When a person listens to loudspeakers, the perceived sound is affected not only by the loudspeaker properties but also by the acoustics of the surroundings. Loudspeaker equalization can be used to correct the loudspeaker-room response. However, when the listener moves in front of the loudspeakers, both the loudspeaker response and room effect change. In order for the best correction to be achieved at all times, adaptive equalization is proposed in this paper. A loudspeaker-correction system using the listener’s current location to determine the correction parameters is proposed. The position of the listener’s head is located using a depth- sensing camera, and suitable equalizer settings are then selected based on measurements and interpolation.Aftercorrectingfortheloudspeaker’sresponseatmultiplelocationsandchangingtheequalizationinrealtimebasedontheuser’slocation,aloudspeakerresponsewithreducedcolorationisachievedcomparedtonocalibrationorconventionalcalibrationmethods,withthemagnitude-responsedeviationsdecreasingfrom10.0to5.6dBwithinthepassbandofahigh-qualityloudspeaker.Theproposedmethodcanimprovetheaudiomonitoringinmusicstudiosandotheroccasionsinwhichasinglelistenerismovinginarestrictedspace.
{"title":"Loudspeaker Equalization for a Moving Listener","authors":"Joel Lindfors, Juho Liski, V. Välimäki","doi":"10.17743/jaes.2022.0020","DOIUrl":"https://doi.org/10.17743/jaes.2022.0020","url":null,"abstract":"When a person listens to loudspeakers, the perceived sound is affected not only by the loudspeaker properties but also by the acoustics of the surroundings. Loudspeaker equalization can be used to correct the loudspeaker-room response. However, when the listener moves in front of the loudspeakers, both the loudspeaker response and room effect change. In order for the best correction to be achieved at all times, adaptive equalization is proposed in this paper. A loudspeaker-correction system using the listener’s current location to determine the correction parameters is proposed. The position of the listener’s head is located using a depth- sensing camera, and suitable equalizer settings are then selected based on measurements and interpolation.Aftercorrectingfortheloudspeaker’sresponseatmultiplelocationsandchangingtheequalizationinrealtimebasedontheuser’slocation,aloudspeakerresponsewithreducedcolorationisachievedcomparedtonocalibrationorconventionalcalibrationmethods,withthemagnitude-responsedeviationsdecreasingfrom10.0to5.6dBwithinthepassbandofahigh-qualityloudspeaker.Theproposedmethodcanimprovetheaudiomonitoringinmusicstudiosandotheroccasionsinwhichasinglelistenerismovinginarestrictedspace.","PeriodicalId":50008,"journal":{"name":"Journal of the Audio Engineering Society","volume":" ","pages":""},"PeriodicalIF":1.4,"publicationDate":"2022-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"49538954","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":4,"RegionCategory":"工程技术","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Juan Carlos Franco Hernández, Bogdan Baǎcilǎ, Tim S. Brookes, E. De Sena
A new publicly available dataset of microphone impulse responses (IRs) has been gener- ated. The dataset covers 25 microphones, including a Class-1 measurement microphone and polar pattern variations for seven of the microphones. Microphones that were included had omnidirectional, cardioid, supercardioid, and bidirectional polar patterns; condenser, moving-coil, and ribbon transduction types; single and dual diaphragms; multiple body and head basket shapes;smallandlargediaphragms;andend-addressandside-addressdesigns.Usingacustom-developedcomputer-controlledprecisionturntable,IRswerecapturedquasi-anechoicallyatincidentanglesfrom0 ◦ to 355 ◦ in steps of 5 ◦ and at source-to-microphone distances of 0.5, 1.25, and 5 m. The resulting dataset is suitable for perceptual and objective studies related to the incident-angle–dependent response of microphones and for the development of tools for predicting and emulating on-axis and off-axis microphone characteristics. The captured IRs allow generation of frequency response plots with a degree of detail not commonly available in manufacturer-supplied data sheets and are also particularly well-suited to harmonic distortion analysis.
