A good acoustic environment is absolutely essential to maintaining a high level satisfaction and moral health among residents. Noise and other boresome sounds come from both in- door and outdoor sources. For the resid...A good acoustic environment is absolutely essential to maintaining a high level satisfaction and moral health among residents. Noise and other boresome sounds come from both in- door and outdoor sources. For the residential buildings adjacent to heavy traffic roads, outdoors traffic noise is the main source that affects indoor acoustic quality and health. Ventilation and outdoor noise prevention become a pair of contradictions for the residents in China nowadays for those buildings adjacent to heavy traffic roads. It is investigated that traffic noise emission is mainly con- stituted by the motors of trucks, buses and motorcycles as well as brake. In this paper, two methods of traffic noise reduction on the indoor sound environment and comfort are carried out to study and compare the residential buildings adjacent to heavy traffic roadway in a city. One is to install noise barriers on the two sides of the roadway, which consist of sound-proof glass and plas- tic materials. The effect of sound-insulation of this method is heavily dependent on the relative distance between the noise bar- rier and indoors. A reduction of sound with an average pressure level of 2–15dB is achieved on the places behind and under the noise barrier. However, for the equivalent of noise barrier height, the noise reduction effect is little. As for the places of higher than the noise barrier, the traffic noise will be even strengthened by 3–7dB. Noise increment can be seen at the points of distance farther than 15m and height more than noise barrier; the noise reduction effect is not satisfactory or even worsened. In addition, not every location is appropriate to install the noise barrier along the heavy traffic roads. The other method of noise reduction for the buildings adjacent to heavy traffic is to install the airproof and soundproof windows, which is the conversion from natural venti- lation to mechanical ventilation. A reduction of sound with an average pressure level of 5dB to 17dB can be achieved compared with common glass windows, if adopting sound proof glass win- dows. These two methods are helpful to isolate high frequency noise but not for low frequency noise. For those frequency noises, installing thick and cotton curtain and porous carpet can only decrease 2.4–4.5dB, which hardly contributes to indoor sound comfort, so further study is demanded to cut down traffic noise, especially to cut down the low frequency noise.展开更多
The proposed secure communication approach adopts the proposed algorithm of Analysis-By- Synthesis (ABS) speech information hiding to establish a Secret Speech Subliminai Channel (SSSC) for speech secure communica...The proposed secure communication approach adopts the proposed algorithm of Analysis-By- Synthesis (ABS) speech information hiding to establish a Secret Speech Subliminai Channel (SSSC) for speech secure communication over PSTN (Public Switched Telephone Network), and employs the algorithm of ABS speech information extracting to recovery the secret information, This approach is more reliable, covert and securable than traditional and chaotic secure communication.展开更多
To capture the presence of speech embedded in nonspeech events and background noise in shortwave non-cooperative communication, an algorithm for speech-stream detection in noisy environments is presented based on Empi...To capture the presence of speech embedded in nonspeech events and background noise in shortwave non-cooperative communication, an algorithm for speech-stream detection in noisy environments is presented based on Empirical Mode Decomposition (EMD) and statistical properties of higher-order cumulants of speech signals. With the EMD, the noise signals can be decomposed into different numbers of IMFs. Then, the fourth-order cumulant ( FOC ) can be used to extract the desired feature of statistical properties for IMF components. Since the higher-order eumulants are blind for Gaussian signals, the proposed method is especially effective regarding the problem of speech-stream detection, where the speech signal is distorted by Gaussian noise. With the self-adaptive decomposition by EMD, the proposed method can also work well for non-Gaussian noise. The experiments show that the proposed algorithm can suppress different noise types with different SNRs, and the algorithm is robust in real signal tests.展开更多
