Probability Distributions.- Linear Models for Regression.- Linear Models for Classification.- Neural Networks.- Kernel Methods.- Sparse Kernel Machines.- Graphical Models.- Mixture Models and EM.- Approximate Inference.- Sampling Methods.- Continuous Latent Variables.- Sequential Data.- Combining Models.

Pattern Recognition and Machine Learning

The paper discusses the theory behind the dual-tree transform, shows how complex wavelets with good properties can be designed, and illustrates a range of applications in signal and image processing The authors use the complex number symbol C in CWT to avoid confusion with the often-used acronym CWT for the (different) continuous wavelet transform The four fundamentals, intertwined shortcomings of wavelet transform and some solutions are also discussed Several methods for filter design are described for dual-tree CWT that demonstrates with relatively short filters, an effective invertible approximately analytic wavelet transform can indeed be implemented using the dual-tree approach

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The dual-tree complex wavelet transform

Digital processing of speech signals

A fractional delay filter is a device for bandlimited interpolation between samples. It finds applications in numerous fields of signal processing, including communications, array processing, speech processing, and music technology. We present a comprehensive review of FIR and allpass filter design techniques for bandlimited approximation of a fractional digital delay. Emphasis is on simple and efficient methods that are well suited for fast coefficient update or continuous control of the delay value. Various new approaches are proposed and several examples are provided to illustrate the performance of the methods. We also discuss the implementation complexity of the algorithms. We focus on four applications where fractional delay filters are needed: synchronization of digital modems, incommensurate sampling rate conversion, high-resolution pitch prediction, and sound synthesis of musical instruments.

Splitting the unit delay [FIR/all pass filters design]

Digital Coding of Waveforms

Splitting the Unit Delay - Tools for fractional delay filter design

Modern audio techniques, such as audio coding and sound reproduction, emphasize the modeling of auditory perception as one of the cornerstones for system design. A methodology, frequency-warped digital signal processing, is presented in a tutorial paper as a means to design or implement digital signal-processing algorithms directly in a way that is relevant for auditory perception. Several audio applications are considered in which this approach shows advantages when used as a design or implementation tool or as a conceptual framework of design.

Frequency-warped signal processing for audio applications

Frequency-warped signal processing techniques are attractive to many wideband speech and audio applications since they have a clear connection to the frequency resolution of human hearing. A warped version of linear predictive coding (LPC) is studied. The performance of conventional and warped LPC algorithms are compared in a simulated coding system using listening tests and conventional technical measures. The results indicate that the use of warped techniques is beneficial especially in wideband coding and may result in savings of one bit per sample compared to the conventional algorithm while retaining the same subjective quality.

A comparison of warped and conventional linear predictive coding

Phone segmentation in ASR is usually performed indirectly by Viterbi decoding of HMM output. Direct approaches also exist, e.g., blind speech segmentation algorithms. In either case, performance of automatic speech segmentation algorithms is often measured using automated evaluation algorithms and used to optimize a segmentation system’s performance. However, evaluation approaches reported in literature were found to be lacking. Also, we have determined that increases in phone boundary location detection rates are often due to increased over-segmentation levels and not to algorithmic improvements, i.e., by simply adding random boundaries a better hit-rate can be achieved when using current quality measures. Since established measures were found to be insensitive to this type of random boundary insertion, a new R-value quality measure is introduced that indicates how close a segmentation algorithm’s performance is to an ideal point of operation.

/pdf/an-improved-speech-segmentation-quality-measure-the-r-value-ofmghjbl1s.pdf

Unto K. Laine

Papers

Splitting the unit delay [FIR/all pass filters design]

Splitting the Unit Delay - Tools for fractional delay filter design

Frequency-warped signal processing for audio applications

A comparison of warped and conventional linear predictive coding

An Improved Speech Segmentation Quality Measure: the R-value