On the effect of SNR and superdirective beamforming in speaker diarisation in meetings

This paper examines the effect of sensor performance on speaker diarisation in meetings and investigates the use of more advanced beamforming techniques, beyond the typically employed delay-sum beamformer, for mitigating the effects of poorer sensor performance. We present superdirective beamforming and investigate how different time difference of arrival (TDOA) smoothing and beamforming techniques influence the performance of state-of-the-art diarisation systems. We produced and transcribed a new corpus of meetings recorded in the instrumented meeting room using a high SNR analogue and a newly developed low SNR digital MEMS microphone array (DMMA.2). This research demonstrates that TDOA smoothing has a significant effect on the diarisation error rate and that simple noise reduction and beamforming schemes suffice to overcome audio signal degradation due to the lower SNR of modern MEMS microphones. Index Terms — Speaker diarisation in meetings, digital MEMS microphone array, time difference of arrival (TDOA), superdirective beamforming 1

An Innovations-Based Noise Cancelling Technique on Inverse Kepstrum Whitening Filter and Adaptive FIR Filter in Beamforming Structure

This paper presents an acoustic noise cancelling technique using an inverse kepstrum system as an innovations-based whitening application for an adaptive finite impulse response (FIR) filter in beamforming structure. The inverse kepstrum method uses an innovations-whitened form from one acoustic path transfer function between a reference microphone sensor and a noise source so that the rear-end reference signal will then be a whitened sequence to a cascaded adaptive FIR filter in the beamforming structure. By using an inverse kepstrum filter as a whitening filter with the use of a delay filter, the cascaded adaptive FIR filter estimates only the numerator of the polynomial part from the ratio of overall combined transfer functions. The test results have shown that the adaptive FIR filter is more effective in beamforming structure than an adaptive noise cancelling (ANC) structure in terms of signal distortion in the desired signal and noise reduction in noise with nonminimum phase components. In addition, the inverse kepstrum method shows almost the same convergence level in estimate of noise statistics with the use of a smaller amount of adaptive FIR filter weights than the kepstrum method, hence it could provide better computational simplicity in processing. Furthermore, the rear-end inverse kepstrum method in beamforming structure has shown less signal distortion in the desired signal than the front-end kepstrum method and the front-end inverse kepstrum method in beamforming structure

Noise cancellation over spatial regions using adaptive wave domain processing

This paper proposes wave-domain adaptive processing for noise cancellation within a large spatial region. We use fundamental solutions of the Helmholtz wave-equation as basis functions to express the noise field over a spatial region and show the wave-domain processing directly on the decomposition coefficients to control the entire region. A feedback control system is implemented, where only a single microphone array is placed at the boundary of the control region to measure the residual signals, and a loudspeaker array is used to generate the anti-noise signals. We develop the adaptive wave-domain filtered-x least mean square algorithm. Simulation results show that using the proposed method the noise over the entire control region can be significantly reduced with fast convergence in both free-field and reverberant environmentsThanks to Australian Research Councils Discovery Projects funding scheme (project no. DP140103412). The work of J. Zhang was sponsored by the China Scholarship Council with the Australian National University

Recording, Analysis and Playback of Spatial Sound Field using Novel Design Methods of Transducer Arrays

