Download Digital Emulation of Analog Companding Algorithms for FM-Radio Transmission Analog compander systems have been used to suppress the perception of noise in low dynamic range analog signal storage (tape recording) and signal transmission (FM radio). Commercial compander systems have been analyzed with respect to their signal processing requirements. The general structures of single- and multiband compander systems have been implemented on a high performance audio PC workstation. Audio tests and measurements with the optimized compander algorithms and parameters show very good performance. Even for transmission channels with very low signal-to-noise ratio (SNR of only 40 dB) an optimized digital multi-band compander emulation removes the channel noise perceptively from the output signal of the transmission system.
Download High Quality Voice Transformations Based on Modeling Radiated Voice Pulses in Frequency Domain This paper introduces a method to transform voice based on modeling the radiated voice pulses in frequency domain. This approach tries to combine the strengths of classical time and frequency domain techniques into a single framework, providing both an independent control of each voice pulse and flexible timbre and phase modification capabilities.
Download Adaptive Effects Based on STFT, Using a Source-Filter Model This paper takes the opportunity of presenting a set of new adaptive effects to propose a generic scheme for adaptive effects built upon a spectral source-filter decomposition and a Short-Time Fourier analysis-resynthesis. This allows for a better formalization of the involved signal processing algorithms and leads to a simple classification of adaptive effects already presented in the literature, that falls into this category. We discuss the motivation and the advantages of combining source-filter modeling and phase vocoder representation for the design of adaptive digital audio effects. Then we detail the general structure that includes STFT analysis and re-synthesis scheme, the source filter decomposition, and an adaptive control unit composed of a feature extraction system and a sound mapping unit that might be driven by a gestural control section.
Download Software for Measuring and Improving Esophageal Voices The main aim of this paper is to describe a new software program for esophageal speech treatment developed at the University of Deusto. The software tool, named “ESOIMPROVE”, allows both to characterize and to modify this speech, and provides the necessary framework to achieve a high quality and intelligible transformed esophageal speech by applying a complete range of sound effects and algorithms. In this field, this tool represents a considerable advance in the study of these voices. The final objective of the project is to obtain an esophageal speech with acceptable levels of quality and intelligibility, and some more works in this direction are being actually developed.
Download Improvement of Esophageal Voices' Pitch In this paper it is described a new algorithm for esophageal speech regeneration, based on pitch and jitter modification. Traditional phase vocoder and resampling pitch scaling techniques have been used to develop a new adaptive method which scales the low esophageal speech pitch and applies a variable scaling factor significantly reducing its jitter. This method has shown to considerably improve esophageal speech quality, reducing its hoarseness and increasing its intelligibility. The presented algorithm pretends to be an important step forward in the regeneration of esophageal speech.
Download Acoustical Simulations of the Human Vocal Tract Using the 1D and 2D Digital Waveguide Software Model This paper details software under development that uses the digital waveguide physical model to represent the sound creation mechanism and environment associated with the production of speech, specifically the human vocal tract. Focus is directed towards a comparison between the existing 1D waveguide method, on which several studies have already been conducted, and the developing 2D waveguide mesh method. The construction of the two models and the application of the tract geometry is examined, in addition, the inclusion of dynamic articulatory variations to increase the ability of such systems to create natural sounding speech is discussed. Results obtained from each suggest that the 2D model is capable of producing similarly accurate vowel spectra to that already accomplished with the 1D version, although speech-like sounds created with the 2D mesh appear to exhibit greater realism.
Download Room Impulse Response Shaping for Enhancement of Perceived Spaciousness and Auditory Distance Room impulse response is one of most important information with localization of sound source in 3D audio. Actually we can adjust the distance and spaciousness of a sound source with impulse response of the room. Through consequent experiments, we found that C80 and EDT are varying systematically with sound source distances, and these variations are due to early reflection decay curves. This paper contains brief explanation of the two parameters as auditory distance cues, shaping of early reflection decay curves for control of auditory distance, and psychological test results of auditory distance control with early refection decay curve shaping. With these validations, we can confirm early reflection decay curve shape is effective factor for control of perceptual auditory distance and spaciousness in the room.
Download Semi-automatic Ambience Generation Ambiances are background recordings used in audiovisual productions to make listeners feel they are in places like a pub or a farm. Accessing to commercially available atmosphere libraries is a convenient alternative to sending teams to record ambiances yet they limit the creation in different ways. First, they are already mixed, which reduces the flexibility to add, remove individual sounds or change its panning. Secondly, the number of ambient libraries is limited. We propose a semi-automatic system for ambiance generation. The system creates ambiances on demand given text queries by fetching relevant sounds from a large sound effect database and importing them into a sequencer multitrack project. Ambiances of diverse nature can be created easily. Several controls are provided to the users to refine the type of samples and the sound arrangement.
Download MATCONCAT: An Application for Exploring Concatenative Sound Synthesis Using MATLAB The author has developed an application in MATLAB implementing concatenative sound synthesis (CSS) using feature matching. CSS is a process of combining short pieces of recorded sound to construct new sonic forms. Historically, CSS was developed for text-to-speech synthesis, but recently it has been explored as a musical sound synthesis method. The results have been called ‘musaics,’ the sonic analogue to mosaics made from small pieces of colored tile. Though this MATLAB application is less sophisticated than other audio mosaic algorithms, it is meant to be a free and open application for demonstrating and experimenting with the process. The author has used this application to create many interesting and entertaining sound examples. It has also been used to create several electroacoustic compositions. The application, and all of the sound examples presented here, can be downloaded for free from http://www.mat.ucsb.edu/˜b.sturm.