470 research outputs found
On the eigenfilter design method and its applications: a tutorial
The eigenfilter method for digital filter design involves the computation of filter coefficients as the eigenvector of an appropriate Hermitian matrix. Because of its low complexity as compared to other methods as well as its ability to incorporate various time and frequency-domain constraints easily, the eigenfilter method has been found to be very useful. In this paper, we present a review of the eigenfilter design method for a wide variety of filters, including linear-phase finite impulse response (FIR) filters, nonlinear-phase FIR filters, all-pass infinite impulse response (IIR) filters, arbitrary response IIR filters, and multidimensional filters. Also, we focus on applications of the eigenfilter method in multistage filter design, spectral/spacial beamforming, and in the design of channel-shortening equalizers for communications applications
Digital filter design using root moments for sum-of-all-pass structures from complete and partial specifications
Published versio
Residual echo signal in critically sampled subband acoustic echo cancellers based on IIR and FIR filter banks
Published versio
PERFORMANCE IMPROVEMENT OF MULTICHANNEL AUDIO BY GRAPHICS PROCESSING UNITS
Multichannel acoustic signal processing has undergone major development
in recent years due to the increased complexity of current audio processing
applications. People want to collaborate through communication with the
feeling of being together and sharing the same environment, what is considered
as Immersive Audio Schemes. In this phenomenon, several acoustic
e ects are involved: 3D spatial sound, room compensation, crosstalk cancelation,
sound source localization, among others. However, high computing
capacity is required to achieve any of these e ects in a real large-scale system,
what represents a considerable limitation for real-time applications.
The increase of the computational capacity has been historically linked
to the number of transistors in a chip. However, nowadays the improvements
in the computational capacity are mainly given by increasing the
number of processing units, i.e expanding parallelism in computing. This
is the case of the Graphics Processing Units (GPUs), that own now thousands
of computing cores. GPUs were traditionally related to graphic or image
applications, but new releases in the GPU programming environments,
CUDA or OpenCL, allowed that most applications were computationally
accelerated in elds beyond graphics. This thesis aims to demonstrate
that GPUs are totally valid tools to carry out audio applications that require
high computational resources. To this end, di erent applications in
the eld of audio processing are studied and performed using GPUs. This
manuscript also analyzes and solves possible limitations in each GPU-based
implementation both from the acoustic point of view as from the computational
point of view. In this document, we have addressed the following
problems:
Most of audio applications are based on massive ltering. Thus, the
rst implementation to undertake is a fundamental operation in the audio
processing: the convolution. It has been rst developed as a computational
kernel and afterwards used for an application that combines multiples convolutions
concurrently: generalized crosstalk cancellation and equalization.
The proposed implementation can successfully manage two di erent and
common situations: size of bu ers that are much larger than the size of the
lters and size of bu ers that are much smaller than the size of the lters.
Two spatial audio applications that use the GPU as a co-processor have been developed from the massive multichannel ltering. First application
deals with binaural audio. Its main feature is that this application is able
to synthesize sound sources in spatial positions that are not included in the
database of HRTF and to generate smoothly movements of sound sources.
Both features were designed after di erent tests (objective and subjective).
The performance regarding number of sound source that could be rendered
in real time was assessed on GPUs with di erent GPU architectures. A
similar performance is measured in a Wave Field Synthesis system (second
spatial audio application) that is composed of 96 loudspeakers. The proposed
GPU-based implementation is able to reduce the room e ects during
the sound source rendering.
A well-known approach for sound source localization in noisy and reverberant
environments is also addressed on a multi-GPU system. This
is the case of the Steered Response Power with Phase Transform (SRPPHAT)
algorithm. Since localization accuracy can be improved by using
high-resolution spatial grids and a high number of microphones, accurate
acoustic localization systems require high computational power. The solutions
implemented in this thesis are evaluated both from localization and
from computational performance points of view, taking into account different
acoustic environments, and always from a real-time implementation
perspective.
