Current Organizations / Institutes

The organizations/institutes from the scholar's latest article


Shunsuke Kidani has a total of 22 co-authors. Here are the top co-authors by number of shared publications:


Latest Publications

  • Open Access

    Lack of Effect of Chlordiazepoxide on the Number of Falls in a Novel Falling‐Trap Test in Male C57BL / 6N Mice

    by Shunsuke Kidani, Masatoshi Ukezono, Yuji Takano, Yasuhide Mitsumoto

    Behavioral tests in mice and other animals are widely used to study anxiety and to evaluate the efficacy of anxiolytic compounds. However, the validity of behavioral tests used to assess anxiety‐like behavior has been questioned because their results are often inconsistent and not readily replicated across studies. We developed a simple falling‐trap test to examine whether the number of falls could serve as an index of anxiety‐like behavior. The test is designed to produce falls from an elevated path, based on the hypothesis that less‐anxious mice would be more prone to fall. Male C57BL/6N mice were given vehicle or chlordiazepoxide (CDP), and their behavior was evaluated using both the novel test and the elevated plus maze (EPM). The outcomes were the number of falls in the falling‐trap test and open‐arm time in the EPM. In the falling‐trap test, CDP‐treated mice did not show a statistically significant increase in the number of falls compared with control mice. In contrast, CDP significantly increased open‐arm time in the EPM. Thus, under the present experimental conditions, these findings did not support the hypothesis that the number of falls in the falling‐trap test could serve as a measure of anxiety‐like behavior. Further studies are needed to determine the conditions under which this assay could be useful and to establish its reliability and validity.

  • Open Access

    Cross-Domain Generalization From Synthesized to Human-Imitated Speech Detection

    by Khalid Zaman, Kai Li, Anuwat Chaiwongyen, Yasufumi Uezu, Shunsuke Kidani, Masashi Unoki


  • Open Access

    Ability of Human Auditory Perception to Distinguish Human-Imitated Speech

    by Khalid Zaman, Kai Li, Islam J. A. M. Samiul, Yasufumi Uezu, Shunsuke Kidani, Masashi Unoki

    Distinguishing human-imitated speech from genuine speech presents a significant challenge for listeners due to their natural resemblance. Human auditory perception (HAP) has been widely studied to understand its mechanisms, with HAP-based acoustic features and metrics applied in various applications to assess sound quality and discriminate sound events. Leveraging these insights, this study specifically aims to evaluate HAP’s effectiveness in differentiating genuine from imitated speech through a systematic subject test. To this end, the study applies HAP to the task of distinguishing genuine from imitated speech, using a specially developed dataset of human-imitated speech, due to the lack of comparable publicly available datasets. A three-phase, human-centered approach was used to evaluate HAP ability, and participants achieved an average accuracy of 70.10% in distinguishing genuine from imitated speech in the final test. Additionally, a feasibility study was conducted using two feature sets for machine classification; among the timbral features, boominess and depth performed best with accuracies of 62% and 60%, respectively, while general features like Mel-spectrograms reached 51%. These results underscore the importance of auditory-related features in effectively detecting imitated speech.


  • Open Access

    Computational models of sound-quality metrics using method for calculating loudness with gammatone/gammachirp auditory filterbank

    by Takuto Isoyama, Shunsuke Kidani, Masashi Unoki

    Sound-quality metrics (SQMs), such as sharpness, roughness, and fluctuation strength, are calculated using a standard method for calculating loudness (Zwicker method, ISO532B, 1975). Since ISO 532 had been revised to contain the Zwicker method (ISO 5321) and Moore-Glasberg method (ISO 532-2) in 2017, the classical computational SQM model should also be revised in accordance with these revisions. A roex auditory filterbank used with the Moore-Glasberg method is defined separately in the frequency domain not to have impulse responses. It is therefore difficult to construct a computational SQM model, e.g., the classical computational SQM model, on the basis of ISO 532-2. We propose a method for calculating loudness using the time-domain gammatone or gammachirp auditory filterbank instead of the roex auditory filterbank to solve this problem. We also propose three computational SQM models based on ISO 532-2 to use with the proposed loudness method. We evaluated the root-mean squared errors (RMSEs) of the calculated loudness with the proposed and Moore-Glasberg methods. We then evaluated the RMSEs of the calculated SQMs with the proposed method and human data of SQMs. We found that the proposed method can be considered as a time-domain method for calculating loudness on the basis of ISO 532-2 because the RMSEs are very small. We also found that the proposed computational SQM models can effectively account for the human data of SQMs compared with the classical computational SQM model in terms of RMSEs.

