• Title/Summary/Keyword: microphone in real ear

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Characteristics of Sound Response in Ear Canal of Human and Reproduction of Acoustical Space (인간 이도의 소리응답특성과 음향공간의 재현)

  • Ahn, Tae-Soo;Lee, Doo-Ho
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.21 no.9
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    • pp.842-849
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    • 2011
  • The human ear canal amplifies the sound pressure level at specific frequency bands. The characteristics of the ear canal are very similar to those of curved cylindrical tube. In this study, the characteristics of sound transfer in human ear canal were measured and the acoustical space of ear canal was reproduced from the canal cavity geometry. For the measurement of sound transfer function in ear canal, a probe microphone and a reference microphone were used. The sound transfer functions were measured for 5 human subjects. To reproduce the acoustical space of the ear canal, two kinds of ear simulator were designed. The first one is a straight cylindrical tube type and the other is a real-shape ear of which geometry was taken from a micro-CT scanning of a human ear. The characteristics of the reproduced apparatus were compared with those of the human and a commercial ear simulator, RA0045 of G.R.A.S. Inc. The comparison results show that the developed apparatus well represent the ear canal characteristics in the low frequency, but have limited coincidence in level over high frequency range.

Formant frequency changes of female voice /a/, /i/, /u/ in real ear (실이에서 여자 음성 /ㅏ/, /ㅣ/, /ㅜ/의 포먼트 주파수 변화)

  • Heo, Seungdeok;Kang, Huira
    • Phonetics and Speech Sciences
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    • v.9 no.1
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    • pp.49-53
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    • 2017
  • Formant frequencies depend on the position of tongue, the shape of lips, and larynx. In the auditory system, the external ear canal is an open-end resonator, which can modify the voice characteristics. This study investigates the effect of the real ear on formant frequencies. Fifteen subjects ranging from 22 to 30 years of age participated in the study. This study employed three corner vowels: the low central vowel /a/, the high front vowel /i/, and the high back vowel /u/. For this study, the voice of a well-educated undergraduate who majored in speech-language pathology, was recorded with a high performance condenser microphone placed in the upper pinna and in the ear canal. Paired t-test showed that there were significant difference in the formant frequencies of F1, F2, F3, and F4 between the free field and the real ear. For /a/, all formant frequencies decreased significantly in the real ear. For /i/, F2 increased and F3 and F4 decreased. For /u/, F1 and F2 increased, but F3 and F4 decreased. It seems that these voice modifications in the real ear contribute to interpreting voice quality and understanding speech, timbre, and individual characteristics, which are influenced by the shape of the outer ear and external ear canal in such a way that formant frequencies become centralized in the vowel space.

Individual Fit Testing of Hearing Protection Devices Based on Microphone in Real Ear

  • Biabani, Azam;Aliabadi, Mohsen;Golmohammadi, Rostam;Farhadian, Maryam
    • Safety and Health at Work
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    • v.8 no.4
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    • pp.364-370
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    • 2017
  • Background: Labeled noise reduction (NR) data presented by manufacturers are considered one of the main challenging issues for occupational experts in employing hearing protection devices (HPDs). This study aimed to determine the actual NR data of typical HPDs using the objective fit testing method with a microphone in real ear (MIRE) method. Methods: Five available commercially earmuff protectors were investigated in 30 workers exposed to reference noise source according to the standard method, ISO 11904-1. Personal attenuation rating (PAR) of the earmuffs was measured based on the MIRE method using a noise dosimeter (SVANTEK, model SV102). Results: The results showed that means of PAR of the earmuffs are from 49% to 86% of the nominal NR rating. The PAR values of earmuffs when a typical eyewear was worn differed statistically (p < 0.05). It is revealed that a typical safety eyewear can reduce the mean of the PAR value by approximately 2.5 dB. The results also showed that measurements based on the MIRE method resulted in low variability. The variability in NR values between individuals, within individuals, and within earmuffs was not the statistically significant (p > 0.05). Conclusion: This study could provide local individual fit data. Ergonomic aspects of the earmuffs and different levels of users experience and awareness can be considered the main factors affecting individual fitting compared with the laboratory condition for acquiring the labeled NR data. Based on the obtained fit testing results, the field application of MIRE can be employed for complementary studies in real workstations while workers perform their regular work duties.