In this paper, a free-space vortex channel model of the radio vortex system is proposed to describe the propagation characteristics of vortex signals over the vortex channel. The key idea is to combine the Laguerre-Ga...In this paper, a free-space vortex channel model of the radio vortex system is proposed to describe the propagation characteristics of vortex signals over the vortex channel. The key idea is to combine the Laguerre-Gaussian(LG) modes in the optical field with the free-space propagation model in the radio field. The proposed free-space vortex channel model is derived from the electric field expression of the LG modes and the freespace channel transfer function of the freespace propagation model theoretically. Simulation results verify that the proposed model could reflect the vortex channel characteristics better than the currently used free-space propagation model.展开更多
The purpose of this articleis to address the presence of seven recomposed ricercari by Jacques Buus in the Music Manuscript 242 from the Library of the University of Coimbra in Portugal (P-Cug MM 242). These recompo...The purpose of this articleis to address the presence of seven recomposed ricercari by Jacques Buus in the Music Manuscript 242 from the Library of the University of Coimbra in Portugal (P-Cug MM 242). These recompositions, probably copied in the third quarter of the 16th century, were made after the previously copied Buus's ricercari in Music Manuscript 48 (P-Cug MM 48) of the same library, which were based on the Libro primo de ricercari a quattrovoci, published in Venice in 1547 by Antonio Gardane. In this paper, the author intend to focus in two main aspects of the research done on this subject. The first topic concerns the score-format of both manuscripts 48 & 242, which testify the instrumental activity in mid-sixteenth century Portugal. The author will demonstrate that this format served once, in the Santa Cruz Monastery in Coimbra, as a didactic tool in the teaching of counterpoint through the music of a northern European master such as Jacques Buus. The copies in the manuscripts were never intended to be used as a performing support--they contain many errors of vertical coordination between the voices that make the performance impossible. The second topic focuses on Buus' recomposed ricercari, which were the object of many cuts, brief recomposed bridges, newly inserted sections, and written glosa figurations. Through these recompositions, the author will describe the theoretic assimilation of formal processes, of style, mode, counterpoint and performing practice. The achievement of this paper is to offer historic musicology researchers a new perspective about the enormous influence that Buus' ricercari from his Libro primo.., had in the learning processes of music composition and in the development of didactic and performing practices in the Santa Cruz Monastery in Coimbra, during the mid-sixteenth century.展开更多
The Present work reports the variability of the derived sound channel and its parameters (surface sound velocity, conjugate depth, SLD (Sonic Layer depth) and SOFAR(Sound Fixing and Ranging) depth) has been pres...The Present work reports the variability of the derived sound channel and its parameters (surface sound velocity, conjugate depth, SLD (Sonic Layer depth) and SOFAR(Sound Fixing and Ranging) depth) has been presented over the Bay of Bengal and Arabian Sea. We use World Ocean Atlas Annual data (2013) on temperature; salinity of North Indian Ocean (0°-25°N; 50°-95°E) and its bathymetry have been utilized for the present computation. The depth of the sound channel axis increases towards the northern latitudes in the Arabian Sea, while it decreases in the Bay of Bengal. Coming to the conjugate depth, it shows variation from 120-400 m in the Bay of Bengal and 50-320 m in the Arabian Sea. The range of SLD is varying between 20-40 m in the Bay of Bengal and 10-30 m in the Arabian Sea. The Bay of Bengal and the Arabian Sea have depth limited nature of the profile, i.e. surface sound speed exceeds the near bottom values. This has an important implication in the sound propagation in the SOFAR channel. Anticipated acoustic rays in an ocean with depth limited profile will propagate as surface refracted, bottom reflected (RBR) rays. As a result, the effective sound channel lies much below the sea surface.展开更多
Sound propagation in a deep ocean two-axis underwater channel is often complex and difficult to simulate between surface channel and sound fixing and ranging (SOFAR) channel. The beam-displacement ray-mode (BDRM) theo...Sound propagation in a deep ocean two-axis underwater channel is often complex and difficult to simulate between surface channel and sound fixing and ranging (SOFAR) channel. The beam-displacement ray-mode (BDRM) theory is a normal mode method for propagation modeling in horizontally stratified shallow water. An improved method for computing the upper boundary reflection coefficient in the BDRM is proposed and applied to calculate the acoustic fields of a two-axis underwater channel. Transmission losses in the two-axis underwater channel are calculated in the new BDRM. The corresponding results are in good agreement with those from the Kraken code, and furthermore the computed speed of the new BDRM excels the other methods.展开更多
