AES Los Angeles 2014
Paper Session P17
P17 - Signal Processing: Part 2
Sunday, October 12, 9:00 am — 12:00 pm (Room 308 AB)
J. Keith McElveen, Wave Sciences - Charleston, SC, USA
P17-1 A Practical Approach to Robust Speech Recognition Using Two Microphones in Driving Environments—Jaeyoun Cho, Samsung Electronics Co. Ltd. - Suwon-si, Gyeonggi-do, Korea; Seungyeol Lee, Samsung Electronics Co. Ltd. - Suwon-si, Gyeonggi-do, Korea; Inwoo Hwang, Samsung Electronics Co. Ltd. - Suwon-si, Gyeonggi-do, Korea
Now that the technologies related to the automatic speech recognition have been mature enough and applicable to our everyday life, people have started considering speech as the most desirable human-device interaction means and utilized speech recognition in vehicles. Nonetheless, it is still challenging to recognize speech correctly in driving environments for at least two reasons. One is that the speech signal is corrupted by innumerable noise sources such as the engine sound, road friction, music from the radio, even worse the mixture of spoken words by passengers, etc. Another is that the recognition device may be put at any place like cup holder, passenger seat or dashboard. In this paper we propose a robust speech recognition front-end that removes the probable ambient noise in a driving car regardless of where the recognition device is. The proposed method finds the direction of speech and enhances the speech signal by first detecting the existence of speech utterance using only two microphones. This front-end is designed with practical consideration so that its implementation in the mobile device showed higher recognition accuracy, shorter processing latency and lower computing power consumption than any other top-tier methods.
Convention Paper 9191
P17-2 Predistortion of a Bidirectional Cuk Audio Amplifier—Thomas Haagen Birch, Technical University of Denmark - Kgs. Lyngby, Denmark; Dennis Nielsen, Technical University of Denmark - Kgs. Lyngby, Denmark; Arnold Knott, Technical University of Denmark - Kgs. Lyngby, Denmark; Michael A. E. Andersen, Technical University of Denmark - Kgs. Lyngby, Denmark
Some non-linear amplifier topologies are capable of providing a larger voltage gain than one from a DC source, which could make them suitable for various applications. However, the non-linearities introduce a significant amount of harmonic distortion (THD). Some of this distortion could be reduced using predistortion. This paper suggests linearizing a nonlinear bidirectional Cuk audio amplifier using an analog predistortion approach. A prototype power stage was built and results show that a voltage gain of up to 9 dB and reduction in THD from 6% down to 3% was obtainable using this approach.
Convention Paper 9192
P17-3 Frequency Dependent Loss Analysis and Minimization of System Losses in Switch-Mode Audio Power Amplifiers—Akira Yamauchi, Technical University of Denmark - Kgs. Lyngby, Denmark; Arnold Knott, Technical University of Denmark - Kgs. Lyngby, Denmark; Ivan H. H. Jørgensen, Technical University of Denmark - Kgs. Lyngby, Denmark; Michael A. E. Andersen, Technical University of Denmark - Kgs. Lyngby, Denmark
In this paper the frequency dependent losses in switch-mode audio power amplifiers are analyzed and the loss model is improved by taking the voltage dependence of the parasitic capacitance of MOSFETs into account. The estimated power losses are compared to the measurement and great accuracy is achieved. By choosing the optimal switching frequency based on the proposed analysis, the experimental results show that the system power losses of the reference design are minimized and an efficiency improvement of 8% in maximum is achieved without compromising audio performances.
Convention Paper 9193
P17-4 Resolving Delay-Free Loops in Recursive Filters Using the Modified Härmä Method—Will Pirkle, University of Miami - Coral Gables, FL, USA
Resolving delay-free loops in recursive filter structures has been a longstanding problem approached in several different ways including signal flow graph manipulation ,  and more recently with Zavalishin’s instantaneous response technique ,. Härmä demonstrates a method for resolving delay-less loops in recursive filter structures  but the technique is limited to a specific generic loop topology in which the feedforward branch does not implement signal processing; all processing is implemented in one or more delay-less feedback loops. We modify Härmä’s method to accommodate filter processing in the feedforward branch and provide a step-by-step method to resolve delay-less loops in recursive filter structures. We conclude with examples including a new method of synthesizing fourth order filters.
Convention Paper 9194
P17-5 Novel Hybrid Virtual Analog Filters Based on the Sallen-Key Architecture—Will Pirkle, University of Miami - Coral Gables, FL, USA
The Sallen-Key filter structure is a revered analog filter design topology. In Sallen-Key lowpass and highpass filters, the cutoff frequency and resonance (Q) controls are decoupled though the cutoff and resonant frequencies are not. In this paper we demonstrate novel variations on the Sallen-Key architecture and we decouple the resonant and cutoff frequencies. This produces multiple hybrid filter designs including resonant quasi-first order lowpass and highpass filters, resonant quasi-first order low and high shelving filters, decoupled resonant second order filters and doubly resonant quasi-second order lowpass and highpass filters. In the doubly-resonant filters all three frequencies may be decoupled and independently adjustable; they also self-oscillate at both resonant frequencies.
Convention Paper 9195
P17-6 Timbre Imitation and Adaptation for Experimental Music Instruments: An interactive Approach Using Real-Time Digital Signal Processing Framework—Mingfeng Zhang, University of Rochester - Rochester, NY, USA; John Granzow, Stanford University - Stanford, CA, USA; Gang Ren, University of Rochester - Rochester, NY, USA; Mark F. Bocko, University of Rochester - Rochester, NY, USA
We propose a real-time digital signal processing framework to extend the timbre control capability of experimental musical instruments. We focus on two music cognition concepts of timbre imitation and adaption to enable experimental musical instruments to be integrated into existing ensemble works. In timbre imitation, we aim to simulate known timbre patterns to enhance the musical coherence during an ensemble performance. In timbre adaptation, we explore extended timbre manipulation settings such as complementary timbre and contrasting timbre. Our proposed framework is implemented on a low cost real-time digital signal processing system to ensure easy adaptability. Our study is based on saxophone and violin but can be readily generalized to other instrument categories.
Convention Paper 9196