A fast measurement procedure is presented which assesses the large signal performance of the driver and quantifies the dominant loudspeaker nonlinearities in terms of easy to interpret single-valued parameters like the voice coil offset. These parameters can even be determined when the driver is already mounted in a vented-box system. The direct correspondence to production process parameters allow an optimized controlling of the production process.

P1 — 室内音響と聴取実験

P1 - 1 重回帰分析を用いた聴感上の好ましさと音場印象との関係についての研究

Preference of spatial impression shall be different individually, and it is also hard to explain her/his preference by herself/himself. In this paper, we aim at revealing which auditory impressions dominate a subject's preference in the different venues: an auditorium and a concert hall. In the framework of multiple regression analysis, preference as a response variable can be explained by the linear combination of some auditory impressions as explanatory variables, where both preference and auditory impression are obtained by listening tests for chorus music.

P1 - 2 マルチチャンネル再生システムにおける音の大きさの印象制御

In a sound system for large venues such as drama theaters, extending source imagery might enhance the intention of the sound designer integrating with other various controls on spatial impression. With the existing knowledge on the physical characters associated with perceived width, we created an algorithm extending source imagery horizontally in the multichannel reproduction system. The proposed method controls the degree of variation in perceived auditory image width by adding two extra sound sources with the extended angles from the main source. With the optimized angles and gains of the extra sources, the subjective evaluation showed that the proposal method intuitively and effectively extends the perceived width in the context of the multichannel sound system for a large venue.

P1 - 3 放射状に配置したスピーカーを用いた、ピアノ音再生の品質向上

Previously we reported spatial image improvements of Piano sound reproduction using multiple loudspeakers. In this second report, we describe the improvement in the quality of piano sound reproductions attained by increasing the number of multichannel loudspeakers, especially noticeable at higher sound pitches.

P1 - 4 Enhanced presence of electronic orchestral music (Part I): A comparison of loudspeakers and their perceptual effect

When an electronic instrument is reproduced in a space, its sound quality is often dependent on the character of the loudspeaker. In particular, for a composition with multiple tonal variation, the reproduction sound system should represent not only tonal, but also spatial identity, from which listeners can perceive fidelity of the sound field. This paper is the result of a series of investigations into an ideal reproduction system for an ensemble of electronic instruments, which gives listeners more natural and realistic impression that an equivalent acoustical instrument would generate. As the first step, the sound fields associated with six conventional loudspeakers were subjectively compared. The analysis of the subjective response showed that the reproduced sound fields of the electronic ensemble could be located in a two-dimensional perceptual space, which indicates the semantic relationship between attributes and the stimuli. The subsequent multiple regression results showed that the total impression of music was accounted for by two attributes: Distance and Brightness.

In this report, we will describe experiments using headphones to evaluate the sensitivity to the horizontal location of the sound sources. We will also report on the effects of room reflection as well as the effects of degradation of sound quality when using audio encoding. Unexpectedly, the sensitivity did not consistently decrease when the room reflection increased; or when lower bitrates were used.

This paper proposes a sound quality customization method based on an Paired Comparison interactive genetic algorithm (PC-IGA). A user who is not familiar with music and/or audio devices finds it difficult to customize the reproduction parameters to suit his/her preferences. Our solution is a system that customizes the reproduction parameters for headphone use by an intuitive user input panel; the IGA analyzes the user's inputs and generates the appropriate parameters. The system makes it easy, even for novices, to obtain the preferred sound quality. In this paper, the proposed system simplifies the evaluation by using pair comparison. Results of a subjective assessment experiment, demonstrate that the preferred sound quality can be obtained through this system.

Considering the ideal music practice room, are there any tendency beyond the personal preference regarding the size and the shape of the room, the material of the walls, and the reverberation sound? To study about these, author had a questionnaire to ask the requests for music practice rooms to the students who use them usually. From the result of the questionnaire, it became clear that many answerers require `appropriate reverberation' despite differences of music instruments or years of experience. The author tried to survey `appropriate reverberation' and size of the practice room and had experiments focused on relationship between direct sound and early reflections of room. The aim of the experiment was to study relationship between direct sound and single early reflection in a music practice room. As a result, there is no significant deference between each musical instrument, but there is a tendency by grouping the musical instrument. To confirm this result, the rooms were simulated which applied the preferred delay time into actual room size considering mean free path. Two types of room were simulated, one has all reflections and the other has single reflection. From the interview of the player who compares the simulated sound of these rooms, they preferred the room which has all reflections. It is assumed that the following reflections are important for natural impression.

