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Articles 1 - 30 of 65
Full-Text Articles in Signal Processing
Energy Efficiency And Fault Tolerance In Open Ran And Future Internet, Saish Urumkar, Byrav Ramamurthy, Sachin Sharma
Energy Efficiency And Fault Tolerance In Open Ran And Future Internet, Saish Urumkar, Byrav Ramamurthy, Sachin Sharma
Conference papers
Open Radio Access Networks (Open RAN) repre- sent a promising technological advancement within the realm of the future internet. Research efforts are currently directed towards enhancing energy efficiency and fault tolerance, which are critical aspects for both Open RAN and the future internet landscape. In the context of energy saving in Open RAN, there exists a spectrum of methods for achieving energy efficiency. These methods include the toggling of on/off states for different hardware resources such as base station units, distributed units, and radio units. Conversely, for enhancing fault tolerance in Open RAN, Software-Defined Networking (SDN) and OpenFlow based techniques …
Improving Energy Efficiency In Open Ran Through Dynamic Cpu Scheduling, Saish Urumkar, Byrav Ramamurthy, Sachin Sharma
Improving Energy Efficiency In Open Ran Through Dynamic Cpu Scheduling, Saish Urumkar, Byrav Ramamurthy, Sachin Sharma
Conference papers
Open RAN is a promising cellular technology that is currently undergoing extensive research for future wireless radio access networks. Achieving optimal energy efficiency in Open RAN poses a significant challenge. This paper introduces a CPU scheduling algorithm that specifically targets this chal- lenge by optimizing energy consumption at the base station while maintaining optimal performance levels. With the goal of minimizing energy consumption, the proposed algorithm dynamically adjusts the CPU core states, seamlessly switching between active and sleep modes based on the load conditions. To evaluate the algorithm’s effectiveness in terms of energy saving and performance, experimental testing is conducted …
Acoustic Source Localization Using Straight Line Approximations, Swarnadeep Bagchi, Ruairí De Fréin
Acoustic Source Localization Using Straight Line Approximations, Swarnadeep Bagchi, Ruairí De Fréin
Conference papers
The short paper extends an acoustic signal delay estimation method to general anechoic scenario using image processing techniques. The technique proposed in this paper localizes acoustic speech sources by creating a matrix of phase versus frequency histograms, where the same phases are stacked in appropriate bins. With larger delays and multiple sources coexisting in the same matrix, it becomes cluttered with activated bins. This results in high intensity spots on the spectrogram, making source discrimination difficult. In this paper, we have employed morphological filtering, chain-coding and straight line approximations to ignore noise and enhance the target signal features. Lastly, Hough …
The Effects Of The Transient And Performance Loss Rates On Pv Output Performance, Chibuisi Chinasaokwu Okorieimoh, Brian Norton, Michael Conlon
The Effects Of The Transient And Performance Loss Rates On Pv Output Performance, Chibuisi Chinasaokwu Okorieimoh, Brian Norton, Michael Conlon
Conference papers
Solar photovoltaic (PV) panels experience long-term performance degradation as compared to their initial performance, resulting in lower like-per-like efficiencies and performance ratios. Manufacturers of solar photovoltaic modules normally guarantee a lifespan of more than 20 years. To meet such commitments, it is important to monitor and mitigate PV module degradation during this period, as well as beyond, to recognize maintenance and repair needs. Solar PV modules degrade over time, becoming less effective, less reliable, and eventually unusable. The effects of transient and performance loss rates on the output performance of polycrystalline silicon (p-Si) solar PV modules are the focus of …
Extending Instantaneous De-Mixing Algorithms To Anechoic Mixtures, Swarnadeep Bagchi, Ruairí De Fréin
Extending Instantaneous De-Mixing Algorithms To Anechoic Mixtures, Swarnadeep Bagchi, Ruairí De Fréin
Conference papers
The AdRess algorithm separates sources that are mixed using stereo, pan-mixing in a computationally efficient way. Pan-mixing gives the sources a location in the stereo field by introducing a relative attenuation between the versions of the sources that appear on each channel. AdRess achieves separation by constructing only a frequency-attenuation matrix. We introduce a new algorithm called Delayed-AdRess (D-AdRess), where, in addition to the frequency-attenuation matrix, two other matrices namely, frequency-delay and time-delay are used to separate sources from anechoic mixtures. By anechoic mixtures, we mean mixing scenarios where both attenuation and delays are experienced by the source signals.