As an audiovisual medium, computer animations require superior image quality and professional soundtrack to lead audiences into their fascinating virtual world. Without sound, the impact of storytelling is reduced or ...As an audiovisual medium, computer animations require superior image quality and professional soundtrack to lead audiences into their fascinating virtual world. Without sound, the impact of storytelling is reduced or the story is even not under- standable. Despite the importance of sound, most animators are unfamiliar with sound editing software. Limited budget projects such as independent or student works have difficulty hiring sound professionals to create tailor-made soundtrack. Therefore, we need a suitable sound tool to express their individual ideas. In this paper, we propose an approach using schematic for both computer animation and sound. Our approach provides (1) physical simulation on sound through animation parameters and (2) a new channel for animators to add aesthetic values in sound through their experience of using schematics in existing animation software. We demonstrate our idea through several examples, such as Doppler shift, obstacle effect, importance, energetic, and sputtering effect.展开更多
The enhanced variable rate codec (EVRC) is a standard for the 'Speech ServiceOption 3 for Wideband Spread Spectrum Digital System,' which has been employed in both IS-95cellular systems and ANSI J-STC-008 PCS ...The enhanced variable rate codec (EVRC) is a standard for the 'Speech ServiceOption 3 for Wideband Spread Spectrum Digital System,' which has been employed in both IS-95cellular systems and ANSI J-STC-008 PCS (personal communications systems). This paper concentrateson channel decoders that exploit the residual redundancy inherent in the enhanced variable ratecodec bitstream. This residual redundancy is quantified by modeling the parameters as first orderMarkov chains and computing the entropy rate based on the relative frequencies of transitions.Moreover, this residual redundancy can be exploited by an appropriately 'tuned' channel decoder toprovide substantial coding gain when compared with the decoders that do not exploit it. Channelcoding schemes include convolutional codes, and iteratively decoded parallel concatenatedconvolutional 'turbo' codes.展开更多
The article presents a new approach to the tasks of noise assessment and reduction in the urbanized environment endangered by road noise sources. It was proposed to include the acoustic quality model in the currently ...The article presents a new approach to the tasks of noise assessment and reduction in the urbanized environment endangered by road noise sources. It was proposed to include the acoustic quality model in the currently applied quantitative noise assessment in the management of urbanized environment. In particular, this model takes into account subjective features of sound quality, i.e.: loudness, sharpness, roughness, and fluctuation strength as well as noise mmoya^ce assessment obtained in laboratory conditions. The proposed way can be used in estimating investment costs of an acoustic barrier at the design stage展开更多
A novel automatic ultrasonic system used for the inspection of pipeline girth welds is developed, in which a linear phased array transducer using electronic scan is adopted. Optimal array parameters are determined bas...A novel automatic ultrasonic system used for the inspection of pipeline girth welds is developed, in which a linear phased array transducer using electronic scan is adopted. Optimal array parameters are determined based on a mathematical model of acoustic field for linear phased army derived from Huygens' principle. The testing method and the system structure are introduced. The experimental results show that the phased array transducer system has the same detectability as that of conventional ultrasonic transducer system, but the system architecture can be simplified greatly, and the testing flexibility and the testing speed can be improved greatly.展开更多
Multichannel audio signal is more difficult to be compressed than mono and stereo ones.A novel multichannel audio signal compression method based on tensor representation and decomposition is proposed in this paper.Th...Multichannel audio signal is more difficult to be compressed than mono and stereo ones.A novel multichannel audio signal compression method based on tensor representation and decomposition is proposed in this paper.The multichannel audio is represented with 3-order tensor space and is decomposed into core tensor with three factor matrices in the way of channel,time and frequency.Only the truncated core tensor is transmitted which will be multiplied by the pre-trained factor matrices to reconstruct the original tensor space.Objective and subjective experiments have been done to show a very noticeable compression capability with an acceptable output quality.The novelty of the proposed compression method is that it enables both high compression capability and backward compatibility with limited signal distortion to the hearing.展开更多
