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Università della Svizzera italiana
1.
Volontè, Elena.
Subdivision schemes for curve design and image
analysis.
Degree: 2018, Università della Svizzera italiana
URL: http://doc.rero.ch/record/306682 
► Subdivision schemes are able to produce functions, which are smooth up to pixel accuracy, in a few steps through an iterative process. They take as…
(more)
▼ Subdivision schemes are able to produce functions,
which are smooth up to pixel accuracy, in a few steps through an
iterative process. They take as input a coarse control polygon and
iteratively generate new points using some algebraic or geometric
rules. Therefore, they are a powerful tool for creating and
displaying functions, in particular in computer graphics,
computer-aided design, and signal analysis. A lot of research on
univariate subdivision schemes is concerned with the convergence
and the smoothness of the limit curve, especially for schemes where
the new points are a linear combination of points from the previous
iteration. Much less is known for non-linear schemes: in many cases
there are only ad hoc proofs or numerical evidence about the
regularity of these schemes. For schemes that use a geometric
construction, it could be interesting to study the continuity of
geometric entities. Dyn and Hormann propose sufficient conditions
such that the subdivision process converges and the limit curve is
tangent continuous. These conditions can be satisfied by any
interpolatory scheme and they depend only on edge lengths and
angles. The goal of my work is to generalize these conditions and
to find a sufficient constraint, which guarantees that a generic
interpolatory subdivision scheme gives limit curves with continuous
curvature. To require the continuity of the curvature it seems
natural to come up with a condition that depends on the difference
of curvatures of neighbouring circles. The proof of the proposed
condition is not completed, but we give a numerical evidence of it.
A key feature of subdivision schemes is that they can be used in
different fields of approximation theory. Due to their well-known
relation with multiresolution analysis they can be exploited also
in image analysis. In fact, subdivision schemes allow for an
efficient computation of the wavelet transform using the
filterbank. One current issue in signal processing is the analysis
of anisotropic signals. Shearlet transforms allow to do it using
the concept of multiple subdivision schemes. One drawback, however,
is the big number of filters needed for analysing the signal given.
The number of filters is related to the determinant of the
expanding matrix considered. Therefore, a part of my work is
devoted to find expanding matrices that give a smaller number of
filters compared to the shearlet case. We present a family of
anisotropic matrices for any dimension d with smaller determinant
than shearlets. At the same time, these matrices allow for the
definition of a valid directional transform and associated multiple
subdivision schemes.
Advisors/Committee Members: Kai (Dir.), Milvia (Codir.).
Subjects/Keywords: Filterbanks
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APA (6th Edition):
Volontè, E. (2018). Subdivision schemes for curve design and image
analysis. (Thesis). Università della Svizzera italiana. Retrieved from http://doc.rero.ch/record/306682
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
Chicago Manual of Style (16th Edition):
Volontè, Elena. “Subdivision schemes for curve design and image
analysis.” 2018. Thesis, Università della Svizzera italiana. Accessed January 21, 2021.
http://doc.rero.ch/record/306682.
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
MLA Handbook (7th Edition):
Volontè, Elena. “Subdivision schemes for curve design and image
analysis.” 2018. Web. 21 Jan 2021.
Vancouver:
Volontè E. Subdivision schemes for curve design and image
analysis. [Internet] [Thesis]. Università della Svizzera italiana; 2018. [cited 2021 Jan 21].
Available from: http://doc.rero.ch/record/306682.
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
Council of Science Editors:
Volontè E. Subdivision schemes for curve design and image
analysis. [Thesis]. Università della Svizzera italiana; 2018. Available from: http://doc.rero.ch/record/306682
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
Rochester Institute of Technology
2.
Cwitkowitz, Frank C, Jr.
End-to-End Music Transcription Using Fine-Tuned Variable-Q Filterbanks.
Degree: MS, Computer Engineering, 2019, Rochester Institute of Technology
URL: https://scholarworks.rit.edu/theses/10143
► The standard time-frequency representations calculated to serve as features for musical audio may have reached the extent of their effectiveness. General-purpose features such as…
(more)
▼ The standard time-frequency representations calculated to serve as features for musical audio may have reached the extent of their effectiveness. General-purpose features such as Mel-Frequency Spectral Coefficients or the Constant-Q Transform, while being pyschoacoustically and musically motivated, may not be optimal for all tasks. As large, comprehensive, and well-annotated musical datasets become increasingly available, the viability of learning from the raw waveform of recordings widens. Deep neural networks have been shown to perform feature extraction and classification jointly. With sufficient data, optimal filters which operate in the time-domain may be learned in place of conventional time-frequency calculations. Since the spectrum of problems studied by the Music Information Retrieval community are vastly different, rather than relying on the fixed frequency support of each bandpass filter within standard transforms, learned time-domain filters may prioritize certain harmonic frequencies and model note behavior differently based on a specific music task. In this work, the time-frequency calculation step of a baseline transcription architecture is replaced with a learned equivalent, initialized with the frequency response of a Variable-Q Transform. The learned replacement is fine-tuned jointly with a baseline architecture for the task of piano transcription, and the resulting
filterbanks are visualized and evaluated against the standard transform.