P1 - 8 モード合成法を用いた矩形室内音場予測に関する一考察

The sound field in enclosed space of rectangular shape can be effectively described by modal model. In this model, physically reasonable damping term of each mode must be taken into account. This report introduces the outline of derivation process of modal summation formula and several practical methods for introducing damping term. The good agreement of the results obtained in numerical simulations and the experiments indicate the validity of the proposed method.

P1 - 9 ゲームサウンド制作スタジオにおけるミキシング環境に関する調査

Acoustical properties and physical conditions of mixing environments in Japanese video-game sound production studios are described. To research the interchangeabilities and the differences in the mixing environments among Japanese game production studios, acoustical measurements and field surveys of the monitoring conditions were carried out at different mixing studios. We analyzed the measurements and surveys to give the following values for each studio to show the existing conditions and examine the interchangeabilities among mixing environments in Japanese game productions; 1) monitoring responses, 2) reverberation times, 3) averaged absorption coefficients of rooms, 4) playback levels, 5) reference levels of the monitoring chains, 6) physical dimensions of rooms.

A2 — 電気音響変換器

A2 - 1 Distributed Mechanical Parameters Describing Vibration and Sound Radiation of Loudspeaker Drive Units

The mechanical vibration of loudspeaker drive units is described by a set of linear transfer functions and geometrical data which are measured at selected points on the surface of the radiator by using a scanning technique. The mechanical vibration can be summarized to a new quantity called accumulated acceleration level (AAL) which is comparable with the sound pressure level (SPL) if no acoustical cancellation occurs. These derived parameters are the basis for modal analysis and make the relationship between mechanical vibration and sound pressure output more transparent. Finally, the usage of the distributed parameters within finite and boundary element analyses is addressed and conclusions for the loudspeaker design process are made.

This paper describes the development of a super-wide-range microphone with cardioid directivity, which covers the frequency range up to 100 kHz. The proposed microphone adopted a two capsule configuration composed of an omnidirectional capsule covering up to 100 kHz and a bidirectional capsule designed to fit the characteristics of the omnidirectional one. The bidirectional capsule was designed to precisely fit the sensitivity level and the intrinsic noise level of the omnidirectional capsule . The measurement results show that the proposed microphone achieves wide frequency range up to 100 kHz, as well as low noise characteristics and excellent cardioid directivity.

A2 - 3 Development of Narrow-Angle Directional Microphones with Suppressed Rear Sensitivity

Novel microphone structure to enable rear sensitivity to be significantly suppressed is proposed. It improves open-air recording quality by comprising a line microphone capsule and a second order pressure gradient directional microphone. We made two types of prototype microphones whose length are 29 cm and 15 cm. They successfully suppress their rear sensitivity by more than 10 dB compared to conventional line microphones in the frequency range in which major outdoor noise often occurs. Its additional advantage is that they don't need a complicated signal processing circuit and availability of a normal 48 V phantom power supply. This paper describes the fundamental principle of the microphone's rear sensitivity suppression and the measurement results of their acoustic characteristics.

A2 - 4 Field trial of Narrow-Angle Directional Microphones with Suppressed Rear Sensitivity in Outside Broadcasts

NHK began to use the newly-developed narrow-angle directional microphones with suppressed rear sensitivity which were developed for the outside broadcasting. The microphones have a narrow directivity specializing in the front sensitivity. We have tested these microphones in the several TV programs, and discussed their characteristics and practicability. In this paper, we will report an obtained new method of positioning the microphones to pick up objective sounds in a noisy field, comparing with the common one.