{"title":"A Multi-Angle, Multi-Distance Dataset of Microphone Impulse Responses","authors":"Juan Carlos Franco Hernández, Bogdan Baǎcilǎ, Tim S. Brookes, E. De Sena","doi":"10.17743/jaes.2022.0027","DOIUrl":"https://doi.org/10.17743/jaes.2022.0027","url":null,"abstract":"A new publicly available dataset of microphone impulse responses (IRs) has been gener- ated. The dataset covers 25 microphones, including a Class-1 measurement microphone and polar pattern variations for seven of the microphones. Microphones that were included had omnidirectional, cardioid, supercardioid, and bidirectional polar patterns; condenser, moving-coil, and ribbon transduction types; single and dual diaphragms; multiple body and head basket shapes;smallandlargediaphragms;andend-addressandside-addressdesigns.Usingacustom-developedcomputer-controlledprecisionturntable,IRswerecapturedquasi-anechoicallyatincidentanglesfrom0 ◦ to 355 ◦ in steps of 5 ◦ and at source-to-microphone distances of 0.5, 1.25, and 5 m. The resulting dataset is suitable for perceptual and objective studies related to the incident-angle–dependent response of microphones and for the development of tools for predicting and emulating on-axis and off-axis microphone characteristics. The captured IRs allow generation of frequency response plots with a degree of detail not commonly available in manufacturer-supplied data sheets and are also particularly well-suited to harmonic distortion analysis.","PeriodicalId":50008,"journal":{"name":"Journal of the Audio Engineering Society","volume":" ","pages":""},"PeriodicalIF":1.4,"publicationDate":"2022-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"42003071","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":4,"RegionCategory":"工程技术","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Aluminum-Based Push-Pull Electrostatic MEMS Transducer for Earphones","authors":"Aviad Zamir, G. Seiden, H. Kupershmidt","doi":"10.17743/jaes.2022.0035","DOIUrl":"https://doi.org/10.17743/jaes.2022.0035","url":null,"abstract":"","PeriodicalId":50008,"journal":{"name":"Journal of the Audio Engineering Society","volume":" ","pages":""},"PeriodicalIF":1.4,"publicationDate":"2022-11-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"42141113","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":4,"RegionCategory":"工程技术","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Background: After the publication of the Japanese Circulation Society guideline of sleep-disordered breathing (SDB) in 2010, with new evidence and changes to the health insurance system, trends in the practice pattern for SDB in patients with cardiovascular disease (CVD) might have changed.
Methods and results: This study evaluated the temporal changes in the practice pattern for SDB by using a nationwide claim database, the Japanese Registry of All Cardiac and Vascular Diseases - Diagnosis Procedure Combination (JROAD-DPC), from 2012 to 2019. The main findings were: (1) the number of CVD patients diagnosed with SDB increased (especially those with atrial fibrillation [AF] and heart failure [HF]); (2) the number of diagnostic tests for SDB performed during hospitalization increased for AF patients (from 1.3% in 2012 to 1.8% in 2019), whereas it decreased for other CVD patients; (3) the number of patients diagnosed with SDB increased in each type of CVD, except for patients with acute myocardial infarction (AMI); (4) continuous positive airway pressure (CPAP) treatment increased for AF patients (from 15.2% to 17.5%); (5) CPAP treatment decreased for patients with angina pectoris (AP) and AMI, and any treatment decreased for HF patients (from 46.1% to 39.7%); and (6) SDB was treated more often in HF patients than in AF, AP, and AMI patients (41.7% vs. 17.2%, 19.1% and 20.4%, respectively).
Conclusions: The practice pattern for SDB in CVD patients has changed from 2012 to 2019.
{"title":"Temporal Trends in the Practice Pattern for Sleep-Disordered Breathing in Patients With Cardiovascular Diseases in Japan - Insights From the Japanese Registry of All Cardiac and Vascular Diseases - Diagnosis Procedure Combination.","authors":"Ryohei Takeishi, Akiomi Yoshihisa, Yu Hotsuki, Fumiya Anzai, Yu Sato, Yoko Sumita, Michikazu Nakai, Tomofumi Misaka, Yasuchika Takeishi","doi":"10.1253/circj.CJ-22-0082","DOIUrl":"10.1253/circj.CJ-22-0082","url":null,"abstract":"<p><strong>Background: </strong>After the publication of the Japanese Circulation Society guideline of sleep-disordered breathing (SDB) in 2010, with new evidence and changes to the health insurance system, trends in the practice pattern for SDB in patients with cardiovascular disease (CVD) might have changed.</p><p><strong>Methods and results: </strong>This study evaluated the temporal changes in the practice pattern for SDB by using a nationwide claim database, the Japanese Registry of All Cardiac and Vascular Diseases - Diagnosis Procedure Combination (JROAD-DPC), from 2012 to 2019. The main findings were: (1) the number of CVD patients diagnosed with SDB increased (especially those with atrial fibrillation [AF] and heart failure [HF]); (2) the number of diagnostic tests for SDB performed during hospitalization increased for AF patients (from 1.3% in 2012 to 1.8% in 2019), whereas it decreased for other CVD patients; (3) the number of patients diagnosed with SDB increased in each type of CVD, except for patients with acute myocardial infarction (AMI); (4) continuous positive airway pressure (CPAP) treatment increased for AF patients (from 15.2% to 17.5%); (5) CPAP treatment decreased for patients with angina pectoris (AP) and AMI, and any treatment decreased for HF patients (from 46.1% to 39.7%); and (6) SDB was treated more often in HF patients than in AF, AP, and AMI patients (41.7% vs. 17.2%, 19.1% and 20.4%, respectively).