Infinite non-causality in active cancellation of random noise

Active cancellation of broadband random noise requires the detection of the incoming noise with some time advance. In an duct for example this advance must be larger than the delays in the secondary path from the control source to the error sensor. In this paper it is shown that, in some cases, the advance required for perfect noise cancellation is theoretically infinite because the inverse of the secondary path, which is required for control, can include an infinite non-causal response. This is shown to be the result of two mechanisms: in the single-channel case (one control source and one error sensor), this can arise because of strong echoes in the control path. In the multi-channel case this can arise even in free field simply because of an unfortunate placing of sensors and actuators. In the present paper optimal feedforward control is derived through analytical and numerical computations, in the time and frequency domains. It is shown that, in practice, the advance required for significant noise attenuation can be much larger than the secondary path delays. Practical rules are also suggested in order to prevent infinite non-causality from appearing

ULTRA-WIDEBAND NONLINEAR ECHO-CANCELLATION

Hybrid fiber coaxial (HFC) networks are used around the world to distribute cable television and broadband internet services to customers. These networks are governed by the Data-Over-Cable Service Interface Specification (DOCSIS) family of standards, with the most recent version at the time of this writing being DOCSIS 3.1. A frequency division duplex (FDD) spectrum is used in DOCSIS 3.1, where the upstream and downstream signals are separated in frequency to eliminate interference. A possible method to increase signal bandwidths is to use a full-duplex (FDX) spectrum, in which the US and DS signals use the same frequencies at the same time. A main challenge faced when implementing FDX in a DOCSIS node is eliminating the interference in the received US signal caused by the transmitted DS signal. One possible method for eliminating the interference is utilizing an echo-canceling algorithm, which predicts the self-interference (SI) based on the known DS signal and cancels it from the received US signal. Although echo-cancellation algorithms exist for fundamentally similar applications, the DOCSIS FDX case is more complicated for two main reasons. First, the DOCSIS node uses a nonlinear power amplifier to amplify the DS signal. Second, the DS signal is an ultra-wideband signal spanning a frequency range of up to 1.2 GHz. Most of the amplifier modeling techniques discussed in the literature were designed for narrowband wireless signals and will have limited performance when used with ultra-wideband signals. This thesis develops an algorithm to characterize the power amplifier and to predict the harmonics it generates for a given DS signal. These predicted harmonics can be used to cancel the SI signal in a full duplex DOCSIS system. The algorithm, which is referred to as the ultra-wideband memory polynomial (UWB-MP) model, is based on the well-known memory polynomial model with adaptations which allow the model to predict harmonics for ultra-wideband signals. Since a direct implementation of the UWB-MP model in an FPGA would result in very high resource usage, system architecture recommendations are provided. Our proposed implementation of the model compensates for harmonics up to and including the 3rd order, which has a power spectrum extending above 3600 MHz. Using the techniques discussed in this thesis, it is shown that a sampling rate of 4 GHz allows for cancellation of the SI signal while providing a reasonable balance between performance and resource usage. Matlab simulations of a DOCSIS node with various parameters and PA simulation models were conducted. The simulations showed that over 75 dB of cancellation of the SI signal is possible in an idealized hardware setup. It is also demonstrated that AWGN injected into the received signal does not reduce the ability of the model to estimate the PA harmonics, although the noise itself cannot be canceled. Further simulations showed that the UWB-MP model could cancel harmonics whose power is much higher than that specified in DOCSIS. Although the UWB-MP model was designed with memory polynomial type PAs in mind, simulation results show that significant cancellation is possible with PAs that are represented by Wiener models as well. Based on the simulation results, we recommend using a filter of length 20 coefficients for each harmonic in the UWB-MP model, and 60 iterations with 500 samples for estimating the coefficients with the least squares method

Measuring the Phase Variation of a DOCSIS 3.1 Full Duplex Channel

Including a Full Duplex option into DOCSIS introduces several problems. One of the more troublesome issues is the presence of a strong self interference signal that leaks from the transmit side to the receive side of a cable node. This self interference is caused by echoes in the channel that translate the forward travelling transmit signals into a reverse travelling signal, as well as, by leakage from the hybrid coupler used to couple the upstream and downstream signals. To suppress this self interference an echo canceller is implemented to remove the unwanted interference from the received signal. Unfortunately with the high rates of data transmission used in modern day CATV networks the echo canceller needs tremendous precision. A major concern in the implementation of Full Duplex into DOCSIS is if the channels used are even very slightly time varying. The echos in such channels change with time and can be difficult for the echo canceller to track. Changes in the response of the channel cause the echo profile of the network to shift and the echo canceler to re-adapt to the new channel response. The issue with this changing response is that it is possible for the channel to change faster than the echo canceller can adapt, resulting in the interference becoming unacceptably high. Since the channel is a physical network of coaxial cables often exposed to the environment, its propagation properties can be affected by wind swaying pole mounted cables, or by rapid heating from the sun, or sudden shifts in the load of the network. With information on how the physical properties of the cable changes, the engineers designing the echo canceller can know how fast the canceller must adapt to changes and also have a better measure of how reliable its echo cancellation will be. In this thesis the stability of the echo profile of the channel is measured. It is shown that the property of the channel with the greatest potential to rapidly change and cause noise after echo cancellation is the phase response of the channel. Due to this, the approach of this thesis is to measure the fluctuations in the phase of the channel response of a CATV network constructed in the lab. To measure the fluctuations in the phase response of the channel, a PLL (Phase Locked Loop) based circuit is designed and built on an FPGA (Field Programmable Gate Array) and connected to a model of a simple CATV network. The PLL circuit used to measure the phase fluctuations of the channel is designed to be able to measure changes occurring faster than 0.1 Hz and with a power higher than $10^{-7} \: V^2$. The circuit is able to capture data from the channel over a period of 90 seconds. Using this phase variation measurement circuit a series of experiments were performed on a model CATV DOCSIS network. It was found that many physical disturbances to the network had the effect of rapidly shifting the phase response of the network. Heating the cables in the network was found to shift the phase response upwards of $20000\:\mu$radians. Flexing the cables in the network was found to have a peak phase variation of $8000\: \mu$radians with similar effects found from walking over cables. Overall, it was clear that physical effects on the network had the propensity to rapidly shift the network response. Any echo canceller that is designed in the future will have to consider these effects when reporting the cancellation that it is able to achieve

