With the advent of `multimedia', digital signal processing (DSP) of sound has emerged from the shadow of bandwidth limited speech processing to become a research field of its own. To date, most research in DSP applied to sound has been concentrated on speech, which is bandwidth limited to about 4 kilohertz. Speech processing is also limited by the low fidelity typically expected in the telephone network. Today, the main applications of audio DSP are high quality audio coding and the digital generation and manipulation of music signals. They share common research topics including perceptual measurement techniques and analysis/synthesis methods. Additional important topics are hearing aids using signal processing technology and hardware architectures for digital signal processing of audio. In all these areas the last decade has seen a significant amount of application-oriented research. The frequency range of wideband audio has an upper limit of 20 kilohertz and the resulting difference in frequency range and Signal to Noise Ratio (SNR) due to sample size must be taken into account when designing DSP algorithms. There are whole classes of algorithms that the speech community is not interested in pursuing or using. These algorithms and techniques are revealed in this book. This book is suitable for advanced level courses and serves as a valuable reference for researchers in the field. Interested and informed engineers will also find the book useful in their work.

Applications of digital signal processing to audio and acoustics

Encoding techniques and devices are based on a sinusoidal speech representation model. In one aspect of the invention, a pitch-adaptive channel encoding technique for amplitude coding varies the channel spacing in accordance with the pitch of the speaker's voice. In another aspect of the invention, a phase synthesis technique locks rapidly-varying phases into synchrony with the phase of the fundamental. Phase coding techniques which introduce a voice-dependent random phase and a pitch-adaptive quadratic phase dispersion are also performed.

Coding of acoustic waveforms

A harmonic speech coding arrangement where vector quantization is used to improve speech quality Parameters are determined at the analyzer (120) of an illustrative coding arrangement to model the magnitude and phase spectra of the input speech A first codebook of vectors is searched for a vector that closely approximates the difference between the true and estimated magnitude spectra A second codebook of vectors is searched for a vector that closely approximates the difference between the true and the estimated phase spectra Indices and scaling factors for the vectors are communicated to the synthesizer (160) such that scaled vectors can be added into the magnitude and phase spectra for use at the synthesizer in generating speech as a sum of sinusoids

Vector quantization in a harmonic speech coding arrangement

Methods and apparatus for reducing discontinuities between frames of sinusoidally modeled acoustic waveforms, such as speech, which occur when sampling at low frame rates. A Fast Fourier Transform-based overlap-add technique is applied to amplitude, frequency and phase components of sinusoidal waves after frame-to-frame sine wave matching has been performed. Matched sine wave amplitudes and frequencies are linearly interpolated and mid-point phase is estimated such that the mid-frame sine wave is best fit to the most recent half-frame segments of the lagging and leading sine waves. Synthetic mid-frame sine waves are generated using the interpolated amplitude and frequency and estimated phase values. Synthesized acoustic waveforms of high quality from original source waveforms can be produced in sinusoidal analysis/synthesis operations at coding frame rates of 50 Hz and lower. The methods and devices disclosed herein are particularly useful for computationally efficient coding and synthesis of speech waveforms.

Computationally efficient sine wave synthesis for acoustic waveform processing

A system, computer program product, and method provide for reason driven data analysis in a web-based system that is configurable to various types of data and environments. The computer program product provides for navigation of a report tree of reports that are hierarchically organized with respect to reasons derived from an analysis of transactional data in a database. A frame based interface provides for easy and flexible navigation and configuration of the reports and other features. A configuration file is used to rapidly develop and customize the web-based system. Case management functionality is included to allow for management of cases being reviewed using the system.

Webstation: configurable web-based workstation for reason driven data analysis

It has been shown [1] that an analysis/synthesis system based on a sinusoidal representation leads to synthetic speech that is essentially indistinguishable from the original. By exploiting the peak-to-peak correlation of the sine-wave amplitudes [2], a harmonic model for the sine-wave frequencies, and a predictive model for the sine-wave phases [3], it has also been shown that the sine-wave parameters can be coded at 8 kbps. In this paper a new technique is described for coding the sine-wave amplitudes based on the idea of a pitch-adaptive channel vocoder. Using this amplitude-coding strategy and operating at a total bit rate of 4.8 kbps, it was possible to code and transmit enough phase information so that very intelligible, natural sounding speech could be synthesized. This 4.8 kbps system has been implemented in real-time and has achieved a Diagnostic Rhyme Test (DRT) score of 95. At 2.4 kbps no explicit phase information could be coded, but by phase-locking all of the sine waves to the fundamental, by adding a pitch-adaptive quadratic phase, and by adding a voicing dependent random phase to each sine wave, natural sounding synthetic speech could be obtained. This new system is currently being implemented in real-time so that intelligibility tests can be performed.

T. Quatieri

Papers

Multirate sinusoidal transform coding at rates from 2.4 kbps to 8 kbps