Wavetable and sampling synthesis enable the playback of arbitrary sounds, including those with a rich harmonic structure, without increasing the computational complexity. Although resampling allows for changing the pitch of a stored sample, there are artifacts. In particular, increasing the pitch is susceptible to disturbing aliasing artifacts. A novel approach to reduce aliasing, which is based on an integrated wavetable and a differentiation of the output signal, has been proposed previously by Geiger. This paper extends Geiger’s method by integrating the waveform multiple times before storing it, and during playback a sample rate conversion method is applied and the output signal is then differentiated as many times as the wavetable has been integrated. With only a minor increase in computational cost, the use of higher-order filtering reduces aliasing more than first-order techniques.
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Directional Audio Coding (DirAC) is a perceptually motivated microphone technique that models the sound field as a combination of a plane wave and a surrounding diffuse field with a time–frequency resolution that approximates that of the human spatial hearing. In this paper a recently proposed covariance domain spatial-sound rendering method was applied to optimize the DirAC reproduction by minimizing the amount of the decorrelated sound energy. When several semi-independent microphone signals were available, this procedure was shown to improve the overall perceived sound quality, especially with audio content that has an impulsive fine structure, such as applause and speech. In all tests, the covariance rendering method performed similarly or better than the legacy rendering method, making it the preferred choice for performing DirAC synthesis.
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Although an impulse response is the output from a linear system when excited by a pulse, such responses cannot be obtained with a high signal-to-noise ratio (SNR) because the pulse has low energy. Swept sine signals and maximum length sequences are alternative inputs, however, conventional signals still have low SNR problems in some frequency bands. This study is based on a swept-sine that maintains a constant SNR regardless of the frequency. The spectrum of a measurement signal is shaped, adapting to not only the background noise spectrum but also the recursively estimated transfer function of the system itself. To verify the validity of the proposed method, the authors measured the room impulse response in a noisy environment and calculated the room frequency response. The experimental result showed that a frequency response with an almost constant SNR was obtained with two iterations. This approach is useful in reverberation time measurements.
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The nonlinear properties of air with ultrasonic sound allows for the creation of a “virtual microphone,” which is the analog of the ultrasonic narrow-beam loudspeaker. When an ultrasonic wave (pump wave) mixes with a baseband audio sound, sidebands are created around the ultrasonic carrier, and these can be demodulated at the receiver. A preliminary investigation showed that the following technical requirements must be achieved: (a) generation of an ultrasonic wave with small phase noise; (b) reception of the wave over a wide dynamic range to allow for real-time demodulation; (c) a composite dynamic of 120 dB within the region around the 40-kHz carrier in order to achieve a microphone comparable to conventional microphones.
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The music and entertainment industries have developed some momentum toward preventing music-induced hearing loss. This paper briefly summarizes the standards and regulations that constitute occupational noise regulations in the U.S. Although the music and entertainment industries are subject to the same requirements to protect the hearing of their workers as other industries, difficulties applying and enforcing regulations put these workers at risk. This contrasts with the EU, and especially the UK, where the legal authority partnered with stakeholders in working groups to develop, document, implement, and gauge the effectiveness of regulatory legislation. This article is motivated to initiate dialogues among audio professionals working in music and entertainment as to how these industries might be proactive in preventing hearing loss and conserving the hearing of their workers.
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Successful speech communication in vehicles is important because it facilitates social interactions as well as the delivery of navigation and safety information. Noise in vehicles is especially problematic for the elderly population with hearing deficits. A variety of objective speech intelligibility metrics have been explored over the years. Using a vehicle simulation with on-road interior sounds, the speech intelligibility index (SII) was evaluated at the sentence speech reception threshold (sSRT) using various vehicle operating conditions as well as talker and listener configurations. Unlike previous studies that used normal hearing individuals, this study used participants who had various hearing profiles, both normal hearing and hearing impaired. It was found that the SII at sSRT depended on background noise spectra. The measurement and analysis techniques described in this study can be used to design vehicle acoustics to enhance intelligibility.
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The quest for low power consumption with high output power and audio quality continues to drive the “class struggle” in audio amplifier design. The introduction of silicon carbide transistors is one important development in this field, helping to reduce distortions and losses in switching amplifiers. New testing methods challenge conventional wisdom about how best to predict the perceptual effects of amplifier distortion.
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