Robert C. Maher

Fundamental Frequency Estimation of Musical Signals using a two-way Mismatch Procedure

Fundamental frequency (F0) estimation for quasiharmonic signals is an important task in music signal processing. Many previously developed techniques have suffered from unsatisfactory performance due to ambiguous spectra, noise perturbations, wide frequency range, vibrato, and other common artifacts encountered in musical signals. In this paper a new two‐way mismatch (TWM) procedure for estimating F0 is described which may lead to improved results in this area. This computer‐based method uses the quasiharmonic assumption to guide a search for F0 based on the short‐time spectra of an input signal. The estimated F0 is chosen to minimize discrepancies between measured partial frequencies and harmonic frequencies generated by trial values of F0. For each trial F0, mismatches between the harmonics generated and the measured partial frequencies are averaged over a fixed subset of the available partials. A weighting scheme is used to reduce the susceptibility of the procedure to the presence of noise or absence ...

Semi-automatic classification of bird vocalizations using spectral peak tracks.

Automatic off-line classification and recognition of bird vocalizations has been a subject of interest to ornithologists and pattern detection researchers for many years. Several new applications, including bird vocalization classification for aircraft bird strike avoidance, will require real time classification in the presence of noise and other disturbances. The vocalizations of many common bird species can be represented using a sum-of-sinusoids model. An experiment using computer software to perform peak tracking of spectral analysis data demonstrates the usefulness of the sum-of-sinusoids model for rapid automatic recognition of isolated bird syllables. The technique derives a set of spectral features by time-variant analysis of the recorded bird vocalizations, then performs a calculation of the degree to which the derived parameters match a set of stored templates that were determined from a set of reference bird vocalizations. The results of this relatively simple technique are favorable for both clean and noisy recordings.

Modeling and Signal Processing of Acoustic Gunshot Recordings

Audio recordings of gunshots can provide information about the gun location with respect to the microphone(s), the speed and trajectory of the projectile, and in some cases the type of firearm and ammunition. Recordings obtained under carefully controlled conditions can be well-modeled by geometrical acoustics. Special acoustic processing systems for real time gunshot detection and localization are used by the military and law enforcement agencies for sniper detection. Forensic analysis of audio recordings is also used to provide evidence in criminal and civil cases. This paper reviews the distinctive features and limitations of acoustic gunshot analysis using DSP techniques

Audio forensic examination

The field of audio forensics involves many topics familiar to the general audio digital signal processing (DSP) community, such as speech recognition, talker identification, and signal quality enhancement. There is potentially much to be gained by applying modern DSP theory to problems of interest to the forensics community, and this article is written to give the DSP audience some insight into the types of problems and challenges that face practitioners in audio forensic laboratories. However, this article must also present several of the frustrations and pitfalls encountered by signal processing experts when dealing with typical forensic material due to the standards and practices of the legal system.

Overview of Audio Forensics

Audio forensics applies the tools and techniques of audio engineering and digital signal processing to study audio data as part of a legal proceeding or an official investigation of some kind. This chapter summarizes the principal audio forensic tasks, including authentication, enhancement, and interpretation. The chapter explains the relevant procedural and historical background, presents several examples of audio forensic applications, and reviews several important areas for future research and development.

Analytical Expression for Impulse Response Between Two Nodes in 2-D Rectangular Digital Waveguide Mesh

The digital waveguide mesh models multidimensional wave propagation in discrete space and time. The finite difference wave equation used in the mesh is expressed in the space-time domain, which is not straightforward to reveal the analytical expression for the impulse response between a source and a receiver. This letter describes a procedure to derive an analytical impulse response expression between two nodes in a 2-D rectangular digital waveguide mesh based on the wave equation. The analytical expression can provide helpful insights and help verify simulation results for the digital waveguide mesh.

Sub-band coding of audio using recursively indexed quantization

A low-complexity audio data compression technique using sub-band coding (SBC) and a recursively indexed quantizer (RIQ) is proposed. The characteristics of the proposed system are investigated and several relevant implementation issues are described. The objective performance of the proposed system is compared to conventional coding techniques. A real time implementation of the proposed system running on a DSP microprocessor is also discussed. The results indicate that the RIQ-SBC system exhibits objective signal-to-quantization noise ratio 2 to 5 dB higher than other coders of similar computational complexity when processing a variety of wideband audio signals.

Computer Processing of Audio Signals by Exclusion Filters.

A common task in computer music and electroacoustic signal processing is additive mixing of two audio signals. If the two input signals contain discrete spectral components, their sum will typically contain amplitude beating and other interactions between pairs of components with similar frequencies. A new method is described that suppresses spectral interactions during mixing by deriving a time variant “exclusion filter” from the short‐time spectrum of one of the signals in order to prefilter the other signal. This technique allows one of the signals (dominant) to pass through the mixing process with little modification, while the other signal (secondary) is prevented from interaction by attenuation of its conflicting spectral components. The exclusion filter is specified in a flexible manner, which can include such psychoacoustic criteria as critical bands. Audio examples will be presented.

Map Seeking Circuits: a Novel Method of Detecting Auditory Events using Iterative Template Mapping

This paper reports on a new algorithm to detect the presence of a known acoustic signal in an unknown source. The algorithm, map seeking circuits, has been successfully used in the visual domain. The algorithm seeks to find an appropriate transform that will match a stored template to an unknown signal. The algorithm uses superposition to significantly reduce the computational complexity of searching for a given feature in a signal. This results in a linear computational increase rather than an exponential increase as the complexity of the signal increases. The algorithm was tested with a corpus of six instruments. Results varied from 66% for the piano to 94% for the horn

Crossing the Bridge: Taking Audio DSP from the Textbook to the DSP Design Engineer's Bench

A set of seven audio digital signal processing (DSP) lab assignments have been developed for intermediate and advanced students in electrical engineering. The laboratory modules are designed to bridge the gap between a solid understanding of the mathematical principles of DSP and the practical issues that arise when implementing algorithms for real time audio processing. The audio DSP lab assignments assume a prerequisite of a one semester course in discrete-time signals and systems. The exercises include basic delay lines, recirculating delay lines for audio effects (echo, flanging, phasing, artificial reverberation), variable and parametric IIR equalizer filters, real time signal synthesis techniques (pseudo-random noise, wave table lookup methods, and direct computation), FFT and short-time Fourier transform (STFT) methods, and nonlinear and adaptive audio processes