</p><p><strong>Conclusions: </strong>The practice pattern for SDB in CVD patients has changed from 2012 to 2019.</p>","PeriodicalId":50008,"journal":{"name":"Journal of the Audio Engineering Society","volume":"1 1","pages":"1428-1436"},"PeriodicalIF":3.1,"publicationDate":"2022-08-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"85721214","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":4,"RegionCategory":"工程技术","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Chaofeng Lan, Yuqiao Wang, Lei Zhang, Hongyun Zhao
{"title":"Research on Additive Margin Softmax Speaker Recognition Based on Convolutional and Gated Recurrent Neural Networks","authors":"Chaofeng Lan, Yuqiao Wang, Lei Zhang, Hongyun Zhao","doi":"10.17743/jaes.2022.0018","DOIUrl":"https://doi.org/10.17743/jaes.2022.0018","url":null,"abstract":"","PeriodicalId":50008,"journal":{"name":"Journal of the Audio Engineering Society","volume":"1 1","pages":""},"PeriodicalIF":1.4,"publicationDate":"2022-07-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"67642055","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":4,"RegionCategory":"工程技术","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Raimundo Gonzalez, Thomas McKenzie, A. Politis, T. Lokki
The spatial speech reproduction capabilities of a KEMAR mouth simulator, a loudspeaker, the piston on sphere model and a circular harmonic fitting are evaluated in the near-field. The speech directivity of 24 human subjects, both male and female, is measured using a semi-circular microphone array of radius 36.5 cm in the horizontal plane. Impulse responses are captured for the two devices and filters are generated for the two numerical models to emulate their directional effect on speech reproduction. The four repeatable speech sources are evaluated through comparison to the recorded human speech both objectively, through directivity pattern and spectral magnitude differences, and subjectively, through a listening test on perceived coloration. Results show that the repeatable sources perform relatively well under the metric of directivity but irregularities in their directivity patterns introduce audible coloration for off-axis directions.
{"title":"Near-Field Evaluation of Reproducible Speech Sources","authors":"Raimundo Gonzalez, Thomas McKenzie, A. Politis, T. Lokki","doi":"10.17743/jaes.2022.0022","DOIUrl":"https://doi.org/10.17743/jaes.2022.0022","url":null,"abstract":"The spatial speech reproduction capabilities of a KEMAR mouth simulator, a loudspeaker, the piston on sphere model and a circular harmonic fitting are evaluated in the near-field. The speech directivity of 24 human subjects, both male and female, is measured using a semi-circular microphone array of radius 36.5 cm in the horizontal plane. Impulse responses are captured for the two devices and filters are generated for the two numerical models to emulate their directional effect on speech reproduction. The four repeatable speech sources are evaluated through comparison to the recorded human speech both objectively, through directivity pattern and spectral magnitude differences, and subjectively, through a listening test on perceived coloration. Results show that the repeatable sources perform relatively well under the metric of directivity but irregularities in their directivity patterns introduce audible coloration for off-axis directions.","PeriodicalId":50008,"journal":{"name":"Journal of the Audio Engineering Society","volume":" ","pages":""},"PeriodicalIF":1.4,"publicationDate":"2022-07-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"47825546","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":4,"RegionCategory":"工程技术","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
A computationally efficient octave-band graphic equalizer having a linear-phase response is introduced. The linear-phase graphic equalizer is useful in audio applications in which phase distortion is not tolerated, such as in multichannel equalization, parallel processing, phase compatibility of audio equipment, and crossover network design. The structure is based on the interpolated finite impulse response (IFIR) philosophy. The proposed octave-band graphic equalizer uses one prototype low-pass filter, which is a half-band FIR filter designed using the window method. Stretched versions of the prototype filter and its complementary high-pass filter implement all ten band filters needed. The graphic equalizer is realized in the parallel form, in which the outputs of all band filters, scaled with their individual command gain, are added to compute the equalized output signal. The command gains can be used directly as filter band gains. The number of operations needed per sample is only slightly more than that needed for the graphic equalizer based on minimum-phase recursive filters. A comparison with other implementation approaches demonstrates that the proposed structure requires 99% fewer operations than a high-order FIR filter. The proposed filter uses 39% fewer operations per sample than the fast Fourier transform–based filtering method and causes over 78% less latency.