Advisors/Committee Members: Andres Kwasinski.
Subjects/Keywords: Features; Filterbanks; Learning; Music; Transcription; Waveform
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APA (6th Edition):
Cwitkowitz, Frank C, J. (2019). End-to-End Music Transcription Using Fine-Tuned Variable-Q Filterbanks. (Masters Thesis). Rochester Institute of Technology. Retrieved from https://scholarworks.rit.edu/theses/10143
Chicago Manual of Style (16th Edition):
Cwitkowitz, Frank C, Jr. “End-to-End Music Transcription Using Fine-Tuned Variable-Q Filterbanks.” 2019. Masters Thesis, Rochester Institute of Technology. Accessed January 21, 2021.
https://scholarworks.rit.edu/theses/10143.
MLA Handbook (7th Edition):
Cwitkowitz, Frank C, Jr. “End-to-End Music Transcription Using Fine-Tuned Variable-Q Filterbanks.” 2019. Web. 21 Jan 2021.
Vancouver:
Cwitkowitz, Frank C J. End-to-End Music Transcription Using Fine-Tuned Variable-Q Filterbanks. [Internet] [Masters thesis]. Rochester Institute of Technology; 2019. [cited 2021 Jan 21].
Available from: https://scholarworks.rit.edu/theses/10143.
Council of Science Editors:
Cwitkowitz, Frank C J. End-to-End Music Transcription Using Fine-Tuned Variable-Q Filterbanks. [Masters Thesis]. Rochester Institute of Technology; 2019. Available from: https://scholarworks.rit.edu/theses/10143
University of Cambridge
3.
Walters, Thomas C.
Auditory-based processing of communication sounds.
Degree: PhD, 2011, University of Cambridge
URL: http://www.dspace.cam.ac.uk/handle/1810/240577https://www.repository.cam.ac.uk/bitstream/1810/240577/2/license.txt
;
https://www.repository.cam.ac.uk/bitstream/1810/240577/3/Thomas_Walters_PhD_Thesis.pdf.txt
;
https://www.repository.cam.ac.uk/bitstream/1810/240577/4/Thomas_Walters_PhD_Thesis.pdf.jpg
► This thesis examines the possible benefits of adapting a biologically-inspired model of human auditory processing as part of a machine-hearing system. Features were generated by…
(more)
▼ This thesis examines the possible benefits of adapting a biologically-inspired model of human auditory processing as part of a machine-hearing system. Features were generated by an auditory model, and used as input to machine learning systems to determine the content of the sound. Features were generated using the auditory image model (AIM) and were used for speech recognition and audio search. AIM comprises processing to simulate the human cochlea, and a ‘strobed temporal integration’ process which generates a stabilised auditory image (SAI) from the input sound.
The communication sounds which are produced by humans, other animals, and many musical instruments take the form of a pulse-resonance signal: pulses excite resonances in the body, and the resonance following each pulse contains information both about the type of object producing the sound and its size. In the case of humans, vocal tract length (VTL) determines the size properties of the resonance. In the speech recognition experiments, an auditory filterbank was combined with a Gaussian fitting procedure to produce features which are invariant to changes in speaker VTL. These features were compared against standard mel-frequency cepstral coefficients (MFCCs) in a size-invariant syllable recognition task. The VTL-invariant representation was found to produce better results than MFCCs when the system was trained on syllables from simulated talkers of one range of VTLs and tested on those from simulated talkers with a different range of VTLs.
The image stabilisation process of strobed temporal integration was analysed. Based on the properties of the auditory filterbank being used, theoretical constraints were placed on the properties of the dynamic thresholding function used to perform strobe detection. These constraints were used to specify a simple, yet robust, strobe detection algorithm. The syllable recognition system described above was then extended to produce features from profiles of the SAI and tested with the same syllable database as before. For clean speech, performance of the features was comparable to that of those generated from the filterbank output. However when pink noise was added to the stimuli, performance dropped more slowly as a function of signal-to-noise ratio when using the SAI-based AIM features, than when using either the filterbank-based features or the MFCCs, demonstrating the noise-robustness properties of the SAI representation.