文摘A good acoustic environment is absolutely essential to maintaining a high level satisfaction and moral health among residents. Noise and other boresome sounds come from both in- door and outdoor sources. For the residential buildings adjacent to heavy traffic roads, outdoors traffic noise is the main source that affects indoor acoustic quality and health. Ventilation and outdoor noise prevention become a pair of contradictions for the residents in China nowadays for those buildings adjacent to heavy traffic roads. It is investigated that traffic noise emission is mainly con- stituted by the motors of trucks, buses and motorcycles as well as brake. In this paper, two methods of traffic noise reduction on the indoor sound environment and comfort are carried out to study and compare the residential buildings adjacent to heavy traffic roadway in a city. One is to install noise barriers on the two sides of the roadway, which consist of sound-proof glass and plas- tic materials. The effect of sound-insulation of this method is heavily dependent on the relative distance between the noise bar- rier and indoors. A reduction of sound with an average pressure level of 2–15dB is achieved on the places behind and under the noise barrier. However, for the equivalent of noise barrier height, the noise reduction effect is little. As for the places of higher than the noise barrier, the traffic noise will be even strengthened by 3–7dB. Noise increment can be seen at the points of distance farther than 15m and height more than noise barrier; the noise reduction effect is not satisfactory or even worsened. In addition, not every location is appropriate to install the noise barrier along the heavy traffic roads. The other method of noise reduction for the buildings adjacent to heavy traffic is to install the airproof and soundproof windows, which is the conversion from natural venti- lation to mechanical ventilation. A reduction of sound with an average pressure level of 5dB to 17dB can be achieved compared with common glass windows, if adopting sound proof glass win- dows. These two methods are helpful to isolate high frequency noise but not for low frequency noise. For those frequency noises, installing thick and cotton curtain and porous carpet can only decrease 2.4–4.5dB, which hardly contributes to indoor sound comfort, so further study is demanded to cut down traffic noise, especially to cut down the low frequency noise.
基金Supported by the National High Technology Research and Development Program of China (863 Program) (No.2003AA142080, 2004AA775060)the National Natural Sicence Foundation of China (No.60203004)+1 种基金with additional support from the China Post-doctorial Research Foundation (2005-03)the Foundation of Tianjin Key Lab for Advanced Signal Processing(2005).
文摘The proposed secure communication approach adopts the proposed algorithm of Analysis-By- Synthesis (ABS) speech information hiding to establish a Secret Speech Subliminai Channel (SSSC) for speech secure communication over PSTN (Public Switched Telephone Network), and employs the algorithm of ABS speech information extracting to recovery the secret information, This approach is more reliable, covert and securable than traditional and chaotic secure communication.
基金Sponsored by the National Natural Science Foundation of China(Grant No.60475016)the Foundational Research Fund of Harbin Engineering University (Grant No.HEUF04092)
文摘To capture the presence of speech embedded in nonspeech events and background noise in shortwave non-cooperative communication, an algorithm for speech-stream detection in noisy environments is presented based on Empirical Mode Decomposition (EMD) and statistical properties of higher-order cumulants of speech signals. With the EMD, the noise signals can be decomposed into different numbers of IMFs. Then, the fourth-order cumulant ( FOC ) can be used to extract the desired feature of statistical properties for IMF components. Since the higher-order eumulants are blind for Gaussian signals, the proposed method is especially effective regarding the problem of speech-stream detection, where the speech signal is distorted by Gaussian noise. With the self-adaptive decomposition by EMD, the proposed method can also work well for non-Gaussian noise. The experiments show that the proposed algorithm can suppress different noise types with different SNRs, and the algorithm is robust in real signal tests.