Nowadays, a growing interest in the recording and reproduction of spatial audio has been observed. With virtual and augmented reality technologies spreading fast thanks to entertainment and video game industries, also the professional opportunities in the field of engineering are evolving. However, despite many microphone arrays are reaching the market, most of them is not optimized for engineering or diagnostic use and remains mainly confined to voice and music recordings. In this thesis, the design of two new systems for recording and analysing the spatial distribution of sound energy, employing arrays of transducers and cameras, is discussed. Both acoustic and visual spatial information is recorded and combined together to produce static and dynamic colour maps, with a specially designed software and employing Ambisonics and Spatial PCM Sampling (SPS), two common spatial audio formats, for signals processing. The first solution consists in a microphone array made of 32 capsules and a circular array of eight cameras, optimized for low frequencies. The size of the array is designed accordingly to the frequency range of interest for automotive Noise, Vibration & Harshness (NVH) applications. The second system is an underwater probe with four hydrophones and a panoramic camera, with which it is possible to monitor the effects of underwater noise produced by human activities on marine species. Finite Elements Method (FEM) simulations have been used to calculate the array response, thus deriving the filtering matrix and performing theoretical evaluation of the spatial performance. Field tests of the proposed solutions are presented in comparison with the current state-of-the-art equipment. The faithful reproduction of the spatial sound field arouses equally interest. Hence, a method to playback panoramic video with spatial audio is presented, making use of Virtual Reality (VR) technology, spatial audio, individualized Head Related Transfer Functions (HRTFs) and personalized headphones equalization. The work in its entirety presents a complete methodology for recording, analysing and reproducing the spatial information of soundscapes

Spatial Noise-Field Control With Online Secondary Path Modeling: A Wave-Domain Approach

Due to strong interchannel interference in multichannel active noise control (ANC), there are fundamental problems associated with the filter adaptation and online secondary path modeling remains a major challenge. This paper proposes a wave-domain adaptation algorithm for multichannel ANC with online secondary path modelling to cancel tonal noise over an extended region of two-dimensional plane in a reverberant room. The design is based on exploiting the diagonal-dominance property of the secondary path in the wave domain. The proposed wave-domain secondary path model is applicable to both concentric and nonconcentric circular loudspeakers and microphone array placement, and is also robust against array positioning errors. Normalized least mean squares-type algorithms are adopted for adaptive feedback control. Computational complexity is analyzed and compared with the conventional time-domain and frequency-domain multichannel ANCs. Through simulation-based verification in comparison with existing methods, the proposed algorithm demonstrates more efficient adaptation with low-level auxiliary noise.DP14010341

A Background Noise Reduction Technique Using Adaptive Noise Cancellation for Microphone Arrays

Background noise in wind tunnel environments poses a challenge to acoustic measurements due to possible low or negative Signal to Noise Ratios (SNRs) present in the testing environment. This paper overviews the application of time domain Adaptive Noise Cancellation (ANC) to microphone array signals with an intended application of background noise reduction in wind tunnels. An experiment was conducted to simulate background noise from a wind tunnel circuit measured by an out-of-flow microphone array in the tunnel test section. A reference microphone was used to acquire a background noise signal which interfered with the desired primary noise source signal at the array. The technique s efficacy was investigated using frequency spectra from the array microphones, array beamforming of the point source region, and subsequent deconvolution using the Deconvolution Approach for the Mapping of Acoustic Sources (DAMAS) algorithm. Comparisons were made with the conventional techniques for improving SNR of spectral and Cross-Spectral Matrix subtraction. The method was seen to recover the primary signal level in SNRs as low as -29 dB and outperform the conventional methods. A second processing approach using the center array microphone as the noise reference was investigated for more general applicability of the ANC technique. It outperformed the conventional methods at the -29 dB SNR but yielded less accurate results when coherence over the array dropped. This approach could possibly improve conventional testing methodology but must be investigated further under more realistic testing conditions

Application of Carbon Nanotube Thermophones as Duct Noise Cancelling Speakers: Using New Technology with Old Theories

Certain materials can produce sound through heat fluctuations. In recent times, the material capable of this was discovered to be Carbon Nanotubes (CNT). The idea is to create a flexible speaker that can be made into any shape or form to create sound. Once a speaker has been created, an active noise control system can be designed to cancel loud sounds in building and pipe ducts. This technology, similar to noise cancelling headphones, allows a user to create a modular system to mitigate noise in everyday life. Thus, the purpose of this research is to illustrate the abilities of a CNT speaker system to perform active noise control on an HVAC duct application. This paper will start with a brief introduction into CNT technology and thermophone acoustics before diverging into active noise control technology. It will illustrate planar and non –planar wave cancellation and showcase a small scale accelerated life test