A3 — 室内音響

A3 - 1 モード合成法と1次反射音を活用したサラウンドスタジオ設計～NHK CP604スタジオの設計より～

In designing a fine multichannel monitoring environment, `low frequency control' and `wide listening area with natural playback sounds' are the important keys. In this paper, three practical design strategies are introduced to realize these keys. The first strategy is a calculation technique of modal analysis using `modal summation method.' It can predict low frequency properties of a rectangular room with less modeling/computing load. The second strategy is an active reflection control. It is a technique to improve low frequency responses by adding reflection sounds. The third strategy is a DFR (Decorrelated First Reflections). It is a design method of time response using early reflections actively. These strategies are introduced through a real design project, multichannel mixing room of NHK's CP604 Studio. Both calculated and measured data for the project shows good responses to demonstrate usefulness of the design methods proposed here.

A3 - 2 NHK HD520 ポストプロダクションスタジオの室内音響とシステム設計

The loudspeaker setup was compliant with ITU-R BS775-1. Excellent architecture acoustics characteristic. HD520 employed cutting-edge system machine. HD520 was designed for the high quality works. I do presentation of HD520 about that acoustics characteristic and the system construction

A3 - 3 径の異なる円柱群を多層状にランダム配置した拡散反射構造が室内音場に及ぼす影響の基礎的検討

A basic study on a multi-layered diffuser composed with different sized cylinders in random distribution for a diffused reflection mechanism in small rooms is examined. This paper shows the result of the study on the effect upon room acoustic characteristics, especially focused on the eigenmodes at low frequency, the characteristics of diffused reflection, and the sound absorption characteristics of the diffuser. The full and 1/5 scale model experiments and the simulation using the FDTD (Finite Difference method in Time Domain) are used for this investigation.

A3 - 4 固有モード計算による中小空間の低域特性に関する予測・評価手法 -N値と聴感確認による室寸法比の評価の提案-

In acoustic design for small and medium-sized rooms, it is important to control low frequencies. But conventional evaluations methods are not suitable to make objective evaluation to find the only correct answer. In this study, a new evaluation method is proposed which has conventional mode density and effective mode density, represented by an index N, incorporated for mode evaluations. Use of the index N enables SPL deviation and degeneracy to be evaluated with simple numeric values. Moreover listening examination can be also carried out by auralization of room response.

P2 — 測定と信号処理

P2 - 1 音声の柔軟な操作を目的としたVocoderに基づくモーフィングツールの紹介

A flexible framework for voice manipulations based on a high-quality vocoder, TANDEM-STRAIGHT and temporally variable multi-aspect morphing was introduced. This framework was made accessible by introducing graphical user interfaces with supporting tools. The tools provide means to modify existing speech materials interactively and intuitively by modifying parameters directly or interpolating between examples on arbitrarily designed morphing trajectories. Demonstrations of flexible voice manipulations suitable for game applications also will be introduced.

Improve the articulation (clarity) and discrimination performances of audio equipment or audio signal easily by simple circuit constitution to provide a high definition audio signal and audio equipment inexpensively. The way of this solution, that it only using series connected two kinds harmonics generator with asymmetrical difference operation without harmful distortion. Small level harmonics have same rising time to original input signals what hearing white noise or impulse on aural system where asymmetrical difference operation will reduce these components with phasing harmonics generation by Haas effect same time together. Furthermore, these components are reducing harmful distortion components by slow slope filtering with summing to input signals. Consequently, the sounds articulation and discrimination performance of audio signals or audio equipments are improved.

P2 - 3 A new upmixing algorithm based on frequency domain independent component analysis

n this paper, we propose a new noble upmixing algorithm that manipulates front and rear components of the five-channel application from the conventional stereo (two-channel) contents. The proposed method is based on a statistical analysis called Frequency Domain Independent Component Analysis (FDICA). As its name implies, FDICA extracts independent (not necessarily orthogonal) components from a stereo signal, which differs from conventional methods such as Principal Component Analysis (PCA). We created an extended surround imagery by placing a relative independent source to rear channels. The subsequent subjective evaluation showed that the proposed method was preferred to some conventional processors due to the similarity in spatial attributes of the its original multichannel mix.