{"title":"Linear-Phase Octave Graphic Equalizer","authors":"V. Bruschi, V. Välimäki, Juho Liski, S. Cecchi","doi":"10.17743/jaes.2022.0014","DOIUrl":"https://doi.org/10.17743/jaes.2022.0014","url":null,"abstract":"A computationally efficient octave-band graphic equalizer having a linear-phase response is introduced. The linear-phase graphic equalizer is useful in audio applications in which phase distortion is not tolerated, such as in multichannel equalization, parallel processing, phase compatibility of audio equipment, and crossover network design. The structure is based on the interpolated finite impulse response (IFIR) philosophy. The proposed octave-band graphic equalizer uses one prototype low-pass filter, which is a half-band FIR filter designed using the window method. Stretched versions of the prototype filter and its complementary high-pass filter implement all ten band filters needed. The graphic equalizer is realized in the parallel form, in which the outputs of all band filters, scaled with their individual command gain, are added to compute the equalized output signal. The command gains can be used directly as filter band gains. The number of operations needed per sample is only slightly more than that needed for the graphic equalizer based on minimum-phase recursive filters. A comparison with other implementation approaches demonstrates that the proposed structure requires 99% fewer operations than a high-order FIR filter. The proposed filter uses 39% fewer operations per sample than the fast Fourier transform–based filtering method and causes over 78% less latency.","PeriodicalId":50008,"journal":{"name":"Journal of the Audio Engineering Society","volume":" ","pages":""},"PeriodicalIF":1.4,"publicationDate":"2022-07-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"42701641","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":4,"RegionCategory":"工程技术","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Real-Time Transient Reduction in Higher-Order Time-Varying Musical Filters","authors":"Nikhil Deshpande, Russell Wedelich","doi":"10.17743/jaes.2022.0015","DOIUrl":"https://doi.org/10.17743/jaes.2022.0015","url":null,"abstract":"","PeriodicalId":50008,"journal":{"name":"Journal of the Audio Engineering Society","volume":" ","pages":""},"PeriodicalIF":1.4,"publicationDate":"2022-07-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"48516390","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":4,"RegionCategory":"工程技术","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
C. Hold, Thomas McKenzie, Georg Götz, Sebastian J. Schlecht, V. Pulkki
Spatial room impulse responses (SRIRs) capture room acoustics with directional information. SRIRs measured in coupled rooms and spaces with non-uniform absorption distribution may exhibit anisotropic reverberation decays and multiple decay slopes. However, noisy measurements with low signal-to-noise ratios pose issues in analysis and reproduction in practice. This paper presents a method for resynthesis of the late decay of anisotropic SRIRs, effectively removing noise from SRIR measurements. The method accounts for both multi-slope decays and directional reverberation. A spherical filter bank extracts directionally constrained signals from Ambisonic input, which are then analyzed and parameterized in terms of multiple exponential decays and a noise floor. The noisy late reverberation is then resynthesized from the estimated parameters using modal synthesis, and the restored SRIR is reconstructed as Ambisonic signals. The method is evaluated both numerically and perceptually, which shows that SRIRs can be denoised with minimal error as long as parts of the decay slope are above the noise level, with signal-to-noise ratios as low as 40 dB in the presented experiment. The method can be used to increase the perceived spatial audio quality of noise-impaired SRIRs.
{"title":"Resynthesis of Spatial Room Impulse Response Tails With Anisotropic Multi-Slope Decays","authors":"C. Hold, Thomas McKenzie, Georg Götz, Sebastian J. Schlecht, V. Pulkki","doi":"10.17743/jaes.2022.0017","DOIUrl":"https://doi.org/10.17743/jaes.2022.0017","url":null,"abstract":"Spatial room impulse responses (SRIRs) capture room acoustics with directional information. SRIRs measured in coupled rooms and spaces with non-uniform absorption distribution may exhibit anisotropic reverberation decays and multiple decay slopes. However, noisy measurements with low signal-to-noise ratios pose issues in analysis and reproduction in practice. This paper presents a method for resynthesis of the late decay of anisotropic SRIRs, effectively removing noise from SRIR measurements. The method accounts for both multi-slope decays and directional reverberation. A spherical filter bank extracts directionally constrained signals from Ambisonic input, which are then analyzed and parameterized in terms of multiple exponential decays and a noise floor. The noisy late reverberation is then resynthesized from the estimated parameters using modal synthesis, and the restored SRIR is reconstructed as Ambisonic signals. The method is evaluated both numerically and perceptually, which shows that SRIRs can be denoised with minimal error as long as parts of the decay slope are above the noise level, with signal-to-noise ratios as low as 40 dB in the presented experiment. The method can be used to increase the perceived spatial audio quality of noise-impaired SRIRs.","PeriodicalId":50008,"journal":{"name":"Journal of the Audio Engineering Society","volume":" ","pages":""},"PeriodicalIF":1.4,"publicationDate":"2022-07-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"45070778","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":4,"RegionCategory":"工程技术","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
京公网安备 11010802042870号
京ICP备2023020795号-1
扫码关注我们
求助内容：
应助结果提醒方式：