The properties of the auditory filterbank in AIM were also analysed. Three models of the cochlea were considered: the static gammatone filterbank, dynamic compressive gammachirp (dcGC) and the pole-zero filter cascade (PZFC). The dcGC and gammatone are standard filterbank models, whereas the PZFC is a filter cascade, which more accurately models signal propagation in the cochlea. However, while the architecture of the filterbanks is different, they have all been successfully fitted to psychophysical masking data from humans. The abilities of the filterbanks to measure pitch strength were…
Subjects/Keywords: Auditory modelling; Speech recognition; Machine hearing; Auditory filterbanks; Audio analysis; Content-based analysis; Machine perception
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APA ·
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APA (6th Edition):
Walters, T. C. (2011). Auditory-based processing of communication sounds. (Doctoral Dissertation). University of Cambridge. Retrieved from http://www.dspace.cam.ac.uk/handle/1810/240577https://www.repository.cam.ac.uk/bitstream/1810/240577/2/license.txt ; https://www.repository.cam.ac.uk/bitstream/1810/240577/3/Thomas_Walters_PhD_Thesis.pdf.txt ; https://www.repository.cam.ac.uk/bitstream/1810/240577/4/Thomas_Walters_PhD_Thesis.pdf.jpg
Chicago Manual of Style (16th Edition):
Walters, Thomas C. “Auditory-based processing of communication sounds.” 2011. Doctoral Dissertation, University of Cambridge. Accessed January 21, 2021.
http://www.dspace.cam.ac.uk/handle/1810/240577https://www.repository.cam.ac.uk/bitstream/1810/240577/2/license.txt ; https://www.repository.cam.ac.uk/bitstream/1810/240577/3/Thomas_Walters_PhD_Thesis.pdf.txt ; https://www.repository.cam.ac.uk/bitstream/1810/240577/4/Thomas_Walters_PhD_Thesis.pdf.jpg.
MLA Handbook (7th Edition):
Walters, Thomas C. “Auditory-based processing of communication sounds.” 2011. Web. 21 Jan 2021.
Vancouver:
Walters TC. Auditory-based processing of communication sounds. [Internet] [Doctoral dissertation]. University of Cambridge; 2011. [cited 2021 Jan 21].
Available from: http://www.dspace.cam.ac.uk/handle/1810/240577https://www.repository.cam.ac.uk/bitstream/1810/240577/2/license.txt ; https://www.repository.cam.ac.uk/bitstream/1810/240577/3/Thomas_Walters_PhD_Thesis.pdf.txt ; https://www.repository.cam.ac.uk/bitstream/1810/240577/4/Thomas_Walters_PhD_Thesis.pdf.jpg.
Council of Science Editors:
Walters TC. Auditory-based processing of communication sounds. [Doctoral Dissertation]. University of Cambridge; 2011. Available from: http://www.dspace.cam.ac.uk/handle/1810/240577https://www.repository.cam.ac.uk/bitstream/1810/240577/2/license.txt ; https://www.repository.cam.ac.uk/bitstream/1810/240577/3/Thomas_Walters_PhD_Thesis.pdf.txt ; https://www.repository.cam.ac.uk/bitstream/1810/240577/4/Thomas_Walters_PhD_Thesis.pdf.jpg
University of Cambridge
4.
Walters, Thomas C.
Auditory-based processing of communication sounds.