P2 - 4 Evaluation of Synchronized Significant Multi-bits Acoustic Steganography Method

Recently, steganography using multiple media content has been proposed for enhancing information security along with encryption. Research areas range from hidden capacity enlargement, to robustness enhancement of stego data towards attacks and so on. In this paper, model and algorithm of real-time steganography scheme are proposed. This method is implemented to embed secret acoustic data stream which is recorded as synchronously as it is embedded into another acoustic cover data stream. In addition, embedding positions among [1st, 8th] bit in cover data with sampling size of 16-bit can be arbitrarily specified. Experimental results demonstrate that the secret bit stream can be interspersed into significant bit locations in cover without drawing suspicion even though some certain performance degradation is caused.

P2 - 5 聴覚フィルタを用いた音場評価に関する基礎的検討 -動的圧縮型ガンマチャープフィルタの適用-

We attempt to apply various auditory models to evaluation of room acoustics. In this report, we examine the decay curve which is calculated from the impulse response filtered by using dynamic compressive gammachirp filter. Reverberation time which is observed from the obtained decay curve could provide early decay time which is observed by traditional method and have been said to be correlate closely with subjective reverberation time. We apply dynamic compressive gammachirp filter to the evaluation of the impulse responses which are measured at many points in the hall. The results show that the proposed method might evaluate sound field taking into consideration the auditory property of human.

This paper describes a new method for representing a head-related transfer function (HRTF), which is an acoustic transfer function between a sound source and the ear canal entrance. An HRTF is defined as a function on time and space. The spatio-temporal frequency characteristics can be visualized and analyzed by showing the spectrum computed by two-dimensional Fourier transform on time and space. In our experiments, we investigated the basic property of the spatio-temporal frequency characteristic and the difference between all data for the HRTFs obtained by numerical analysis and actual measurements. The reverberation and HRTF individuality were also examined. From the results, the characteristics were mostly concentrated in a specific band frequency, and there were also differences among databases.

Quantitative evaluation measure was proposed for di®used property of distributed mode loudspeaker(DML). Full width at half maximum angle was introduced to evaluate cross-correlation function, Gontcharov had proposed Graphically. We showed DML radiate spatially less coherently than conventional loudspeaker does. We could separate DMLs from other pistonic radiators with proposed measure.

P2 - 8 A Proposed Method of Characterizing Audio Distortion Induced by Power Supply Ripple in Audio Amplifier

Digital input Class-D amplifiers will be the predominant amplifier technology enabling consumer audio systems in the future. The traditional Power Supply Rejection Ratio (PSRR) measurement method cancels supply ripple in Bridge-Tied-Load (BTL) amplifiers, thus is unable to measure audio distortion induced. The proposed method employs innovative measurement of both Intermodulation Distortion (IMD) and Total Harmonic Distortion Plus Noise (THD+N) to more accurately represent audio quality. A new term known as Power Supply Ripple Distortion Factor (PSRDF) is introduced as a figure of merit for audio quality. Examples of how the proposed method effectively characterizes different levels of distortion induced by power supply ripple in closed-loop and open-loop BTL Class-D amplifiers are also presented. The proposed method is also applicable in characterizing all audio power amplifiers.

A4 — ディジタルオーディオ機器とメディア

This paper presents a real-time software of sampling jitter measurement. The software is programed on the MATLAB environment using the data acquisition toolbox to get analog-to-digital converted audio signals in real-time. It can display a jitter waveform and its spectrum. It can also display an amplitude fluctuation waveform and its spectrum, which is often observed as a result of conversion process in a DAC or an ADC. The beneits of the real-time jitter measurement are discussed in terms of the factors truly or not truly affecting jitter and the professional use of the digital audio equipments.