Power-Weighted Lpc Formant Estimation, Ruairí De Fréin
Power-Weighted Lpc Formant Estimation, Ruairí De Fréin
Conference papers
A power-weighted formant frequency estimation procedure based on Linear Predictive Coding (LPC) is presented. It works by pre-emphasizing the dominant spectral components of an input signal, which allows a subsequent estimation step to extract formant frequencies with greater accuracy. The accuracy of traditional LPC formant estimation is improved by this new power-weighted formant estimator for different classes of synthetic signals and for speech. Power-weighted LPC significantly and reliably outperforms LPC and variants of LPC at the task of formant estimation using the VTR formants dataset, a database consisting of the Vocal Tract Resonance (VTR) frequency trajectories obtained by human experts …
New Robust Lpc-Based Method For Time-Resolved Morphology Of High-Noise Multiple Frequency Signals, Jin Xu, Mark Davis, Ruairi De Frein
New Robust Lpc-Based Method For Time-Resolved Morphology Of High-Noise Multiple Frequency Signals, Jin Xu, Mark Davis, Ruairi De Frein
Conference papers
This paper introduces a new time-resolved spectral analysis method based on the Linear Prediction Coding (LPC) method that is particularly suited to the study of the dynamics of low Signal-to-noise Ratio (SNR) signals comprising multiple frequency components. One of the challenges of the time-resolved spectral method is that they are limited by the Heisenberg-Gabor uncertainty principle. Consequently, there is a trade-off between the temporal and spectral resolution. Most of the previous studies are time-averaged methods. The proposed method is a parameterisation method which can directly extract the dominant formants. The method is based on a $z$-plane analysis of the poles …
New Robust Lpc-Based Method For Time-Resolved Morphology Of High-Noise Multiple Frequency Signals, Jin Xu, Ruairí De Fréin, Mark Davis
New Robust Lpc-Based Method For Time-Resolved Morphology Of High-Noise Multiple Frequency Signals, Jin Xu, Ruairí De Fréin, Mark Davis
Conference papers
This paper introduces a new time-resolved spectral analysis method based on the Linear Prediction Coding (LPC) method that is particularly suited to the study of the dynamics of low Signal-to-noise Ratio (SNR) signals comprising multiple frequency components. One of the challenges of the time-resolved spectral method is that they are limited by the Heisenberg-Gabor uncertainty principle. Consequently, there is a trade-off between the temporal and spectral resolution. Most of the previous studies are time-averaged methods. The proposed method is a parameterisation method which can directly extract the dominant formants. The method is based on a z-plane analysis of the poles …
Remedying Sound Source Separation Via Azimuth Discrimination And Re-Synthesis, Ruairí De Fréin
Remedying Sound Source Separation Via Azimuth Discrimination And Re-Synthesis, Ruairí De Fréin
Conference papers
Commercially recorded music since the 1950s has been mixed down from many input sound sources to a two-channel reproduction of these sources. The effect of this approach is to assign sources to locations in a stereo field using a pan-position for each source. The Adress algorithm is a popular way of extracting individual music sound sources from a stereo mixture. A drawback of the Adress algorithm is that when time-frequency components in the stereo mixture are shared between two or more sources, calculating the inter-aural intensity scaling parameter for each source for that time-frequency component is challenging. We show how …
Remedying Sound Source Separation Via Azimuth Discrimination And Re-Synthesis, Ruairí De Fréin
Remedying Sound Source Separation Via Azimuth Discrimination And Re-Synthesis, Ruairí De Fréin
Conference papers
Commercially recorded music since the 1950s has been mixed down from many input sound sources to a two- channel reproduction of these sources. The effect of this approach is to assign sources to locations in a stereo field using a pan- position for each source. The Adress algorithm is a popular way of extracting individual music sound sources from a stereo mixture. A drawback of the Adress algorithm is that when time- frequency components in the stereo mixture are shared between two or more sources, calculating the inter-aural intensity scaling parameter for each source for that time-frequency component is challenging. …