Active Sound Control Approach Using Virtual Microphones for Formation of Quiet Zones at a Chair (좌석의 정음공간 형성을 위한 가상마이크로폰 기반 능동음향제어 기법 연구)

  • Ryu, Seokhoon;Kim, Jeakwan;Lee, Young-Sup
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.25 no.9
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    • pp.628-636
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    • 2015
  • In this study, theoretical and experimental analyses were performed for creating and moving the zone of quiet(ZoQ) to the ear location of a sitter by using active sound control technique. As the ZoQ is actively created at the location of the error microphone basically with an active sound control system using an algorithm such as the filtered-x least mean square(FxLMS), the virtual microphone control(VMC) method was considered to move the location of the ZoQ to around the sitter`s ear. A chair system with microphones and loudspeakers on both sides was manufactured for the experiment and thus an active headrest against the swept narrowband noise as the primary noise was implemented with a real-time controller in which the VMC algorithm was embedded. After the control experiment with and without the VMC method, the location variation of the ZoQ by analyzing the error signals measured by the error and the virtual microphones. Therefore, it is observed that the FxLMS with the VMC technique can provide the re-location of the ZoQ from the error microphone location to the virtual microphone location. Also it is found that the amount of the attenuation difference between the two locations was small.

Quasi-Optimal Linear Recursive DOA Tracking of Moving Acoustic Source for Cognitive Robot Auditory System (인지로봇 청각시스템을 위한 의사최적 이동음원 도래각 추적 필터)

  • Han, Seul-Ki;Ra, Won-Sang;Whang, Ick-Ho;Park, Jin-Bae
    • Journal of Institute of Control, Robotics and Systems
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    • v.17 no.3
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    • pp.211-217
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    • 2011
  • This paper proposes a quasi-optimal linear DOA (Direction-of-Arrival) estimator which is necessary for the development of a real-time robot auditory system tracking moving acoustic source. It is well known that the use of conventional nonlinear filtering schemes may result in the severe performance degradation of DOA estimation and not be preferable for real-time implementation. These are mainly due to the inherent nonlinearity of the acoustic signal model used for DOA estimation. This motivates us to consider a new uncertain linear acoustic signal model based on the linear prediction relation of a noisy sinusoid. Using the suggested measurement model, it is shown that the resultant DOA estimation problem is cast into the NCRKF (Non-Conservative Robust Kalman Filtering) problem [12]. NCRKF-based DOA estimator provides reliable DOA estimates of a fast moving acoustic source in spite of using the noise-corrupted measurement matrix in the filter recursion and, as well, it is suitable for real-time implementation because of its linear recursive filter structure. The computational efficiency and DOA estimation performance of the proposed method are evaluated through the computer simulations.

An Adaptive Microphone Array with Linear Phase Response (선형 위상 특성을 갖는 적응 마이크로폰 어레이)

  • Kang, Hong-Gu;Youn, Dae-Hui;Cha, Il-Hwan
    • The Journal of the Acoustical Society of Korea
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    • v.11 no.3
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    • pp.53-60
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    • 1992
  • Many adaptive beamforming methods have been studied for interference cancellation and speech signal enhancement in telephone conference and auditorium. Main aspect of adaptive beamforming methods for speech signal processing is different from radar, sonar and seismic signal processing because desire output signal should be apt to the human ear. Considering that phase of speech is quite insensible to the human ear, Sondhi proposed a nonlinear constrained optimization technique whose constraint was on the magnitude transfer function from the source to the output. In real environment the phase response of the speech signal affects the human auditorium system. So it is desirable to design linear phase system. In this paper, linear phase beamformer is proposed and sample processing algorithm is also proposed for real time consideration Simulation results show that the proposed algorithm yields more consistent beam patterns and deep nulls to the noise direction than Sondhi's.

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