Degree: PhD, 2011, University of Cambridge
URL: https://doi.org/10.17863/CAM.16336
;
https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.541898
► This thesis examines the possible benefits of adapting a biologically-inspired model of human auditory processing as part of a machine-hearing system. Features were generated by…
(more)
▼ This thesis examines the possible benefits of adapting a biologically-inspired model of human auditory processing as part of a machine-hearing system. Features were generated by an auditory model, and used as input to machine learning systems to determine the content of the sound. Features were generated using the auditory image model (AIM) and were used for speech recognition and audio search. AIM comprises processing to simulate the human cochlea, and a 'strobed temporal integration' process which generates a stabilised auditory image (SAI) from the input sound. The communication sounds which are produced by humans, other animals, and many musical instruments take the form of a pulse-resonance signal: pulses excite resonances in the body, and the resonance following each pulse contains information both about the type of object producing the sound and its size. In the case of humans, vocal tract length (VTL) determines the size properties of the resonance. In the speech recognition experiments, an auditory filterbank was combined with a Gaussian fitting procedure to produce features which are invariant to changes in speaker VTL. These features were compared against standard mel-frequency cepstral coefficients (MFCCs) in a size-invariant syllable recognition task. The VTL-invariant representation was found to produce better results than MFCCs when the system was trained on syllables from simulated talkers of one range of VTLs and tested on those from simulated talkers with a different range of VTLs. The image stabilisation process of strobed temporal integration was analysed. Based on the properties of the auditory filterbank being used, theoretical constraints were placed on the properties of the dynamic thresholding function used to perform strobe detection. These constraints were used to specify a simple, yet robust, strobe detection algorithm. The syllable recognition system described above was then extended to produce features from profiles of the SAI and tested with the same syllable database as before. For clean speech, performance of the features was comparable to that of those generated from the filterbank output. However when pink noise was added to the stimuli, performance dropped more slowly as a function of signal-to-noise ratio when using the SAI-based AIM features, than when using either the filterbank-based features or the MFCCs, demonstrating the noise-robustness properties of the SAI representation. The properties of the auditory filterbank in AIM were also analysed. Three models of the cochlea were considered: the static gammatone filterbank, dynamic compressive gammachirp (dcGC) and the pole-zero filter cascade (PZFC). The dcGC and gammatone are standard filterbank models, whereas the PZFC is a filter cascade, which more accurately models signal propagation in the cochlea. However, while the architecture of the filterbanks is different, they have all been successfully fitted to psychophysical masking data from humans. The abilities of the filterbanks to measure pitch strength were assessed, using…
Subjects/Keywords: 006.3; Auditory modelling; Speech recognition; Machine hearing; Auditory filterbanks; Audio analysis; Content-based analysis; Machine perception
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❌
APA ·
Chicago ·
MLA ·
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CSE |
Export
to Zotero / EndNote / Reference
Manager
APA (6th Edition):
Walters, T. C. (2011). Auditory-based processing of communication sounds. (Doctoral Dissertation). University of Cambridge. Retrieved from https://doi.org/10.17863/CAM.16336 ; https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.541898
Chicago Manual of Style (16th Edition):
Walters, Thomas C. “Auditory-based processing of communication sounds.” 2011. Doctoral Dissertation, University of Cambridge. Accessed January 21, 2021.
https://doi.org/10.17863/CAM.16336 ; https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.541898.
MLA Handbook (7th Edition):
Walters, Thomas C. “Auditory-based processing of communication sounds.” 2011. Web. 21 Jan 2021.
Vancouver:
Walters TC. Auditory-based processing of communication sounds. [Internet] [Doctoral dissertation]. University of Cambridge; 2011. [cited 2021 Jan 21].
Available from: https://doi.org/10.17863/CAM.16336 ; https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.541898.
Council of Science Editors:
Walters TC. Auditory-based processing of communication sounds. [Doctoral Dissertation]. University of Cambridge; 2011. Available from: https://doi.org/10.17863/CAM.16336 ; https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.541898
University of Victoria
5.
Hampton, Peter John.
Robust order N wavelet filterbanks to perform 2-D numerical integration directly from partial difference or gradient measurements.
Degree: Dept. of Electrical and Computer Engineering, 2010, University of Victoria
URL: http://hdl.handle.net/1828/2855
► In this dissertation, a new method for the numerical integration of two-dimensional partial differences is presented. The approach is based on obtaining an estimate of…
(more)
▼ In this dissertation, a new method for the numerical integration of two-dimensional partial differences is presented. The approach is based on obtaining an estimate of the 2-D Haar wavelet decomposition of the integrated differences by filtering and down-sampling the partial difference measurement data as an intermediate step. Then, this decomposition estimate is synthesized into an estimate of the integrated differences.
The
filterbanks required for estimating this decomposition are derived directly from the 2-D Haar Wavelet Analysis Filterbank. The order of operations of this process is manipulated in a novel way so that gradient or partial difference data can be used as input to the filterbank instead of the image data. The original data can then be obtained from this decomposition estimate using unmodified 2-D Haar Wavelet Synthesis
Filterbanks. This use of the wavelet decomposition brings a reduction in computation complexity to less than 10 operations per pixel of the result.
This dissertation shows that the data used for this algorithm may be calculated partial differences or discretely sampled gradient data measurements. This data set may have any-sized convex area of support as long as it is on a Cartesian grid. The method is stable as a component of a closed loop system as shown by simulations of a recently developed woofer-tweeter adaptive optics control system.