Re-Annotation Of Cough Events In The Ami Corpus, Paul Leamy, Damon Berry, David Dorran, Ted Burke
Re-Annotation Of Cough Events In The Ami Corpus, Paul Leamy, Damon Berry, David Dorran, Ted Burke
Conference papers
Cough sounds act as an important indicator of an individual's physical health, often used by medical professionals in diagnosing a patient's ailments. In recent years progress has been made in the area of automatically detecting cough events and, in certain cases, automatically identifying the ailment associated with a particular cough sound. Ethical and sensitivity issues associated with audio recordings of coughs makes it more difficult for this data to be made publicly available. However, without the public availability of a reliable database of cough sounds, developments in the area of audio event detection are likely to be hampered. The purpose …
Distance-Based Cluster Head Election For Mobile Sensing, Ruairí De Fréin, Liam O'Farrell
Distance-Based Cluster Head Election For Mobile Sensing, Ruairí De Fréin, Liam O'Farrell
Conference papers
Energy-efficient, fair, stochastic leader-selection algorithms are designed for mobile sensing scenarios which adapt the sensing strategy depending on the mobile sensing topology. Methods for electing a cluster head are crucially important when optimizing the trade-off between the number of peer-to- peer interactions between mobiles and client-server interactions with a cloud-hosted application server. The battery-life of mobile devices is a crucial constraint facing application developers who are looking to use the convergence of mobile computing and cloud computing to perform environmental sensing. We exploit the mobile network topology, specifically the location of mobiles with respect to the gateway device, to stochastically …
State Acquisition In Computer Networks, Ruairí De Fréin
State Acquisition In Computer Networks, Ruairí De Fréin
Conference papers
We establish that State Acquisition should be per- formed in networks at a rate which is consistent with the rate-of-change of the element or service being observed. We demonstrate that many existing monitoring and service-level prediction tools do not acquire network state in an appropriate manner. To address this challenge: (1) we define the rate-of- change of different applications; (2) we use methods for analysis of unevenly spaced time series, specifically, time series arising from video and voice applications, to estimate the rate-of-change of these services; and finally, (3) we demonstrate how to acquire network state accurately for a number …
Ambiqual – A Full Reference Objective Quality Metric For Ambisonic Spatial Audio, Miroslaw Narbutt, Andrew Allen, Jan Skoglund, Michael Chinen, Andrew Hines
Ambiqual – A Full Reference Objective Quality Metric For Ambisonic Spatial Audio, Miroslaw Narbutt, Andrew Allen, Jan Skoglund, Michael Chinen, Andrew Hines
Conference papers
Streaming spatial audio over networks requires efficient encoding techniques that compress the raw audio content without compromising quality of experience. Streaming service providers such as YouTube need a perceptually relevant objective audio quality metric to monitor users’ perceived quality and spatial localization accuracy. In this paper we introduce a full reference objective spatial audio quality metric, AMBIQUAL, which assesses both Listening Quality and Localization Accuracy. In our solution both metrics are derived directly from the B-format Ambisonic audio. The metric extends and adapts the algorithm used in ViSQOLAudio, a full reference objective metric designed for assessing speech and audio quality. …
Tiled Time Delay Estimation In Mobile Cloud Computing Environments, Ruairí De Fréin
Tiled Time Delay Estimation In Mobile Cloud Computing Environments, Ruairí De Fréin
Conference papers
We present a tiled delay estimation technique in the context of Mobile Cloud Computing (MCC) environments. We examine its accuracy in the presence of multiple sources for (1) sub-sample delays and also (2) in the presence of phase-wrap around. Phase wrap-around is prevalent in MCC because the separation of acoustic sources may be large. We show that tiling a histogram of instantaneous phase estimates can improve delay estimates when phase-wrap around is sig- nificantly present and also when multiple sources are present. We report that error in the delay estimator is generally less than 5% of a sample, when the …
