AES Los Angeles 2014
Paper Session P18
P18 - Applications in Audio: Part 1
Sunday, October 12, 9:00 am — 12:30 pm (Room 309)
Jung Wook (Jonathan) Hong, McGill University - Montreal, QC, Canada; GKL Audio Inc. - Montreal, QC, Canada
P18-1 Measuring Time Varying or Offset Voltage Dependent Harmonic and Intermodulation Distortion via Filter Banks Including a Stairstep Signal and Measuring FM Distortion in IM Distortion Signals—Ronald Quan, Ron Quan Designs - Cupertino, CA, USA
This paper will present methods of measuring dynamic harmonic distortion using filter banks and a staircase signal. The harmonic distortion is measured in real time at each level of the staircase signal. For measuring dynamic time varying 2nd and 3rd order intermodulation distortion, a low frequency sinewave signal is combined with a higher frequency signal. Also, the FM distortion of the intermodulation distortion signals is measured. FM distortion in current mode op amps is measured. Also FM distortion is measured in an op amp with conventional Miller compensation then measured later in an op amp with two-pole compensation. Finally, Volterra Series distortion analysis is included as part of an equation that describes phase or frequency modulation effects in a nonlinear system.
Convention Paper 9197
P18-2 DC Servos and Digitally-Controlled Microphone Preamplifiers—Gary Hebert, That Corp. - Milford, MA, USA
Microphone preamplifiers for professional audio applications require a very wide range of gain and low noise in order to provide a high-quality interface with the vast number of available microphones. In many modern systems the preamplifier gain is controlled indirectly via a digital interface in discrete steps. Often dc servo amplifiers are employed as a means of keeping the dc gain fixed to avoid large changes in output offset voltage while the audio band gain is varied. The resulting highpass filter response varies substantially as a function of the preamplifier gain. We investigate the frequency and time-domain effects of this. We also investigate several approaches to minimize these effects.
Convention Paper 9198
P18-3 The Design of Urban Sound Monitoring Devices—Charlie Mydlarz, New York University, CUSP - New York, NY, USA; Samuel Nacach, New York University - New York, NY, USA; Agnieszka Roginska, New York University - New York, NY, USA; Tae Hong Park, New York University - New York, NY, USA
The urban sound environment of New York City is notoriously loud and dynamic. As such, scientists, recording engineers, and soundscape researchers continuously explore methods to capture and monitor such urban sound environments. One method to accurately monitor and ultimately understand this dynamic environment involves a process of long-term sound capture, measurement and analysis. Urban sound recording requires the use of robust and resilient acoustic sensors, where unpredictable external conditions can have a negative impact on acoustic data quality. Accordingly, this paper describes the design and build of a self-contained urban acoustic sensing device to capture, analyze, and transmit high quality sound from any given urban environment. This forms part of a collaborative effort between New York University’s (NYU) Center for Urban Science and Progress (CUSP) and the NYU Steinhardt School’s Citygram Project. The presented acoustic sensing device prototype incorporates a quad core Android based mini PC with Wi-Fi capabilities, a custom MEMS microphone and a USB audio device. The design considerations, materials used, noise mitigation strategies and the associated measurements are detailed in the following paper.
Convention Paper 9199
P18-4 A Comparison of Real-Time Pitch Detection Algorithms in SuperCollider—Elliot Kermit-Canfield, Stanford University - Stanford, CA, USA
Three readily-available pitch detection algorithms implemented as unit generators in the SuperCollider programming language are evaluated and compared with regard to their accuracy and latency for a variety of test signals consisting of both harmonic and non-harmonic content. Suggestions are made for the type of signal on which each algorithm performs well.
Convention Paper 9200
P18-5 Performance Comparison Between Nested Differentiating Feedback Loops and Classic Three Stage Operational Amplifier Architectures: A SPICE-Based Simulation Approach—Ariel Muszkat, Universidad Nacional de Tres de Febrero - Buenos Aires, Argentina; David Kadener, Universidad Nacional de Tres de Febrero - Buenos Aires, Argentina
Since 1970, the three stage operational amplifier with dominant pole compensation has become the standard basis in amplifier’s architectures. However, during the 1980s and following years the nested differentiating feedback loops (NDFL) concept was introduced by Edward M. Cherry as an attempt to improve the classic power amp performance, mainly distortion caused by class B output stages. The proposal of this work lies on the first part of a comparison between both topologies’ performance in a SPICE based simulator. Most important analyzed parameters are open-loop gain, distortion, transient response, and, of course, stability. In addition, modern semi-conductor devices and improved inner stages are used to make the comparison circuits based on small signal devices such as discrete operational amplifiers.
Convention Paper 9201
P18-6 Making Audio Sound Better One Square Wave at a Time (Or How an Algorithm Called “Undo” Fixes Audio)—Leif Claesson, Omnia Audio - Cleveland, OH, USA
Audio mastering engineers have felt increasing pressure over the years to master recordings at ever increasing loudness levels as compared to other contemporary recordings, by way of dynamic compression, peak limiting, and hard clipping. This pursuit of loudness adds distortion, and reduces fidelity. When radio stations play the compromised audio through their FM processing chains, this confluence of degradation causes serious audio quality issues on air. This paper shall examine what the music endures when broadcast on FM, and how that led to the invention of the “undo” algorithm, which repairs damage caused by these mastering techniques by adaptively de-clipping and de-compressing the mastered recordings.
Convention Paper 9202
P18-7 Acoustic Surveillance of Hazardous Situations Using Nonnegative Matrix Factorization and Hidden Markov Model—Kwang Myung Jeon, Gwangju Institute of Science and Technology (GIST) - Gwangju, Korea; Dong Yun Lee, Gwangju Institute of Science and Technology (GIST) - Gwangju, Korea; Hong Kook Kim, Gwangju Institute of Science and Tech (GIST) - Gwangju, Korea; Myung J. Lee, City University of New York - New York, NY, USA
In this paper an acoustic surveillance method is proposed for accurately detecting hazardous situations under noisy conditions. In order to improve detection accuracy, the proposed method first tries to separate each atypical event from the input noisy audio signal. Next, maximum likelihood classification using multiple hidden Markov models (HMMs) is carried out to decide whether or not an atypical event occurs. Performance evaluation shows that the proposed method achieves higher detection accuracy under various signal-to-noise ratio (SNR) conditions than a conventional HMM-based method.
Convention Paper 9203