Advisors/Committee Members: Agathoklis, Panajotis (supervisor), Bradley, Colin (supervisor).
Subjects/Keywords: Wavelets; Filterbanks; UVic Subject Index::Sciences and Engineering::Engineering::Electrical engineering
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APA ·
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APA (6th Edition):
Hampton, P. J. (2010). Robust order N wavelet filterbanks to perform 2-D numerical integration directly from partial difference or gradient measurements. (Thesis). University of Victoria. Retrieved from http://hdl.handle.net/1828/2855
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
Chicago Manual of Style (16th Edition):
Hampton, Peter John. “Robust order N wavelet filterbanks to perform 2-D numerical integration directly from partial difference or gradient measurements.” 2010. Thesis, University of Victoria. Accessed January 21, 2021.
http://hdl.handle.net/1828/2855.
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
MLA Handbook (7th Edition):
Hampton, Peter John. “Robust order N wavelet filterbanks to perform 2-D numerical integration directly from partial difference or gradient measurements.” 2010. Web. 21 Jan 2021.
Vancouver:
Hampton PJ. Robust order N wavelet filterbanks to perform 2-D numerical integration directly from partial difference or gradient measurements. [Internet] [Thesis]. University of Victoria; 2010. [cited 2021 Jan 21].
Available from: http://hdl.handle.net/1828/2855.
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
Council of Science Editors:
Hampton PJ. Robust order N wavelet filterbanks to perform 2-D numerical integration directly from partial difference or gradient measurements. [Thesis]. University of Victoria; 2010. Available from: http://hdl.handle.net/1828/2855
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
Brno University of Technology
6.
Botlová, Simona.
Komprese EKG signálů: ECG signals compression.
Degree: 2019, Brno University of Technology
URL: http://hdl.handle.net/11012/33165
► This bachelor thesis deals with the compression of ECG signals using wavelet transform and run length encoding. The principles of electrocardiography, compression methods, wavelet transform…
(more)
▼ This bachelor thesis deals with the compression of ECG signals using wavelet transform and run length encoding. The principles of electrocardiography, compression methods, wavelet transform and run length encoding are described in this thesis. There was created a program to compress and decompress ECG signals in MATLAB interface. The wavelet transform settings and their influence on compress ratio and percentage root mean square difference were tested. Subsequently, the appropriate adjustment of algoritm was found and it was used on CSE database compression.
Advisors/Committee Members: Smital, Lukáš (advisor), Vítek, Martin (referee).
Subjects/Keywords: Elektrokardiografia; EKG signál; kompresia; prúdové kódovanie; vlnková transformácia; prahovanie; vlnka; banky filtrov; stupeň dekompozície; kompresný pomer; PRD; MATLAB.; Electrocardiography; ECG signal; compression; run length encoding; wavelet transform; tresholding; wavelet; filterbanks; stage of decomposition; compress ratio; PRD; MATLAB.
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APA ·
Chicago ·
MLA ·
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CSE |
Export
to Zotero / EndNote / Reference
Manager
APA (6th Edition):
Botlová, S. (2019). Komprese EKG signálů: ECG signals compression. (Thesis). Brno University of Technology. Retrieved from http://hdl.handle.net/11012/33165
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
Chicago Manual of Style (16th Edition):
Botlová, Simona. “Komprese EKG signálů: ECG signals compression.” 2019. Thesis, Brno University of Technology. Accessed January 21, 2021.
http://hdl.handle.net/11012/33165.
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
MLA Handbook (7th Edition):
Botlová, Simona. “Komprese EKG signálů: ECG signals compression.” 2019. Web. 21 Jan 2021.
Vancouver:
Botlová S. Komprese EKG signálů: ECG signals compression. [Internet] [Thesis]. Brno University of Technology; 2019. [cited 2021 Jan 21].
Available from: http://hdl.handle.net/11012/33165.
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
Council of Science Editors:
Botlová S. Komprese EKG signálů: ECG signals compression. [Thesis]. Brno University of Technology; 2019. Available from: http://hdl.handle.net/11012/33165
Note: this citation may be lacking information needed for this citation format:
Not specified: Masters Thesis or Doctoral Dissertation
Indian Institute of Science
7.
Shenoy, Ravi R.
Spectral And Temporal Zero-Crossings-Based Signal Analysis.