Aphonic: Adaptive Thresholding For Noise Cancellation In Smart Mobile Environments, Ruairí De Fréin
Aphonic: Adaptive Thresholding For Noise Cancellation In Smart Mobile Environments, Ruairí De Fréin
Conference papers
We propose a signal-channel, adaptive threshold selection technique for binary mask construction, namely APHONIC, (AdaPtive tHreshOlding for NoIse Cancellation) for smart mobile environments. Using this mask, we introduce two noise cancellation techniques that perform robustly in the presence of real-world interfering signals that are typically encountered by mobile users: a violin busker, a subway and busy city square sounds. We demonstrate that when the power of the time-frequency components of the voice of a mobile user does not significantly overlap with the components of the interference signal, the threshold learning and noise cancellation techniques significantly improve the Signal-to-Interference Ratio (SIR) …
Objective Assessment Of Perceptual Audio Quality Using Visqolaudio, Colm Sloan, Damien Kelly, Naomi Harte, Anil C. Kokaram, Andrew Hines
Objective Assessment Of Perceptual Audio Quality Using Visqolaudio, Colm Sloan, Damien Kelly, Naomi Harte, Anil C. Kokaram, Andrew Hines
Conference papers
Digital audio broadcasting services transmit substantial amounts of data that is encoded to minimize bandwidth whilst maximizing user quality of experience. Many large service providers continually alter codecs to improve the encoding process. Performing subjective tests to validate each codec alteration would be impractical, necessitating the use of objective perceptual audio quality models. This paper evaluates the quality scores from ViSQOLAudio, an objective perceptual audio quality model, against the quality scores of PEAQ, POLQA, and PEMO-Q on three datasets containing fullband audio encoded with a variety of codecs and bitrates. The results show that ViSQOLAudio was more accurate than all …
Bitrate Classification Of Twice-Encoded Audio Using Objective Quality Features, Colm Sloan, Damien Kelly, Naomi Harte, Anil C. Kokaram, Andrew Hines
Bitrate Classification Of Twice-Encoded Audio Using Objective Quality Features, Colm Sloan, Damien Kelly, Naomi Harte, Anil C. Kokaram, Andrew Hines
Conference papers
When a user uploads audio files to a music stream- ing service, these files are subsequently re-encoded to lower bitrates to target different devices, e.g. low bitrate for mobile. To save time and bandwidth uploading files, some users encode their original files using a lossy codec. The metadata for these files cannot always be trusted as users might have encoded their files more than once. Determining the lowest bitrate of the files allows the streaming service to skip the process of encoding the files to bitrates higher than that of the uploaded files, saving on processing and storage space. This …
Measuring And Monitoring Speech Quality For Voice Over Ip With Polqa, Visqol And P.563, Andrew Hines, Eoin Gillen, Naomi Harte
Measuring And Monitoring Speech Quality For Voice Over Ip With Polqa, Visqol And P.563, Andrew Hines, Eoin Gillen, Naomi Harte
Conference papers
There are many types of degradation which can occur in Voice over IP (VoIP) calls. Of interest in this work are degradations which occur independently of the codec, hardware or network in use. Specifically, their effect on the subjective and objec- tive quality of the speech is examined. Since no dataset suit- able for this purpose exists, a new dataset (TCD-VoIP) has been created and has been made publicly available. The dataset con- tains speech clips suffering from a range of common call qual- ity degradations, as well as a set of subjective opinion scores on the clips from 24 …
Quantized Nonnegative Matrix Factorization, Ruairí De Fréin
Quantized Nonnegative Matrix Factorization, Ruairí De Fréin
Conference papers
Even though Nonnegative Matrix Factorization (NMF) in its original form performs rank reduction and signal compaction implicitly, it does not explicitly consider storage or transmission constraints. We propose a Frobenius-norm Quantized Nonnegative Matrix Factorization algorithm that is 1) almost as precise as traditional NMF for decomposition ranks of interest (with in 1-4dB), 2) admits to practical encoding techniques by learning a factorization which is simpler than NMF’s (by a factor of 20-70) and 3) exhibits a complexity which is comparable with state-of-the-art NMF methods. These properties are achieved by considering the quantization residual via an outer quantization optimization step, in …
