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1.	Björnson, Emil, et al. (författare) Massive MIMO systems with hardware-constrained base stations 2014 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781479928927 ; , s. 3142-3146 Konferensbidrag (refereegranskat)abstract Massive multiple-input multiple-output (MIMO) systems are cellular networks where the base stations (BSs) are equipped with unconventionally many antennas. Such large antenna arrays offer huge spatial degrees-of-freedom for transmission optimization; in particular, great signal gains, resilience to imperfect channel knowledge, and small inter-user interference are all achievable without extensive inter-cell coordination. The key to cost-efficient deployment of large arrays is the use of hardware-constrained base stations with low-cost antenna elements, as compared to today's expensive and power-hungry BSs. Low-cost transceivers are prone to hardware imperfections, but it has been conjectured that the excessive degrees-of-freedom of massive MIMO would bring robustness to such imperfections. We herein prove this claim for an uplink channel with multiplicative phase-drift, additive distortion noise, and noise amplification. Specifically, we derive a closed-form scaling law that shows how fast the imperfections increase with the number of antennas.
2.	Adalbjörnsson, Stefan Ingi, et al. (författare) Estimating Multiple Pitches Using Block Sparsity 2013 Ingår i: [Host publication title missing]. - 1520-6149. ; , s. 6220-6224 Konferensbidrag (refereegranskat)
3.	Ahrens, Jens, 1978 (författare) Auralization of Omnidirectional Room Impulse Responses Based on the Spatial Decomposition Method and Synthetic Spatial Data 2019 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2019-May, s. 146-150 Konferensbidrag (refereegranskat)abstract The spatial decomposition method decomposes acoustic room impulse responses into a pressure signal and a direction of arrival for each time instant of the pressure signal. An acoustic space can be auralized by distributing the pressure signal over the available loudspeakers or head-related transfer functions so that the required instantaneous propagation direction is recreated. We present a user study that demonstrates based on binaural auralization that the arrival directions can be synthesized from random data such that the auralization is nearly indistinguishable from the auralization of the original data. The presented concept constitutes the fundament of a highly scalable spatialization method for omnidirectional room impulse responses.
4.	Ahrens, Jens, 1978, et al. (författare) Interpolation and Range Extrapolation of Sound Source Directivity Based on a Spherical Wave Propagation Model 2020 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2020-May, s. 4662-4666 Konferensbidrag (refereegranskat)abstract Approaches for incorporating sound source directivity into wave-based room acoustic simulations using a spherical harmonic representation have been presented recently. Normally, the directivity is measured or prescribed on a spherical surface centered at the nominal source position. In wave-based simulations, this directivity can be represented through a locally-defined driving term acting at the source location. In practice, the directivity of real-world sound sources like musical instruments or industrial machinery can only be measured approximately in terms of spatial resolution and accuracy. We show that the measurement data can be augmented such that the impairments due to the limitations of the measurement accuracy are mitigated. We revisit the previously proposed approach of only using the angle-dependent magnitude of the measured directivity together with a spherical-wave propagation model and demonstrate its potential by means of numerical simulations based on two case studies.
5.	Ahrens, Jens, 1978, et al. (författare) The Far-Field Equatorial Array for Binaural Rendering 2021 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2021-June, s. 421-425 Konferensbidrag (refereegranskat)abstract We present a method for obtaining a spherical harmonic representation of a sound field based on a microphone array along the equator of a rigid spherical scatterer. The two-dimensional plane wave de-composition of the incoming sound field is computed from the microphone signals. The influence of the scatterer is removed under the assumption of distant sound sources, and the result is converted to a spherical harmonic (SH) representation, which in turn can be rendered binaurally. The approach requires an order of magnitude fewer microphones compared to conventional spherical arrays that operate at the same SH order at the expense of not being able to accurately represent non-horizontally-propagating sound fields. Although the scattering removal is not perfect at high frequencies at low harmonic orders, numerical evaluation demonstrates the effectiveness of the approach.