A4 - 2 ＭＰＥＧ－４ＡＬＳによるライブハウスにおける演奏音のロスレス伝送

鎌本優（日本電信電話株式会社，ＮＴＴコミュニケーション科学基礎研究所）, 原田登（日本電信電話株式会社，ＮＴＴコミュニケーション科学基礎研究所）, 守谷健弘（日本電信電話株式会社，ＮＴＴコミュニケーション科学基礎研究所）, 金順暎（日本電信電話株式会社，ＮＴＴ未来ねっと研究所）, 藤井竜也（日本電信電話株式会社，ＮＴＴ未来ねっと研究所）

This paper reports an experimental result of lossless transmission of sound data. Sixteen-channel acoustic signals compressed losslessly by MPEG-4 ALS were transmitted from the live house to the cafeteria via IP network in order to provide high-quality music. At the cafeteria, received sound data were decoded perfectly and appropriately remixed for adjusting to the environment of the location. Combination of high-definition video and audio data enables more fans to enjoy the performance of the musicians not only in live house but also in any other places at the same time. The experimental result shows that sound data are successfully compressed to around 38% of its original size losslessly, and the ALS can save the data rate about 11 Mbps.

Audio and broadcast archive warehouses require scalable storage technology which is expandable, robust and with fast enough access for synchronous read, write and transmit operations. The archive technology chosen by an organization is likely to be a multi-tiered system comprised of different storage mediums, each with their own strengths. This paper investigates the storage options for broadcast and audio archiving, and details opportunities for the future.

Coincident microphone setups are well known for their unique flexibility in terms of stereophonic imaging. How-ever, their reputation in terms of their spatial reproduction is bad. This prejudice was produced both by non-optimal setups having insufficient signal separation and non-optimal microphones of the 60s and 70s. These shortcomings nowadays can well be avoided. Coincident setups and, in particular, M/S setups for Stereo and Surround exist that are popular for their outstanding practicability. When care is taken regarding parameters like directional imaging and diffuse field correlation, a coincident setup can compete with spaced setups also regarding the spatial reproduc-tion. A particular look is taken on the Double M/S technique for Stereo and Surround.

A5 — 心理音響・知覚と聴取実験

Two experiments and two analyses were performed in order to evaluate sounds of snare drums quantitatively as a preparative study for quantitative evaluation of drum sets. The experiments were performed with a set of stimuli generated by 10 snare drums. ``Method of Paired Comparison'' and ``Rating Scale Method'' were used to the experiments. ``Multi Dimensional Scaling(MDS)'' and ``Multiple Regression Analysis'' were used in the analyses. From these, connections between attribute ratings for each snare drum's sounds and the two-dimensional stimulus space based on psychological distance were obtained. Besides, from a supposition that the sounds of snare drums are consisted of high-and low-frequency parts separated at 2kHz that each of them mainly contain sounds of a snare and drum's shells respectively, ``Spectral Centroid'' and ``Sustain Time 60(ST60)'' for the two parts were calculated, and Correlation Coefficient between the physical measure and the two-dimensional stimulus space were calculated. A series of studies revealed that each dimensions achieved by MDS were related to ST60 (especially sustain of drum's shells) and Spectral Centroid below 2kHz (pitch of drum's shells).

A5 - 2 Contextual dependence of auditory attribute ratings: Incorporating stimulus blocking factors into physical predictors for apparent source width and sharpness

When auditory attribute ratings are dependent on experimental context, incorporating task variables into prediction models should improve the success of those models. One such task variable is the way in which stimuli are blocked together into multiple comparison trials for collecting ratings. In this study, contextual dependence of physical predictors for two auditory attributes, apparent source width and sharpness, was examined by incorporated stimulus blocking factors into the prediction equations. The results showed that such an approach to prediction gave a much better fit for obtained sharpness ratings than did conventional physical measures. In contrast, ratings of auditory source width were found to be much less affected by context, perhaps because listeners rely upon a more absolute internal reference for spatial extent when making subjective judgments of width.

A5 - 3 Investigations of Using M-S Technique in Loudspeaker Array for Sound Reinforcement

Loudspeaker array design employing M-S technique is a novel approach in sound reinforcement. The experimental design proposes a number of advantages over conventional L-R stereo system for applications in proscenium theaters and black box theaters. It is evident that a solid center sonic image and consistent stereo width can be enjoyed across a broader area in the auditorium especially with low aspect ratio when compared to L-R stereo system. This paper summarizes the attempts of M-S speaker array design, alignment procedures and signal processing applied in previous theatrical productions, and a speaker system for subjective testing is devised. Various stereo music materials are tested and the subjective test results are discussed.