On Inpainting The Adress Algorithm, Derry Fitzgerald, Dan Barry
On Inpainting The Adress Algorithm, Derry Fitzgerald, Dan Barry
Conference papers
The Adress algorithm has been demonstrated to be capable of separating sound sources from instantaneous linear mixtures, provided that the sources have a unique pan position in the stereo field. However, a shortcoming of the Adress algorithm is that all time-frequency bins outside of the chosen azimuth range are set to zero, resulting in audible artifacts in the resynthesised sound. Here we show that an inpainting algorithm based on NMF is capable of estimating these missing values and improves on the results obtained using Adress only.
Vocal Separation Using Nearest Neighbours And Median Filtering, Derry Fitzgerald
Vocal Separation Using Nearest Neighbours And Median Filtering, Derry Fitzgerald
Conference papers
Recently, single channel vocal separation algorithms have been proposed which exploit the fact that most popular music can be regarded as a repeating musical background over which a locally non-repeating vocal signal is superimposed. In this paper we describe a novel vocal separator inspired by these approaches which finds the k nearest neighbours to each frame of a spectrogram of the mixture signal. The median value of these frames is then used as the estimate of the background music at the current frame. This is then used to generate a mask on the original complex-valued spectrogram before inversion to the …
User Assisted Separation Using Tensor Factorisations, Derry Fitzgerald
User Assisted Separation Using Tensor Factorisations, Derry Fitzgerald
Conference papers
Recent research has demonstrated that user assisted techniques, where the user provides a ”guide” version of the source to be separated, are capable of giving good sound source separation. Here the user sings or plays along with the target source, and the user input is used to guide the separation towards the source of interest. This is typically done in a factorisation framework, such as non-negative matrix factorisation. Here we extend such approaches to a tensor factorisation framework to deal with multichannel signals. Further, we demonstrate how this framework can be used to improve the output from other user assisted …
On The Use Of Masking Filters In Sound Source Separation, Derry Fitzgerald, Rajesh Jaiswal
On The Use Of Masking Filters In Sound Source Separation, Derry Fitzgerald, Rajesh Jaiswal
Conference papers
Many sound source separation algorithms, such as NMF and related approaches, disregard phase information and operate only on magnitude or power spectrograms. In this context, generalised Wiener filters have been widely used to generate masks which are applied to the original complex-valued spectrogram before inversion to the time domain, as these masks have been shown to give good results. However, these masks may not be optimal from a perceptual point of view. To this end, we propose new families of masks and compare their performance to generalised Wiener filter masks using three different factorisation-based separation algorithms. Further, to-date no analysis …
Distributed Formal Concept Analysis Algorithms Based On An Iterative Mapreduce Framework, Ruairí De Fréin, Biao Xu, Eric Robson, Mícheál Ó Fóghlú
Distributed Formal Concept Analysis Algorithms Based On An Iterative Mapreduce Framework, Ruairí De Fréin, Biao Xu, Eric Robson, Mícheál Ó Fóghlú
Conference papers
While many existing formal concept analysis algorithms are efficient, they are typically unsuitable for distributed implementation. Taking the MapReduce (MR) framework as our inspiration we introduce a distributed approach for performing formal concept mining. Our method has its novelty in that we use a light-weight MapReduce runtime called Twister which is better suited to iterative algorithms than recent distributed approaches. First, we describe the theoretical foundations underpinning our distributed formal concept analysis approach. Second, we provide a representative exemplar of how a classic centralized algorithm can be implemented in a distributed fashion using our methodology: we modify Ganter’s classic algorithm …
