AES Show: Make the Right Connections Audio Engineering Society

AES San Francisco 2008
Poster Session P26

Sunday, October 5, 2:30 pm — 4:00 pm

P26 - Audio Digital Signal Processing and Effects—Part 2

P26-1 Applications of Algorithmically-Generated Digital Audio for Web-Based Sonic Measure Ear TrainingChristopher Ariza, Towson University - Towson, MD, USA
This paper examines applications of algorithmically-generated digital audio for a new type of ear training. This approach, called sonic measure ear training, circumvents the many limits of MIDI-based aural testing, and may offer a valuable resource for computer musicians and audio engineers. The Post-Ut system, introduced here, is the first web-based ear training system to offer sonic measure ear-training. After describing the design of the Post-Ut system, including the use of athenaCL, Csound, Python, and MySQL, the audio generation procedures are examined in detail. The design of questions and perceptual considerations are evaluated, and practical applications and opportunities for future development are outlined.
Convention Paper 7645 (Purchase now)

P26-2 A Perceptual Model-Based Speech Enhancement AlgorithmRongshan Yu, Dolby Laboratories - San Francisco, CA, USA
This paper presents a perceptual model-based speech enhancement algorithm. The proposed algorithm measures the amount of the audible noise in the input noisy speech explicitly by using a psychoacoustic model, and decides an appropriate amount of noise reduction accordingly to achieve good noise level reduction without introducing significant distortion to the clean speech embedded in the input noisy signal. The proposed algorithm also mitigates the musical noise problem commonly encountered in conventional speech enhancement algorithms by having the amount of noise reduction adapt to the instantly estimated noise amplitude. Good performance of the proposed algorithm has been confirmed through objective and subjective tests.
Convention Paper 7646 (Purchase now)

P26-3 Real Time Implementation of an ESPRIT-Based Bass Enhancement AlgorithmLorenzo Palestini, Emanuele Moretti, Paolo Peretti, Stefania Cecchi, Laura Romoli, Francesco Piazza, UniversitĂ  Politecnica delle Marche - Ancona, Italy
This paper presents a software real-time implementation for the NU-Tech platform of a bass enhancement algorithm based on the FAPI subspace tracker and the ESPRIT algorithm for fundamentals estimation to realize bass improvement of small loudspeakers exploiting the well known psychoacoustic phenomenon of the missing fundamental. Comparative informal listening tests have been performed to validate the virtual bass improvement, and their results show that the proposed method is well appreciated.
Convention Paper 7647 (Purchase now)

P26-4 Low-Power Implementation of a Subband Acoustic Echo Canceller for Portable DevicesJulie Johnson, David Hermann, John Wdowiak, Edward Chau, Hamid Sheikhzadeh, ON Semiconductor - Waterloo, Ontario, Canada
Portable audio communication devices require increasingly superior audio quality while using minimal power. Devices such as cell phones with speakerphone functionality can generate substantial acoustic echo due to the proximity of the microphone and speaker. To improve the audio quality in such devices, an oversampled subband acoustic echo canceller has been implemented on a miniature low-power dual core DSP system. This application is comprised of three subband-based algorithms: a Pseudo-Affine Projection adaptive filter, an Ephraim-Malah based single-microphone noise reduction algorithm, and a novel nonlinear residual echo suppressor. The system consumes less than 4 mW of power when configured with a 128 ms filter. Real-world tests indicate an echo return loss enhancement of greater than 30 dB for typical input levels.
Convention Paper 7648 (Purchase now)

P26-5 A Digital Model of the Echoplex Tape DelaySteinunn Arnardottir, Jonathan S. Abel, Julius O. Smith, Stanford University - Stanford, CA, USA
The Echoplex is a tape delay unit featuring fixed playback and erase heads, a moveable record head, and a tape loop moving at roughly 8 ips. The relatively slow tape speed allows large frequency shifts, including "sonic booms" and shifting of the tape bias signal into the audio band. Here, the Ecxhoplex tape delay is modeled with read, write, and erase pointers moving along a circular buffer. The model separately generates the quasiperiodic capstan and pinch wheel components and drift of the observed fluctuating time delay. This delay drives an interpolated write simulating the record head. To prevent aliasing in the presence of a changing record head speed, an anti-aliasing filter with a variable cutoff frequency is described.
Convention Paper 7649 (Purchase now)

P26-6 A Digital Reverberator Modeled after the Scattering of Acoustic Waves by Trees in a ForestKyle Spratt, Jonathan S. Abel, Stanford University - Stanford, CA, USA
A digital reverberator modeled after the scattering of acoustic waves among trees in an idealized forest is presented. Termed "treeverb," the technique simulates forest acoustics using a network of digital waveguides, with bi-directional delay lines connecting trees represented by multi-port scattering junctions. The reverberator is designed by selecting tree locations and diameters, with waveguide delays determined by inter-tree distances, and scattering filters fixed according to tree-to-tree angles and trunk diameters. The scattering is modeled as that of plane waves normally incident on a rigid cylinder, and a simple low-order scattering filter is presented and shown to closely approximate the theoretical scattering. Small forests are seen to yield dense, gated reverb-like impulse responses.