Finally, This manuscript addresses also massive multichannel ltering
when the lters present an In nite Impulse Response (IIR). Two cases are
analyzed in this manuscript: 1) IIR lters composed of multiple secondorder
sections, and 2) IIR lters that presents an allpass response. Both
cases are used to develop and accelerate two di erent applications: 1) to
execute multiple Equalizations in a WFS system, and 2) to reduce the
dynamic range in an audio signal.Belloch Rodríguez, JA. (2014). PERFORMANCE IMPROVEMENT OF MULTICHANNEL AUDIO BY GRAPHICS PROCESSING UNITS [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/40651TESISPremios Extraordinarios de tesis doctorale
Bacterial Foraging Based Channel Equalizers
A channel equalizer is one of the most important subsystems in any digital
communication receiver. It is also the subsystem that consumes maximum computation
time in the receiver. Traditionally maximum-likelihood sequence estimation (MLSE) was
the most popular form of equalizer. Owing to non-stationary characteristics of the
communication channel MLSE receivers perform poorly. Under these circumstances
‘Maximum A-posteriori Probability (MAP)’ receivers also called Bayesian receivers
perform better.
Natural selection tends to eliminate animals with poor “foraging strategies” and favor the
propagation of genes of those animals that have successful foraging strategies since they
are more likely to enjoy reproductive success. After many generations, poor foraging
strategies are either eliminated or shaped into good ones (redesigned). Logically, such
evolutionary principles have led scientists in the field of “foraging theory” to
hypothesize that it is appropriate to model the activity of foraging as an optimization
process.
This thesis presents an investigation on design of bacterial foraging based channel
equalizer for digital communication. Extensive simulation studies shows that the
performance of the proposed receiver is close to optimal receiver for variety of channel
conditions. The proposed receiver also provides near optimal performance when channel
suffers from nonlinearities
Doctor of Philosophy
dissertationHearing aids suffer from the problem of acoustic feedback that limits the gain provided by hearing aids. Moreover, the output sound quality of hearing aids may be compromised in the presence of background acoustic noise. Digital hearing aids use advanced signal processing to reduce acoustic feedback and background noise to improve the output sound quality. However, it is known that the output sound quality of digital hearing aids deteriorates as the hearing aid gain is increased. Furthermore, popular subband or transform domain digital signal processing in modern hearing aids introduces analysis-synthesis delays in the forward path. Long forward-path delays are not desirable because the processed sound combines with the unprocessed sound that arrives at the cochlea through the vent and changes the sound quality. In this dissertation, we employ a variable, frequency-dependent gain function that is lower at frequencies of the incoming signal where the information is perceptually insignificant. In addition, the method of this dissertation automatically identifies and suppresses residual acoustical feedback components at frequencies that have the potential to drive the system to instability. The suppressed frequency components are monitored and the suppression is removed when such frequencies no longer pose a threat to drive the hearing aid system into instability. Together, the method of this dissertation provides more stable gain over traditional methods by reducing acoustical coupling between the microphone and the loudspeaker of a hearing aid. In addition, the method of this dissertation performs necessary hearing aid signal processing with low-delay characteristics. The central idea for the low-delay hearing aid signal processing is a spectral gain shaping method (SGSM) that employs parallel parametric equalization (EQ) filters. Parameters of the parametric EQ filters and associated gain values are selected using a least-squares approach to obtain the desired spectral response. Finally, the method of this dissertation switches to a least-squares adaptation scheme with linear complexity at the onset of howling. The method adapts to the altered feedback path quickly and allows the patient to not lose perceivable information. The complexity of the least-squares estimate is reduced by reformulating the least-squares estimate into a Toeplitz system and solving it with a direct Toeplitz solver. The increase in stable gain over traditional methods and the output sound quality were evaluated with psychoacoustic experiments on normal-hearing listeners with speech and music signals. The results indicate that the method of this dissertation provides 8 to 12 dB more hearing aid gain than feedback cancelers with traditional fixed gain functions. Furthermore, experimental results obtained with real world hearing aid gain profiles indicate that the method of this dissertation provides less distortion in the output sound quality than classical feedback cancelers, enabling the use of more comfortable style hearing aids for patients with moderate to profound hearing loss. Extensive MATLAB simulations and subjective evaluations of the results indicate that the method of this dissertation exhibits much smaller forward-path delays with superior howling suppression capability
Compensation of fibre impairments in coherent optical systems
Tese de mestrado integrado. Engenharia Electrotécnica e de Computadores. Faculdade de Engenharia. Universidade do Porto. 201
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