  • Open Access

    Contributions of Temporal Modulation Cues in Temporal Amplitude Envelope of Speech to Urgency Perception

    by Masashi Unoki, Miho Kawamura, Maori Kobayashi, Shunsuke Kidani, Junfeng Li, Masato Akagi

    We previously investigated the perception of noise-vocoded speech to determine whether the temporal amplitude envelope (TAE) of speech plays an important role in the perception of linguistic information as well as non-linguistic information. However, it remains unclear if these TAEs also play a role in the urgency perception of non-linguistic information. In this paper, we comprehensively investigated whether the TAE of speech contributes to urgency perception. To this end, we compared noise-vocoded stimuli containing TAEs identical to those of original speech with those containing TAEs controlled by low-pass or high-pass filtering. We derived degrees of urgency from a paired comparison of the results and then used them as a basis to clarify the relationship between the temporal modulation components in TAEs of speech and urgency perception. Our findings revealed that (1) the perceived degrees of urgency of noise-vocoded stimuli are similar to those of the original, (2) significant cues for urgency perception are temporal modulation components of the noise-vocoded stimuli higher than the modulation frequency of 6 Hz, (3) additional significant cues for urgency perception are temporal modulation components of the noise-vocoded stimuli lower than the modulation frequency of 8 Hz, and (4) the TAE of the time-reversal speech is not likely to contain important cues for the perception of urgency. We therefore conclude that temporal modulation cues in the TAE of speech are a significant component in the perception of urgency.


  • Open Access

    Contribution of Common Modulation Spectral Features to Vocal-Emotion Recognition of Noise-Vocoded Speech in Noisy Reverberant Environments

    by Taiyang Guo, Zhi Zhu, Shunsuke Kidani, Masashi Unoki

    In one study on vocal emotion recognition using noise-vocoded speech (NVS), the high similarities between modulation spectral features (MSFs) and the results of vocal-emotion-recognition experiments indicated that MSFs contribute to vocal emotion recognition in a clean environment (with no noise and no reverberation). Other studies also clarified that vocal emotion recognition using NVS is not affected by noisy reverberant environments (signal-to-noise ratio is greater than 10 dB and reverberation time is less than 1.0 s). However, the contribution of MSFs to vocal emotion recognition in noisy reverberant environments is still unclear. We aimed to clarify whether MSFs can be used to explain the vocal-emotion-recognition results in noisy reverberant environments. We analyzed the results of vocal-emotion-recognition experiments and used an auditory-based modulation filterbank to calculate the modulation spectrograms of NVS. We then extracted ten MSFs as higher-order statistics of modulation spectrograms. As shown from the relationship between MSFs and vocal-emotion-recognition results, except for extremely high noisy reverberant environments, there were high similarities between MSFs and the vocal emotion recognition results in noisy reverberant environments, which indicates that MSFs can be used to explain such results in noisy reverberant environments. We also found that there are two common MSFs (MSKTk (modulation spectral kurtosis) and MSTLk (modulation spectral tilt)) that contribute to vocal emotion recognition in all daily environments.


  • Open Access

    Blind Speech Watermarking Method with Frame Self-Synchronization Based on Spread-Spectrum Using Linear Prediction Residue

    by Takuto Isoyama, Shunsuke Kidani, Masashi Unoki

    State-of-the-art speech watermarking techniques enable speech signals to be authenticated and protected against any malicious attack to ensure secure speech communication. In general, reliable speech watermarking methods must satisfy four requirements: inaudibility, robustness, blind-detectability, and confidentiality. We previously proposed a method of non-blind speech watermarking based on direct spread spectrum (DSS) using a linear prediction (LP) scheme to solve the first two issues (inaudibility and robustness) due to distortion by spread spectrum. This method not only effectively embeds watermarks with small distortion but also has the same robustness as the DSS method. There are, however, two remaining issues with blind-detectability and confidentiality. In this work, we attempt to resolve these issues by developing an approach called the LP-DSS scheme, which takes two forms of data embedding for blind detection and frame synchronization. We incorporate blind detection with frame synchronization into the scheme to satisfy blind-detectability and incorporate two forms of data embedding process, front-side and back-side embedding for blind detection and frame synchronization, to satisfy confidentiality. We evaluated these improved processes by carrying out four objective tests (PESQ, LSD, Bit-error-rate, and accuracy of frame synchronization) to determine whether inaudibility and blind-detectability could be satisfied. We also evaluated all combinations with the two forms of data embedding for blind detection with frame synchronization by carrying out BER tests to determine whether confidentiality could be satisfied. Finally, we comparatively evaluated the proposed method by carrying out ten robustness tests against various processing and attacks. Our findings showed that an inaudible, robust, blindly detectable, and confidential speech watermarking method based on the proposed LP-DSS scheme could be achieved.


  • Open Access

    Speaker anonymization by modifying fundamental frequency and x-vector singular value

    by Candy Olivia Mawalim, Kasorn Galajit, Jessada Karnjana, Shunsuke Kidani, Masashi Unoki


  • Presentation effect of cue tone on tuning of auditory filter for several frequencies

    by Shunsuke Kidani, Ryota Miyauchi, Masashi Unoki

    Japan's largest platform for academic e-journals: J-STAGE is a full text database for reviewed academic papers published by Japanese societies