6.	Alegría, Juan Vidal, et al. (författare) Cell-Free Massive MIMO: Exploiting The Wax Decomposition 2022 Ingår i: 2022 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2022 - Proceedings. - 1520-6149. - 9781665405409 ; 2022-May, s. 5393-5397 Konferensbidrag (refereegranskat)abstract Cell-free massive multiple-input multiple-output (MIMO) consists of a large set of distributed access points (APs) serving a number of users. The APs can be far from each other, and they can also have a big number of antennas. Thus, decentralized architectures have to be considered so as to reduce the interconnection bandwidth to a central processing unit (CPU) and make the system scalable. On the other hand, the APs in a heterogeneous network might have limited processing capabilities and fully-decentralized processing may not be available. In a recent paper, a trade-off between level of decentralization and decentralized processing complexity has been identified. Furthermore, a novel matrix decomposition - the WAX decomposition - which, if applicable to the channel matrix, allows for exploitation of said trade-off without loss of information. The results on WAX decomposition are only available for random channel matrices without specific structures, while in a cell-free massive MIMO scenario the channel can have sparse structures. In this work, we study the applicability of WAX decomposition to cell-free massive MIMO with its implications to the above-mentioned trade-off.
7.	Alegría, Juan Vidal, et al. (författare) Modular binary tree architecture for distributed large intelligent surface 2021 Ingår i: International Conference on Acoustics, Speech, and Signal Processing (ICASSP). - 1520-6149. - 9781728176062 - 9781728176055 ; 2021-June, s. 4865-4869 Konferensbidrag (refereegranskat)abstract Large intelligent surface (LIS) is a technology that extends massive MIMO by considering an even greater number of antennas distributed throughout vast areas. In order to be able to implement this technology, it is crucial to consider decentralized architectures so as to make the whole system scalable. We consider a LIS divided into several LIS panels of smaller size, which can be located far away from each other. We present a modular architecture that allows combining different LIS panels using a binary tree. This architecture is also valid in a cell-free massive MIMO scenario. We make use of a newly defined matrix decomposition, the WAX decomposition, to define the modules that are used within our architecture. We also study the lossless dimensionality reduction in the data to be processed, which can be achieved using our proposed architecture.
8.	Bengtsson, Tomas, 1983, et al. (författare) Regularized Optimization for Joint Super-Resolution and High Dynamoc Range Image Reconstruction in a Percetually Uniform Domain. 2012 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781467300469 ; , s. 1097-1100 Konferensbidrag (refereegranskat)abstract This paper discusses resolution enhancement of a set of images with varying exposure durations, having a high combined dynamic range. So far, little has been said in relation to the Human Visual System when it comes to Super-Resolution and High Dynamic Range fusion, unlike the case for traditional Super-Resolution where errors are measured with respect to human perception in the pixel domain. We propose a Super-Resolution method in the Lab* domain to bridge the gap and present some image reconstruction results.
9.	Bengtsson, Tomas, 1983, et al. (författare) Super-resolution reconstruction of high dynamic range images with perceptual weighting of errors 2013 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781479903566 ; , s. 2212-2216 Konferensbidrag (refereegranskat)abstract Super-Resolution and High Dynamic Range image reconstruction are two different signal processing techniques that share in common that they utilize information from multiple observations of the same scene to enhance visual image quality. In this paper, both techniques are merged in a common model, and the focus is to solve the reconstruction problem in a suitable image domain, which relates to the perception of the Human Visual System. Simulated results are presented, including a comparison with a conventional method, demonstrating the benefits of the proposed approach, in this case avoiding some severe reconstruction artifacts.
10.	Beygi, Sajjad, et al. (författare) Geometry-Based Stochastic Modeling and Estimation of Vehicle to Vehicle Channels 2014 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781479928927 ; , s. 4289-4293 Konferensbidrag (refereegranskat)abstract In this paper, a geometry-based stochastic channel model (GSCM) for vehicle-to-vehicle (V2V) wireless communica- tion is developed. The channel model reveals that the channel representation in delay-Doppler domain can be divided into four regions. In each region, the V2V channel can be modeled using a hybrid sparse/diffuse (HSD) model. Prior art on hybrid channel estimation for linear time-invariant channels is extended to the time-varying case. Furthermore, the effects of pulse shape leakage are explicitly determined and compensated. Simulation results shows that exploiting the V2V channel properties in the delay-Doppler domain, yields significantly improved channel estimates over unstructured approaches (more than 10 dB gain in SNR).