基金supported in part by National Science Foundation for Distinguished Young Scholars of China with Grant number 61325004Major Program of National Natural Science Foundation of Hubei in China with Grant number 2016CFA009+2 种基金the Fundamental Research Funds for the Central Universities with Grant number 2015ZDTD012the National Natural Science Foundation of China under Grant No.61463035the Research Foundation of the Education Department of Jiangxi Province under Grant No.GJJ150198
文摘In this paper, a free-space vortex channel model of the radio vortex system is proposed to describe the propagation characteristics of vortex signals over the vortex channel. The key idea is to combine the Laguerre-Gaussian(LG) modes in the optical field with the free-space propagation model in the radio field. The proposed free-space vortex channel model is derived from the electric field expression of the LG modes and the freespace channel transfer function of the freespace propagation model theoretically. Simulation results verify that the proposed model could reflect the vortex channel characteristics better than the currently used free-space propagation model.
文摘The purpose of this articleis to address the presence of seven recomposed ricercari by Jacques Buus in the Music Manuscript 242 from the Library of the University of Coimbra in Portugal (P-Cug MM 242). These recompositions, probably copied in the third quarter of the 16th century, were made after the previously copied Buus's ricercari in Music Manuscript 48 (P-Cug MM 48) of the same library, which were based on the Libro primo de ricercari a quattrovoci, published in Venice in 1547 by Antonio Gardane. In this paper, the author intend to focus in two main aspects of the research done on this subject. The first topic concerns the score-format of both manuscripts 48 & 242, which testify the instrumental activity in mid-sixteenth century Portugal. The author will demonstrate that this format served once, in the Santa Cruz Monastery in Coimbra, as a didactic tool in the teaching of counterpoint through the music of a northern European master such as Jacques Buus. The copies in the manuscripts were never intended to be used as a performing support--they contain many errors of vertical coordination between the voices that make the performance impossible. The second topic focuses on Buus' recomposed ricercari, which were the object of many cuts, brief recomposed bridges, newly inserted sections, and written glosa figurations. Through these recompositions, the author will describe the theoretic assimilation of formal processes, of style, mode, counterpoint and performing practice. The achievement of this paper is to offer historic musicology researchers a new perspective about the enormous influence that Buus' ricercari from his Libro primo.., had in the learning processes of music composition and in the development of didactic and performing practices in the Santa Cruz Monastery in Coimbra, during the mid-sixteenth century.
文摘The Present work reports the variability of the derived sound channel and its parameters (surface sound velocity, conjugate depth, SLD (Sonic Layer depth) and SOFAR(Sound Fixing and Ranging) depth) has been presented over the Bay of Bengal and Arabian Sea. We use World Ocean Atlas Annual data (2013) on temperature; salinity of North Indian Ocean (0°-25°N; 50°-95°E) and its bathymetry have been utilized for the present computation. The depth of the sound channel axis increases towards the northern latitudes in the Arabian Sea, while it decreases in the Bay of Bengal. Coming to the conjugate depth, it shows variation from 120-400 m in the Bay of Bengal and 50-320 m in the Arabian Sea. The range of SLD is varying between 20-40 m in the Bay of Bengal and 10-30 m in the Arabian Sea. The Bay of Bengal and the Arabian Sea have depth limited nature of the profile, i.e. surface sound speed exceeds the near bottom values. This has an important implication in the sound propagation in the SOFAR channel. Anticipated acoustic rays in an ocean with depth limited profile will propagate as surface refracted, bottom reflected (RBR) rays. As a result, the effective sound channel lies much below the sea surface.
