Musical genres are categorical labels created by humans to characterize pieces of music. A musical genre is characterized by the common characteristics shared by its members. These characteristics typically are related to the instrumentation, rhythmic structure, and harmonic content of the music. Genre hierarchies are commonly used to structure the large collections of music available on the Web. Currently musical genre annotation is performed manually. Automatic musical genre classification can assist or replace the human user in this process and would be a valuable addition to music information retrieval systems. In addition, automatic musical genre classification provides a framework for developing and evaluating features for any type of content-based analysis of musical signals. In this paper, the automatic classification of audio signals into an hierarchy of musical genres is explored. More specifically, three feature sets for representing timbral texture, rhythmic content and pitch content are proposed. The performance and relative importance of the proposed features is investigated by training statistical pattern recognition classifiers using real-world audio collections. Both whole file and real-time frame-based classification schemes are described. Using the proposed feature sets, classification of 61% for ten musical genres is achieved. This result is comparable to results reported for human musical genre classification.

/pdf/musical-genre-classification-of-audio-signals-4c1fbmnwe0.pdf

Musical genre classification of audio signals

Digital audio, video, images, and documents are flying through cyberspace to their respective owners. Unfortunately, along the way, individuals may choose to intervene and take this content for themselves. Digital watermarking and steganography technology greatly reduces the instances of this by limiting or eliminating the ability of third parties to decipher the content that he has taken. The many techiniques of digital watermarking (embedding a code) and steganography (hiding information) continue to evolve as applications that necessitate them do the same. The authors of this second edition provide an update on the framework for applying these techniques that they provided researchers and professionals in the first well-received edition. Steganography and steganalysis (the art of detecting hidden information) have been added to a robust treatment of digital watermarking, as many in each field research and deal with the other. New material includes watermarking with side information, QIM, and dirty-paper codes. The revision and inclusion of new material by these influential authors has created a must-own book for anyone in this profession.

*This new edition now contains essential information on steganalysis and steganography
*New concepts and new applications including QIM introduced
*Digital watermark embedding is given a complete update with new processes and applications

Digital Watermarking and Steganography

https://wiki.inf.ed.ac.uk/twiki/pub/CSTR/Speak08To09/STRAIGHT-original.pdf

Restructuring speech representations using a pitch-adaptive time-frequency smoothing and an instantaneous-frequency-based F0 extraction: possible role of a repetitive structure in sounds

Voice conversion, as considered in this paper, is defined as modifying the speech signal of one speaker (source speaker) so that it sounds as if it had been pronounced by a different speaker (target speaker). Our contribution includes the design of a new methodology for representing the relationship between two sets of spectral envelopes. The proposed method is based on the use of a Gaussian mixture model of the source speaker spectral envelopes. The conversion itself is represented by a continuous parametric function which takes into account the probabilistic classification provided by the mixture model. The parameters of the conversion function are estimated by least squares optimization on the training data. This conversion method is implemented in the context of the HNM (harmonic+noise model) system, which allows high-quality modifications of speech signals. Compared to earlier methods based on vector quantization, the proposed conversion scheme results in a much better match between the converted envelopes and the target envelopes. Evaluation by objective tests and formal listening tests shows that the proposed transform greatly improves the quality and naturalness of the converted speech signals compared with previous proposed conversion methods.

Continuous probabilistic transform for voice conversion

There has been a great deal of material in the area of discrete-time transforms that has been published in recent years. This book does an excellent job of presenting important aspects of such material in a clear manner. The book has 11 chapters and a very useful appendix. Seven of these chapters are essentially devoted to the Fourier series/transform, discrete Fourier transform, fast Fourier transform (FFT), and applications of the FFT in the area of spectral estimation. Chapters 8 through 10 deal with many other discrete-time transforms and algorithms to compute them. Of these transforms, the KarhunenLoeve, the discrete cosine, and the Walsh-Hadamard transform are perhaps the most well-known. A lucid discussion of number theoretic transforms i5 presented in Chapter 11. This reviewer feels that the authors have done a fine job of compiling the pertinent material and presenting it in a concise and clear manner. There are a number of problems at the end of each chapter, an appreciable number of which are challenging. The authors have included a comprehensive set of references at the end of the book. In brief, this book is a valuable contribution to the general areas of signal processing and communications. It can be used for a graduate level course in perhaps two ways. One would be to cover the first seven chapters in great detail. The other would be to cover the whole book by focussing on different topics in a selective manner. This book  by  Elliott  and Rao  is extremely  useful to researchers/engineers who are working  in the areas of signal processing and communications. It i s also an excellent reference book, and hence a valuable addition to one’s library

http://ieeexplore.ieee.org/iel5/5/31347/01457444.pdf

Multirate digital signal processing

Non-parametric techniques for pitch-scale and time-scale modification of speech

The phase vocoder is a well established tool for time scaling and pitch shifting speech and audio signals via modification of their short-time Fourier transforms (STFTs). In contrast to time-domain time-scaling and pitch-shifting techniques, the phase vocoder is generally considered to yield high quality results, especially for large modification factors and/or polyphonic signals. However, the phase vocoder is also known for introducing a characteristic perceptual artifact, often described as "phasiness", "reverberation", or "loss of presence". This paper examines the problem of phasiness in the context of time-scale modification and provides new insights into its causes. Two extensions to the standard phase vocoder algorithm are introduced, and the resulting sound quality is shown to be significantly improved. Moreover, the modified phase vocoder is shown to provide a factor-of-two decrease in computational cost.

/pdf/improved-phase-vocoder-time-scale-modification-of-audio-58unmeo4xv.pdf

Improved phase vocoder time-scale modification of audio

HNS (harmonic plus noise synthesis), an analysis/modification/synthesis model based on a harmonic plus noise representation of the speech signal, is presented. Significant improvements over previous work on the subject are proposed at both the analysis and the synthesis stages: a model of harmonically related sinusoids with linearly varying complex amplitudes for the representation of the deterministic part of the signal; a joint time-domain and frequency-domain representations of the stochastic part of the signal; and a pitch-synchronous PSOLA (pitch-synchronized overlap-add)-like synthesis scheme. Informal listening-tests demonstrate the effectiveness of this approach for time-scale modifications. >

HNS: Speech modification based on a harmonic+noise model

A fingerprint of an audio signal is generated based on the energy content in frequency subbands. Processing techniques assure a robust identification fingerprint that will be useful for signals altered subsequent to the generation of the fingerprint. The fingerprint is compared to a database to identify the audio signal.

Process for identifying audio content

The phase-vocoder is usually presented as a high-quality solution for time-scale modification of signals, pitch-scale modifications usually being implemented as a combination of timescaling and sampling rate conversion. We present two new phase-vocoder-based techniques which allow direct manipulation of the signal in the frequency-domain, enabling such applications as pitch-shifting, chorusing, harmonizing, partial stretching and other exotic modifications which cannot be achieved by the standard time-scale sampling-rate conversion scheme. The new techniques are based on a very simple peak-detection stage, followed by a peak-shifting stage. The very simplest one allows for 50% overlap but restricts the precision of the modifications, while the most flexible techniques requires a more expensive 75% overlap.

/pdf/new-phase-vocoder-techniques-for-pitch-shifting-harmonizing-393viqaajs.pdf

Jean Laroche

Papers

Non-parametric techniques for pitch-scale and time-scale modification of speech

Improved phase vocoder time-scale modification of audio

HNS: Speech modification based on a harmonic+noise model

Process for identifying audio content

New phase-vocoder techniques for pitch-shifting, harmonizing and other exotic effects