Degree: PhD, Faculty of Engineering, 2017, Indian Institute of Science
URL: http://etd.iisc.ac.in/handle/2005/2660
► We consider real zero-crossing analysis of the real/imaginary parts of the spectrum, namely, spectral zero-crossings (SZCs). The two major contributions are to show that: (i)…
(more)
▼ We consider real zero-crossing analysis of the real/imaginary parts of the spectrum, namely, spectral zero-crossings (SZCs). The two major contributions are to show that: (i) SZCs provide enable temporal localization of transients; and (b) SZCs are suitable for modeling transient signals. We develop a spectral dual of Kedem’s result linking temporal zero-crossing rate (ZCR) to the spectral centroid. The key requirement is stationarity, which we achieve through random-phase modulations of the time-domain signal. Transient signals are not amenable to modelling in the time domain since they are bursts of energy localized in time and lack structure. We show that the spectrum of transient signals have a rich modulation structure, which leads to an amplitude-modulation – frequency-modulation (AM-FM) model of the spectrum.
We generalize Kedem’s arc-cosine formula for lags greater than one. For the specific case of a sinusoid in white Gaussian noise, He and Kedem devised an iterative filtering algorithm, which leads to a contraction mapping. An autoregressive filter of order one is employed and the location of the pole is the parameter that is updated based on the filtered output. We use the higher-order property, which relates the autocorrelation to the expected ZCR of the filtered process, between lagged ZCR and higher-lag autocorrelation to develop an iterative higher-order autoregressive-filtering scheme, which stabilizes the ZCR and consequently provides robust estimates of the autocorrelation at higher lags.
Next, we investigate ZC properties of critically sampled outputs of a maximally decimated M-channel power complementary analysis filterbank (PCAF) and derive the relationship between the ZCR of the input Gaussian process at lags that are integer multiples of M in terms of the subband ZCRs. Based on this result, we propose a robust autocorrelation estimator for a signal consisting of a sum of sinusoids of fixed amplitudes and uniformly distributed random phases. Robust subband ZCRs are obtained through iterative filtering and the subband variances are estimated using the method-of-moments estimator. We compare the performance of the proposed estimator with the sample auto-correlation estimate in terms of bias, variance, and mean-squared error, and show through simulations that the performance of the proposed estimator is better than the sample auto- correlation for medium to low SNR.
We then consider the ZC statistics of the real/imaginary parts of the discrete Fourier spectrum. We introduce the notion of the spectral zero-crossing rate (SZCR) and show that, for transients, it gives information regarding the location of the transient. We also demonstrate the utility of SZCR to estimate interaural time delay between the left and right head-related impulse responses. The accuracy of interaural time delay plays a vital role in binaural synthesis and a comparison of the performance of the SZCR estimates with that of the cross-correlation estimates illustrate that spectral zeros alone contain enough information for…
Advisors/Committee Members: Patwardhan, Pushkar Prasad (advisor).
Subjects/Keywords: Spectral Signal Analysis; Temporal Zero-Crossings; Spectral Zero-Crossing Signal Analysis; Temporal Zero-Crossing Signal Analysis; Signal Processing; Spectral Analysis; Temporal Analysis; Spectral-Envelop-Group Delay Models; Transients; Filterbanks; Spectral Zero-Crossing Rate; Electrical Engineering
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MLA ·
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APA (6th Edition):
Shenoy, R. R. (2017). Spectral And Temporal Zero-Crossings-Based Signal Analysis. (Doctoral Dissertation). Indian Institute of Science. Retrieved from http://etd.iisc.ac.in/handle/2005/2660
Chicago Manual of Style (16th Edition):
Shenoy, Ravi R. “Spectral And Temporal Zero-Crossings-Based Signal Analysis.” 2017. Doctoral Dissertation, Indian Institute of Science. Accessed January 21, 2021.
http://etd.iisc.ac.in/handle/2005/2660.
MLA Handbook (7th Edition):
Shenoy, Ravi R. “Spectral And Temporal Zero-Crossings-Based Signal Analysis.” 2017. Web. 21 Jan 2021.
Vancouver:
Shenoy RR. Spectral And Temporal Zero-Crossings-Based Signal Analysis. [Internet] [Doctoral dissertation]. Indian Institute of Science; 2017. [cited 2021 Jan 21].
Available from: http://etd.iisc.ac.in/handle/2005/2660.
Council of Science Editors:
Shenoy RR. Spectral And Temporal Zero-Crossings-Based Signal Analysis. [Doctoral Dissertation]. Indian Institute of Science; 2017. Available from: http://etd.iisc.ac.in/handle/2005/2660
. 
Open Access Theses and Dissertations