基金This project was supported by National Defense Research Found (No. 9140A03050206JB1501)
文摘Sound propagation in a deep ocean two-axis underwater channel is often complex and difficult to simulate between surface channel and sound fixing and ranging (SOFAR) channel. The beam-displacement ray-mode (BDRM) theory is a normal mode method for propagation modeling in horizontally stratified shallow water. An improved method for computing the upper boundary reflection coefficient in the BDRM is proposed and applied to calculate the acoustic fields of a two-axis underwater channel. Transmission losses in the two-axis underwater channel are calculated in the new BDRM. The corresponding results are in good agreement with those from the Kraken code, and furthermore the computed speed of the new BDRM excels the other methods.
文摘As an audiovisual medium, computer animations require superior image quality and professional soundtrack to lead audiences into their fascinating virtual world. Without sound, the impact of storytelling is reduced or the story is even not under- standable. Despite the importance of sound, most animators are unfamiliar with sound editing software. Limited budget projects such as independent or student works have difficulty hiring sound professionals to create tailor-made soundtrack. Therefore, we need a suitable sound tool to express their individual ideas. In this paper, we propose an approach using schematic for both computer animation and sound. Our approach provides (1) physical simulation on sound through animation parameters and (2) a new channel for animators to add aesthetic values in sound through their experience of using schematics in existing animation software. We demonstrate our idea through several examples, such as Doppler shift, obstacle effect, importance, energetic, and sputtering effect.
文摘The enhanced variable rate codec (EVRC) is a standard for the 'Speech ServiceOption 3 for Wideband Spread Spectrum Digital System,' which has been employed in both IS-95cellular systems and ANSI J-STC-008 PCS (personal communications systems). This paper concentrateson channel decoders that exploit the residual redundancy inherent in the enhanced variable ratecodec bitstream. This residual redundancy is quantified by modeling the parameters as first orderMarkov chains and computing the entropy rate based on the relative frequencies of transitions.Moreover, this residual redundancy can be exploited by an appropriately 'tuned' channel decoder toprovide substantial coding gain when compared with the decoders that do not exploit it. Channelcoding schemes include convolutional codes, and iteratively decoded parallel concatenatedconvolutional 'turbo' codes.
文摘The article presents a new approach to the tasks of noise assessment and reduction in the urbanized environment endangered by road noise sources. It was proposed to include the acoustic quality model in the currently applied quantitative noise assessment in the management of urbanized environment. In particular, this model takes into account subjective features of sound quality, i.e.: loudness, sharpness, roughness, and fluctuation strength as well as noise mmoya^ce assessment obtained in laboratory conditions. The proposed way can be used in estimating investment costs of an acoustic barrier at the design stage
文摘A novel automatic ultrasonic system used for the inspection of pipeline girth welds is developed, in which a linear phased array transducer using electronic scan is adopted. Optimal array parameters are determined based on a mathematical model of acoustic field for linear phased army derived from Huygens' principle. The testing method and the system structure are introduced. The experimental results show that the phased array transducer system has the same detectability as that of conventional ultrasonic transducer system, but the system architecture can be simplified greatly, and the testing flexibility and the testing speed can be improved greatly.
基金This work was partially supported by the National Natural Science Foundation of China under Grants No.11161140319,No.61001188,the Specialized Research Fund for the Doctoral Program of Higher Education under Grant No.20101101110020,the Fund for Basic Research from Beijing Institute of Technology under Grant No.20120542011,the Fund for Beijing Higher Education Young Elite Teacher Project under Grant No.YETP1202
文摘Multichannel audio signal is more difficult to be compressed than mono and stereo ones.A novel multichannel audio signal compression method based on tensor representation and decomposition is proposed in this paper.The multichannel audio is represented with 3-order tensor space and is decomposed into core tensor with three factor matrices in the way of channel,time and frequency.Only the truncated core tensor is transmitted which will be multiplied by the pre-trained factor matrices to reconstruct the original tensor space.Objective and subjective experiments have been done to show a very noticeable compression capability with an acceptable output quality.The novelty of the proposed compression method is that it enables both high compression capability and backward compatibility with limited signal distortion to the hearing.