Upmixing From Mono : A Source Separation Approach, Derry Fitzgerald
Upmixing From Mono : A Source Separation Approach, Derry Fitzgerald
Conference papers
We present a system for upmixing mono recordings to stereo through the use of sound source separation techniques. The use of sound source separation has the advantage of allowing sources to be placed at distinct points in the stereo field, resulting in more natural sounding upmixes. The system separates an input signal into a number of sources, which can then be imported into a digital audio workstation for upmixing to stereo. Considerations to be taken into account when upmixing are discussed, and a brief overview of the various sound source separation techniques used in the system are given. The effectiveness …
Shifted Nmf Using An Efficient Constant-Q Transform For Monaural Sound Source Separation, Rajesh Jaiswal, Derry Fitzgerald, Eugene Coyle, Scott Rickard
Shifted Nmf Using An Efficient Constant-Q Transform For Monaural Sound Source Separation, Rajesh Jaiswal, Derry Fitzgerald, Eugene Coyle, Scott Rickard
Conference papers
Non-negative Matrix Factorisation (NMF) based algorithms have found application in monaural audio source separation due to their ability to factorize audio spectrogram into additive part-based basis functions, which typically corresponds to individual notes or chords in music. These separated basis functions are usually greater in number than the active sources, hence clustering is needed for individual source signal synthesis. Although, many attempts have been made to improve the clustering of the basis functions to sources, much research is still required in this area. Recently, Shifted NMF based methods have been proposed as a means to avoid clustering these pitched basis …
User Assisted Source Separation Using Non-Negative Matrix Factorisation, Derry Fitzgerald
User Assisted Source Separation Using Non-Negative Matrix Factorisation, Derry Fitzgerald
Conference papers
Much research has been carried out on the use of non-negative matrix factorisation for the purpose of musical sound source separation. However, a notable shortcoming of non-negative matrix factorisation is that the recovered basis functions have to be clustered to sound sources for separation to take place. This has proved to be a difficult problem to solve. As a means of overcoming this problem, we introduce an extension to non-negative matrix factorisation which allows a user to guide the sepa- ration by singing, or playing along with, the source they want to separate. This is done through the use of …
Wavelet Based Islanding Detection Of Dc-Ac Inverter Interfaced Dg Systems, Mohamed Moin Hanif, Malabika Basu, Kevin Gaughan
Wavelet Based Islanding Detection Of Dc-Ac Inverter Interfaced Dg Systems, Mohamed Moin Hanif, Malabika Basu, Kevin Gaughan
Conference papers
The increased penetration of distributed generation (DG) often connected to grid through dc-ac interface has made islanding detection an important and challenging issue to power engineers. Several methods based on passive and active detection scheme have been proposed in the literature. While passive schemes have a large non detection zone (NDZ), concern has been raised on active method due to its degrading power quality effect. This paper proposes a wavelet based passive islanding detection scheme with almost zero NDZ for dc-ac inverter interfaced grid connected DGs. The key idea is to utilize the spectral changes in the higher frequency components …
Information Hiding Using Stochastic Diffusion For The Covert Transmission Of Encrypted Images, Jonathan Blackledge
Information Hiding Using Stochastic Diffusion For The Covert Transmission Of Encrypted Images, Jonathan Blackledge
Conference papers
A principal weakness of all encryption systems is that the output data can be `seen' to be encrypted. In other words, encrypted data provides a 'flag' on the potential value of the information that has been encrypted. In this paper, we provide a novel approach to `hiding' encrypted data in a digital image. We consider an approach in which a plaintext image is encrypted with a cipher using the processes of `stochastic diffusion' and the output quantized into a 1-bit array generating a binary image cipher-text. This output is then `embedded' in a host image which is undertaken either in …