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

Raking the Cocktail Party

We present the concept of an acoustic rake receiver---a microphone beamformer that uses echoes to improve the noise and interference suppression. The rake idea is well-known in wireless communications; it involves constructively combining different multipath components that arrive at the receiver antennas. Unlike spread-spectrum signals used in wireless communications, speech signals are not orthogonal to their shifts. Therefore, we focus on the spatial structure, rather than temporal. Instead of explicitly estimating the channel, we create correspondences between early echoes in time and image sources in space. These multiple sources of the desired and the interfering signal offer additional spatial diversity that we can exploit in the beamformer design. We present several "intuitive" and optimal formulations of acoustic rake receivers, and show theoretically and numerically that the rake formulation of the maximum signal-to-interference-and-noise beamformer offers significant performance boosts in terms of noise and interference suppression. Beyond signal-to-noise ratio, we observe gains in terms of the \emph{perceptual evaluation of speech quality} (PESQ) metric for the speech quality. We accompany the paper by the complete simulation and processing chain written in Python. The code and the sound samples are available online at \url{http://lcav.github.io/AcousticRakeReceiver/}.Comment: 12 pages, 11 figures, Accepted for publication in IEEE Journal on Selected Topics in Signal Processing (Special Issue on Spatial Audio

Spatial, Spectral, and Perceptual Nonlinear Noise Reduction for Hands-free Microphones in a Car

Speech enhancement in an automobile is a challenging problem because interference can come from engine noise, fans, music, wind, road noise, reverberation, echo, and passengers engaging in other conversations. Hands-free microphones make the situation worse because the strength of the desired speech signal reduces with increased distance between the microphone and talker. Automobile safety is improved when the driver can use a hands-free interface to phones and other devices instead of taking his eyes off the road. The demand for high quality hands-free communication in the automobile requires the introduction of more powerful algorithms. This thesis shows that a unique combination of five algorithms can achieve superior speech enhancement for a hands-free system when compared to beamforming or spectral subtraction alone. Several different designs were analyzed and tested before converging on the configuration that achieved the best results. Beamforming, voice activity detection, spectral subtraction, perceptual nonlinear weighting, and talker isolation via pitch tracking all work together in a complementary iterative manner to create a speech enhancement system capable of significantly enhancing real world speech signals. The following conclusions are supported by the simulation results using data recorded in a car and are in strong agreement with theory. Adaptive beamforming, like the Generalized Side-lobe Canceller (GSC), can be effectively used if the filters only adapt during silent data frames because too much of the desired speech is cancelled otherwise. Spectral subtraction removes stationary noise while perceptual weighting prevents the introduction of offensive audible noise artifacts. Talker isolation via pitch tracking can perform better when used after beamforming and spectral subtraction because of the higher accuracy obtained after initial noise removal. Iterating the algorithm once increases the accuracy of the Voice Activity Detection (VAD), which improves the overall performance of the algorithm. Placing the microphone(s) on the ceiling above the head and slightly forward of the desired talker appears to be the best location in an automobile based on the experiments performed in this thesis. Objective speech quality measures show that the algorithm removes a majority of the stationary noise in a hands-free environment of an automobile with relatively minimal speech distortion

A study on acoustic emission signal propagation of a small size multi-cylinder diesel engine

Acoustic emission has been found effective in offering earlier fault detection and improving identification capabilities of faults. However, the sensors are inherently uncalibrated. This paper presents a source to sensor paths calibration technique which can lead to diagnosis of faults in a small size multi-cylinder diesel engine. Preliminary analysis of the acoustic emission (AE) signals is outlined, including time domain, time-frequency domain, and the root mean square (RMS) energy. The results reveal how the RMS energy of a source propagates to the adjacent sensors. The findings lead to allocate the source and estimate its inferences to the adjacent sensor, and finally help to diagnose the small size diesel engines by minimising the crosstalk from multiple cylinders