A5 - 4 奏楽堂の可変天井による残響の変化と録音への影響について

A concert hall Sogakudo has adjustable ceiling panels. Acoustical changes due to the adjustable ceiling panels were investigated, and how those changes affect the recordings made in the venue. Impulse responses were recorded at seven settings of ceiling panels. Especially, recordings using the ``Omni-8'' array were made at reference position, 1m near, and 1m far from the loudspeaker. Subjective evaluations of 12 sound stimuli of four ceiling panel settings and three frontal microphone positions were done. Three music excerpts were convoluted with the 12 impulse response files resulting in 36 music stimuli. Three attributes were used for the evaluation. As a result, we got some information such as ceiling panel setting usually used for piano was rated higher in "Comfortableness" in all stimuli played solo.

P3 — 三次元音響と再生

P3 - 1 仮想環境のためのマルチチャネル音響合成法に関する検討

A method on multi-channel sound synthesis is proposed for creating virtual acoustic environment in computer graphics productions. Impulse responses with spatial information are generated from virtual sound source distribution. For desired reproduction channel setups, total impulse response is decomposed into channel components, and each response is convolved with source signal. The method is easily adaptable when location and direction of observation point in virtual graphical scene switches frequently.

In this paper, we proposed a new method using HRTFs that contain room reverberations(R-HRTF). The reverberation is not added to the dry sound source separated with HRTF but contained at their measured process in the HRTFs. We measured the HRTFs in a real reverberant environment for directions of azimuth 45, 90, 135 (left side) and elevation from 0 to 90 (step of 10 degrees) degrees then constructed a 3D sound system with the measured R-HRTF with headphones, examine if the sound reality is improved. As a result, we succeed to create 3D spatial sound system with more reality compared with traditional HRTFs sound system.

Artificial reverberation is often used to increase reality and prevent the in-the-head localization in a headphone based 3-D sound system. In traditional method, monaural reverberations were used. In this research, we measured impulse responses of an ordinary room by Four Point Microphone method, and calculated the sound intensity vectors by the Sound Intensity method. From the sound intensity vectors, we obtained the image sound sources. A binaural reverberation was reconstructed by the estimated image sound sources. Comparison experiments were conducted for 3 kinds of reverberations, i.e., monaural reverberation, binaural reverberation and binaural reverberation added with Head-Related Transfer Function. From the results, we could clarify the 3-D sounds reconstructed by binaural reverberation with head-related transfer function has the best spatiality.

P3 - 4 低周波数帯域における両耳間相関係数の弁別調査とダミーヘッドマイクを用いた実音場計測インパルス応答による検証

高山泰典（日本文理大学），近藤善隆（日本文理大学），今井佐智代（千葉工業大学），松本博樹（日本文理大学），末廣一美（日本文理大学），岩上知広（千葉工業大学），楳田美奈子（千葉工業大学），福島学（日本文理大学），柳川博文（千葉工業大学）

We investigated the discrimination of sounds that have interaural correlation coefficient (ICC) of low frequency sound below around 100 Hz by using 1/4 octave band noise in dichotic listening. We also measured ICCs at low frequencies in several rooms with different volumes. The ICCs were obtained from an impulse response recorded by a head and torso simulator. At reference ICC 1.0 the lower threshold is 0.98; at 0.9 it is 0.85, and the upper threshold is 0.96; and at 0.8 the lower one is 0.68, and the upper is 0.88. The equivalent subjective diffuseness for the 1/4 octave band noise with the center frequency of 106 Hz and ICC of 0.9, are obtained by ICC of 0.71 at 250 Hz and ICC of 0.34 at 500 Hz. The measured ICC in a small room with a volume of less than 100 m3 is almost 1, and the ICC in large room with a volume of more than 1700 m3 is about 0.8.

