Spatial Multizone Soundfield Reproduction Design

<p>It is desirable for people sharing a physical space to access different multimedia information streams simultaneously. For a good user experience, the interference of the different streams should be held to a minimum. This is straightforward for the video component but currently difficult for the audio sound component. Spatial multizone soundfield reproduction, which aims to provide an individual sound environment to each of a set of listeners without the use of physical isolation or headphones, has drawn significant attention of researchers in recent years. The realization of multizone soundfield reproduction is a conceptually challenging problem as currently most of the soundfield reproduction techniques concentrate on a single zone. This thesis considers the theory and design of a multizone soundfield reproduction system using arrays of loudspeakers in given complex environments. We first introduce a novel method for spatial multizone soundfield reproduction based on describing the desired multizone soundfield as an orthogonal expansion of formulated basis functions over the desired reproduction region. This provides the theoretical basis of both 2-D (height invariant) and 3-D soundfield reproduction for this work. We then extend the reproduction of the multizone soundfield over the desired region to reverberant environments, which is based on the identification of the acoustic transfer function (ATF) from the loudspeaker over the desired reproduction region using sparse methods. The simulation results confirm that the method leads to a significantly reduced number of required microphones for an accurate multizone sound reproduction compared with the state of the art, while it also facilitates the reproduction over a wide frequency range. In addition, we focus on the improvements of the proposed multizone reproduction system with regard to practical implementation. The so-called 2.5D multizone oundfield reproduction is considered to accurately reproduce the desired multizone soundfield over a selected 2-D plane at the height approximately level with the listener’s ears using a single array of loudspeakers with 3-D reverberant settings. Then, we propose an adaptive reverberation cancelation method for the multizone soundfield reproduction within the desired region and simplify the prior soundfield measurement process. Simulation results suggest that the proposed method provides a faster convergence rate than the comparative approaches under the same hardware provision. Finally, we conduct the real-world implementation based on the proposed theoretical work. The experimental results show that we can achieve a very noticeable acoustic energy contrast between the signals recorded in the bright zone and the quiet zone, especially for the system implementation with reverberation equalization.</p>

Spatial Multizone Soundfield Reproduction Design

<p>It is desirable for people sharing a physical space to access different multimedia information streams simultaneously. For a good user experience, the interference of the different streams should be held to a minimum. This is straightforward for the video component but currently difficult for the audio sound component. Spatial multizone soundfield reproduction, which aims to provide an individual sound environment to each of a set of listeners without the use of physical isolation or headphones, has drawn significant attention of researchers in recent years. The realization of multizone soundfield reproduction is a conceptually challenging problem as currently most of the soundfield reproduction techniques concentrate on a single zone. This thesis considers the theory and design of a multizone soundfield reproduction system using arrays of loudspeakers in given complex environments. We first introduce a novel method for spatial multizone soundfield reproduction based on describing the desired multizone soundfield as an orthogonal expansion of formulated basis functions over the desired reproduction region. This provides the theoretical basis of both 2-D (height invariant) and 3-D soundfield reproduction for this work. We then extend the reproduction of the multizone soundfield over the desired region to reverberant environments, which is based on the identification of the acoustic transfer function (ATF) from the loudspeaker over the desired reproduction region using sparse methods. The simulation results confirm that the method leads to a significantly reduced number of required microphones for an accurate multizone sound reproduction compared with the state of the art, while it also facilitates the reproduction over a wide frequency range. In addition, we focus on the improvements of the proposed multizone reproduction system with regard to practical implementation. The so-called 2.5D multizone oundfield reproduction is considered to accurately reproduce the desired multizone soundfield over a selected 2-D plane at the height approximately level with the listener’s ears using a single array of loudspeakers with 3-D reverberant settings. Then, we propose an adaptive reverberation cancelation method for the multizone soundfield reproduction within the desired region and simplify the prior soundfield measurement process. Simulation results suggest that the proposed method provides a faster convergence rate than the comparative approaches under the same hardware provision. Finally, we conduct the real-world implementation based on the proposed theoretical work. The experimental results show that we can achieve a very noticeable acoustic energy contrast between the signals recorded in the bright zone and the quiet zone, especially for the system implementation with reverberation equalization.</p>

Development of experimental vibro-acoustic transfer function for a system with combined rolling-sliding motion

Combined rolling-sliding contact is present in popular non-linear systems such as cam-follower, gears, clutches, and brakes. These systems produce significant noise due to complex contact between the components during operation. The noise generated is a strong function of the contact parameters and excitation to the system. The objective of this study is to develop a transfer function to quantify the vibro-acoustic noise for various contact conditions. Acceleration, reaction forces, and acoustic pressure measurements are made on a cam-follower setup with combined rolling-sliding contact. Experiments are performed under different conditions of friction, lubrication, load, and speed. Contact forces are back-calculated using the kinematics. The transfer function relating the acoustic pressure to different forces is estimated. It is observed that the contact parameters govern the transfer function and hence the vibro-acoustic systems. The developed transfer function is useful in designing better sub-systems with combined rolling-sliding contact to reduce noise exposure, as a direct technique to relate the contact parameters to the noise does not exist. This study can be extended to other complex systems such as gears and clutches.

