Instantaneous pitch estimation algorithm based on multirate sampling

The paper presents an algorithm for accurate pitch estimation that takes advantage of the sinusoidal model with instantaneous parameters. The algorithm decomposes the signal into subband components, extracts their instantaneous parameters and evaluates period candidate generating function (PCGF). In order to achieve high accuracy for low and high-pitched sounds it is assumed that possible pitch variation range is proportional to current pitch value. The bandwidths of the decomposition filters and length of the analysis frame are scaled for each period candidate by multirate sampling. The algorithm is compared to other widely used pitch extractors on artificial quasiperiodic signals and natural speech. The proposed algorithm shows a remarkable frequency and time resolution for pitch-modulated sounds and performs well both in clean and noisy conditions

Multirate Frequency Transformations: Wideband AM-FM Demodulation with Applications to Signal Processing and Communications

The AM-FM (amplitude & frequency modulation) signal model finds numerous applications in image processing, communications, and speech processing. The traditional approaches towards demodulation of signals in this category are the analytic signal approach, frequency tracking, or the energy operator approach. These approaches however, assume that the amplitude and frequency components are slowly time-varying, e.g., narrowband and incur significant demodulation error in the wideband scenarios. In this thesis, we extend a two-stage approach towards wideband AM-FM demodulation that combines multirate frequency transformations (MFT) enacted through a combination of multirate systems with traditional demodulation techniques, e.g., the Teager-Kasiser energy operator demodulation (ESA) approach to large wideband to narrowband conversion factors. The MFT module comprises of multirate interpolation and heterodyning and converts the wideband AM-FM signal into a narrowband signal, while the demodulation module such as ESA demodulates the narrowband signal into constituent amplitude and frequency components that are then transformed back to yield estimates for the wideband signal. This MFT-ESA approach is then applied to the various problems of: (a) wideband image demodulation and fingerprint demodulation, where multidimensional energy separation is employed, (b) wideband first-formant demodulation in vowels, and (c) wideband CPM demodulation with partial response signaling, to demonstrate its validity in both monocomponent and multicomponent scenarios as an effective multicomponent AM-FM signal demodulation and analysis technique for image processing, speech processing, and communications based applications

Aircraft adaptive learning control

The optimal control theory of stochastic linear systems is discussed in terms of the advantages of distributed-control systems, and the control of randomly-sampled systems. An optimal solution to longitudinal control is derived and applied to the F-8 DFBW aircraft. A randomly-sampled linear process model with additive process and noise is developed

Auditory Spectrum-Based Pitched Instrument Onset Detection

In this paper, a method for onset detection of music signals using auditory spectra is proposed. The auditory spectrogram provides a time-frequency representation that employs a sound processing model resembling the human auditory system. Recent work on onset detection employs DFT-based features describing spectral energy and phase differences, as well as pitch-based features. These features are often combined for maximizing detection performance. Here, the spectral flux and phase slope features are derived in the auditory framework and a novel fundamental frequency estimation algorithm based on auditory spectra is introduced. An onset detection algorithm is proposed, which processes and combines the aforementioned features at the decision level. Experiments are conducted on a dataset covering 11 pitched instrument types, consisting of 1829 onsets in total. Results indicate that auditory representations outperform various state-of-the-art approaches, with the onset detection algorithm reaching an F-measure of 82.6%

A Method for the Design of Multirate Sampled-Data Digital Flight Control Systems of Piloted Aircraft

The initial flight-test operations of piloted aircraft, in which Digital Flight Control (DFC) systems were first employed, exposed handling qualities problems that were not predicted during the design stage. Subsequent studies attributed the cause of these problems to the techniques used in the design of the digital control systems. The particular feature which unites the reported difficulties is that, an infinite-resolution sampled-data model is assumed for the design process but the practical DFC implementation is realised as an amplitude-quantised sampled-data system

ЦИФРОВЫЕ БАНКИ ФИЛЬТРОВ ДЛЯ СОВРЕМЕННЫХ ЗАДАЧ ОБРАБОТКИ ЗВУКОВЫХ СИГНАЛОВ

The paper reviews techniques of digital filter bank synthesis that can be applied for contemporary speech processing challenges. The paper describes practical experience of using digital filter banks in original systems of sound processing, namely, musical player with noise-aware audio enhancement and hearing aid application for a smartphone.В работе выполнен обзор способов синтеза цифровых банков фильтров, которые могут применяться для решения современных прикладных задач обработки звуковых сигналов. Описывается практический опыт использования цифровых банков фильтров в оригинальных системах обработки звука: музыкальном плеере с функцией повышения разборчивости звучания при прослушивании в шумной акустической обстановке, а также слуховом аппарате на базе смартфона

Audio- ja puhesignaalien aika-asteikon muuttaminen

In audio time-scale modification (TSM), the duration of an audio recording is changed while retaining its local frequency content. In this thesis, a novel phase vocoder based technique for TSM was developed, which is based on the new concept of fuzzy classification of points in the time-frequency representation of an input signal. The points in the time-frequency representation are classified into three signal classes: tonalness, noisiness, and transientness. The information from the classification is used to preserve the distinct nature of these components during modification. The quality of the proposed method was evaluated by means of a listening test. The proposed method scored slightly higher than a state-of-the-art academic TSM technique, and similarly as a commercial TSM software. The proposed method is suitable for high-quality TSM of a wide variety of audio and speech signals.Äänen aika-asteikon muuttamisessa äänitteen pituutta muokataan niin, että sen paikallinen taajuussisältö säilyy samanlaisena. Tässä diplomityössä kehitettiin uusi, vaihevokooderiin pohjautuva menetelmä äänen aika-asteikon muuttamiseen. Menetelmä perustuu äänen aikataajuusesityksen pisteiden sumeaan luokitteluun. Pisteet luokitellaan soinnillisiksi, kohinaisiksi ja transienttisiksi määrittämällä jatkuva totuusarvo pisteen kuulumiselle kuhunkin näistä luokista. Sumeasta luokittelusta saatua tietoa käytetään hyväksi näiden erilaisten signaalikomponenttien ominaisuuksien säilyttämiseen aika-asteikon muuttamisessa. Esitellyn menetelmän laatua arvioitiin kuuntelukokeen avulla. Esitelty menetelmä sai kokeessa hieman paremmat pisteet kuin viimeisintä tekniikkaa edustava akateeminen menetelmä, ja samanlaiset pisteet kuin kaupallinen ohjelmisto. Esitelty menetelmä soveltuu monenlaisien musiikki- ja puhesignaalien aika-asteikon muuttamiseen

Analysis and resynthesis of polyphonic music

This thesis examines applications of Digital Signal Processing to the analysis, transformation, and resynthesis of musical audio. First I give an overview of the human perception of music. I then examine in detail the requirements for a system that can analyse, transcribe, process, and resynthesise monaural polyphonic music. I then describe and compare the possible hardware and software platforms. After this I describe a prototype hybrid system that attempts to carry out these tasks using a method based on additive synthesis. Next I present results from its application to a variety of musical examples, and critically assess its performance and limitations. I then address these issues in the design of a second system based on Gabor wavelets. I conclude by summarising the research and outlining suggestions for future developments