P3 - 5 耳の逆側における簡略化されたHRTFを用いた音像定位能力の主観評価

渡邉貫治 (秋田県立大学システム科学技術学部), 小玉亮介 (秋田県立大学大学院システム科学技術研究科), 佐藤宗純 (秋田県立大学システム科学技術学部), 高根昭一 (秋田県立大学システム科学技術学部), 安倍幸治 (秋田県立大学システム科学技術学部)

Simplification of head-related transfer functions (HRTFs) is important for effective implementation of their synthesis from computational point of view. It can be found from the frequency resolution of the auditory system that the detailed spectral form of the HRTFs is not evaluated at high frequency region. This may enable the simplification of the HRTFs to some extent. In this paper, the HRTF on the contralateral side was flattened in the higher frequency region than a certain frequency so as to retain the interaural level difference and interaural time difference. To evaluate the influence of simplified HRTFs, a localization test was carried out. The experimental results showed that HRTFs on the contralateral side could be simplified above 4 kHz.

P3 - 6 後方音源の頭部伝達関数における低域の周波数特性上の谷の空間分布

Previously it was reported that the lowest-frequency spectral notch (first notch) on head-related transfer function (HRTF) play a role in sound image localization; high-frequency spectral notches (called N1 and N2) are important for sound image localization in elevation. However, another spectral notch (labeled N0) at relatively lower frequency appear in the same frequency range for sound sources behind a listener. In this study, we examined the spatial distribution of such low-frequency notches based on several subjects' measured HRTFs, and subsequently investigated whether such spectral notches can contribute to sound localization. The results showed that, in some cases, N0 exists at lower frequency than N1, indicating that a N0 provides localization cues for sound sources behind a listener.

This is a report on our challenge and achievement of creating unique performance specifications and sound quality through adoption of smooth curved surface for overall dynamic speaker system design. Enclosure is shaped like an egg and speaker diaphragm is curved to match the overall enclosure form. In order to achieve smooth and even sound radiation without any disturbance, speaker edge surround and frame are not visible from outside and there is no speaker grill. As a result, we were able to develop a speaker system with very high S/N ratio across wide frequency range, allowing reproduction of subtle nuances, and with ability to create superb sound stage. Furthermore, a large number of favorable attributes, in addition to wide dispersion and smooth sound radiation, could be observed.

P3 - 8 スーパーハイビジョン・２２．２マルチチャンネル音響システムにおけるコンテンツ制作手法

The 22.2 multichannel sound system is a high-presence sound format that is superior to the 5.1 surround sound system, and NHK intends it to be the format for the next generation of TV broadcasting. NHK started to make Super Hi-Vision (SHV) content for experiments and exhibitions in 2002 and exhibited the 22.2 multichannel sound system as part of the SHV system for the first time at the Aichi World Expo in 2005. This system has also been demonstrated at the National Association of Broadcasters (NAB) convention and the International Broadcasting Convention (IBC). This paper reports on the 22.2 system's method of recording sound fields and the technique of achieving ``3D stereophony'' in post-production for the program "Gift" that was exhibited at NAB2009.

P3 - 9 ２２．２マルチチャンネルサラウンドによる大型音楽番組「紅白歌合戦」の３次元立体音響のためのサウンドデザイン及びリアルタイムミキシング手法

The public viewing of ``Kouhaku Uta Gassenn'', one of the most famous music programs in Japan, was held at NHK Fureai Hall in Tokyo on December 31, 2008. The program was mixed live using the 22.2 multichannel system, which is highly expected to be a potential audio format for future TV broadcasting audio format. This paper introduces an overview of this public viewing, including the 22.2 multichannel sound system flow and live three-dimensional sound mixing technique.

AES 東京コンベンション 2009
技術発表詳細

A1 — 機器と測定

P1 — 室内音響と聴取実験

A2 — 電気音響変換器

A3 — 室内音響

P2 — 測定と信号処理

A4 — ディジタルオーディオ機器とメディア

A5 — 心理音響・知覚と聴取実験

P3 — 三次元音響と再生

AES 東京コンベンション 2009 技術発表詳細

A1 — 機器と測定

P1 — 室内音響と聴取実験

A2 — 電気音響変換器

A3 — 室内音響

P2 — 測定と信号処理

A4 — ディジタルオーディオ機器とメディア

A5 — 心理音響・知覚と聴取実験

P3 — 三次元音響と再生

AES 東京コンベンション 2009
技術発表詳細