Sound field reproduction using multilayer equivalent source method

Sound field reproduction aims to create or reproduce a desired sound environment, where both the audio content and the spatial property of the sound field are preserved. For a practical reproduction system which is usually placed in a real 'listening room', acoustic transfer function measurement of the loudspeaker array is a time consuming work. The equivalent source method is an option to interpolate loudspeaker array acoustic transfer functions over the target region in reverberant sound field and has been implemented in the preceding researches. However, the selection of the optimized distances of the equivalent sources remains a challenging problem, especially considering the complex acoustic environment in reverberant room. In this work, we apply a multilayer equivalent source method. A simulation is conducted in virtual listening rooms with different reverberation conditions to investigate the reproduction performance of the proposed method. The comparison with the conventional single layer equivalent source method is provided.

Computing Radiated Sound Power using Quadratic Power Transfer Vector (QPTV)

Pressure Acoustic Transfer Functions or Vectors (PATVs) relate the surface velocity of a structure to the sound pressure level at a field point in the surrounding fluid. These functions depend only on the structure geometry, properties of the fluid medium (sound speed and characteristic density), the excitation frequency and the location of the field point, but are independent of the surface velocity values themselves. Once the pressure acoustic transfer function is computed between a structure and a specified field point, we can compute pressure at this point for any boundary velocity distribution by simply multiplying the forcing function (surface velocity) with the acoustic transfer function. These PATVs are usually computed by application of the Reciprocity Principle, and their computation is well understood. In this work, we present a novel way to compute the Velocity Acoustic Transfer Vector (VATV) which is a relation between the surface velocity of the structure and fluid particle velocity at a field point. To our knowledge, the computation of the VATV is completely new and has not been published in earlier works. By combining the PATVs and VATVs at a number of field points surrounding the structure, we obtain the Quadratic Power Transfer Vector (QPTV) that allows us to compute the sound power radiated by a structure for ANY surface velocity distribution. This allows rapid computation of the sound power for an arbitrary surface velocity distributions and is useful in designing quiet structures by minimizing the sound power radiated.

Research on improving measurement accuracy of acoustic transfer function of underwater vehicle

To address the problem that acoustic transfer functions with underwater platforms cannot be measured accurately, this paper presents a method based on phase compensation to improve the accuracy of acoustic transfer function measurements on underwater platforms. The time-domain impulse response signals with multiple cycles are first collected and intercepted, and then their phase differences are estimated using the least-squares method, and phase compensation is used to align the phases of all the signals, and then the impulse response signals are weighted and averaged over all the impulse response signals to cancel out the random noise. The water pool test proves that this method reduces the measurement random noise while obtaining a high-fidelity time domain transfer function, which effectively improves the signal-to-noise ratio of the measurement. The method adopts only one measurement signal, and without changing the measurement system, the random noise is cancelled out by the in-phase superposition of the multi-cycle impulse response signals to avoid the nonlinear distortion of the measurement results.

Residual Echo Suppression Considering Harmonic Distortion and Temporal Correlation

In acoustic echo cancellation, a certain level of residual echo resides in the output of the linear echo canceller because of the nonlinearity of the power amplifier, loudspeaker, and acoustic transfer function in addition to the estimation error of the linear echo canceller. The residual echo in the current frame is correlated not only to the linear echo estimates for the harmonically-related frequency bins in the current frame, but also with linear echo estimates, residual echo estimates, and microphone signals in adjacent frames. In this paper, we propose a residual echo suppression scheme considering harmonic distortion and temporal correlation in the short-time Fourier transform domain. To exploit residual echo estimates and microphone signals in past frames without the adverse effect of the near-end speech and noise, we adopt a double-talk detector which is tuned to have a low false rejection rate of double-talks. Experimental results show that the proposed method outperformed the conventional approach in terms of the echo return loss enhancement during single-talk periods and the perceptual evaluation of speech quality scores during double-talk periods.

Measurement of Head-Related Transfer Functions: A Review

A head-related transfer function (HRTF) describes an acoustic transfer function between a point sound source in the free-field and a defined position in the listener’s ear canal, and plays an essential role in creating immersive virtual acoustic environments (VAEs) reproduced over headphones or loudspeakers. HRTFs are highly individual, and depend on directions and distances (near-field HRTFs). However, the measurement of high-density HRTF datasets is usually time-consuming, especially for human subjects. Over the years, various novel measurement setups and methods have been proposed for the fast acquisition of individual HRTFs while maintaining high measurement accuracy. This review paper provides an overview of various HRTF measurement systems and some insights into trends in individual HRTF measurements.