11.	Björnson, Emil, Professor, 1983-, et al. (författare) On the impact of transceiver impairments on af relaying 2013 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing. - : Institute of Electrical and Electronics Engineers (IEEE). ; , s. 4948-4952, s. 4948-4952 Konferensbidrag (refereegranskat)abstract Recently, it was shown that transceiver hardware impairments have a detrimental impact on the performance of communication systems, especially for high-rate systems. The vast majority of technical contributions in the area of relaying assume ideal transceiver hardware. This paper quantifies the impact of transceiver hardware impairments in dual-hop Amplify-and-Forward (AF) relaying, both for fixed and variable gain relays. The outage probability (OP) in this practical scenario is a function of the instantaneous end-to-end signal-to-noise-and-distortion ratio (SNDR). This paper derives closed-form expressions for the exact and asymptotic OPs under Rayleigh fading, accounting for hardware impairments at both the transmitter and the relay. The performance loss is small at low spectral efficiency, but can otherwise be very substantial. In particular, it turns out that for high signal-to-noise ratio (SNR), the instantaneous end-to-end SNDR converges to a deterministic constant, called the SNDR ceiling, which is inversely proportional to the level of impairments. This stands in stark contrast to the ideal hardware case for which the end-to-end SNDR grows without bound in the high SNR regime.
12.	Bohlin, Patrik, 1974, et al. (författare) Performance Evaluation of MIMO Communication Systems based Superimposed Pilots 2004 Ingår i: IEEE International Conference on Acoustics, Speech, and Signal Processing; Montreal, Que; Canada; 17 May 2004 through 21 May 2004. - 1520-6149. ; 4, s. IV-425-IV-428- Konferensbidrag (refereegranskat)abstract In the recent past, a number of publications have suggested superimposed pilots (SIP) for channel estimation in MIMO systems. However, the performance gain achieved by SIP compared to conventional (time-multiplexed) training is still questionable. In this paper we introduce a framework for modeling a general SIP scheme, of which the conventional scheme is a special case. Utilizing this framework, we derive a lower bound on the channel capacity to compare the performance between the two systems. It is found that in certain scenarios (high SNR, many receive antennas and short coherence times), it is beneficial to also transmit data during the training mode (i.e., use SIP). The main conclusion though, is that for most cases where these kind of schemes would be realistic, i.e. for systems with coherence lengths of more than a few symbols, the general SIP-scheme reduces to the conventional scheme. Hence, rendering the same capacity.
13.	Brynolfsson, Johan, et al. (författare) Smooth Time-Frequency Estimation using Covariance Fitting 2014 Ingår i: Acoustics, Speech and Signal Processing (ICASSP), 2014 IEEE International Conference on. - 1520-6149. ; , s. 779-783 Konferensbidrag (refereegranskat)abstract In this paper, we introduce a time-frequency spectral estimator for smooth spectra, allowing for irregularly sampled measurements. A non-parametric representation of the time dependent (TD) covariance matrix is formed by assuming that the spectrum is piecewise linear. Using this representation, the time-frequency spectrum is then estimated by solving a convex covariance fitting problem, which also, as a byproduct, provides an enhanced estimation of the TD covariance matrix. Numerical examples using simulated non-stationary processes show the preferable performance of the proposed method as compared to the classical Wigner-Ville distribution and a smoothed spectrogram.
14.	Buchberger, Andreas, 1990, et al. (författare) Learned Decimation for Neural Belief Propagation Decoders 2021 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2021-June, s. 8273-8277 Konferensbidrag (refereegranskat)abstract We introduce a two-stage decimation process to improve the performance of neural belief propagation (NBP), recently introduced by Nachmani et al., for short low-density parity-check (LDPC) codes. In the first stage, we build a list by iterating between a conventional NBP decoder and guessing the least reliable bit. The second stage iterates between a conventional NBP decoder and learned decimation, where we use a neural network to decide the decimation value for each bit. For a (128,64) LDPC code, the proposed NBP with decimation outperforms NBP decoding by 0.75dB and performs within 1dB from maximum-likelihood decoding at a block error rate of 10-4.
15.	Butt, Naveed, et al. (författare) Robust Fundamental Frequency Estimation in the Presence of Inharmonicities 2013 Ingår i: [Host publication title missing]. - 1520-6149. ; , s. 5499-5503 Konferensbidrag (refereegranskat)
16.	Castañeda, Oscar, et al. (författare) Soft-Output Finite Alphabet Equalization for mmWave Massive MIMO 2020 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2020-May, s. 1764-1767 Konferensbidrag (refereegranskat)abstract Nxt-generation wireless systems are expected to combine millimeter-wave (mmWave) and massive multi-user multiple-input multiple-output (MU-MIMO) technologies to deliver high data-rates. These technologies require the basestations (BSs) to process high-dimensional data at extreme rates, which results in high power dissipation and system costs. Finite-alphabet equalization has been proposed recently to reduce the power consumption and silicon area of uplink spatial equalization circuitry at the BS by coarsely quantizing the equalization matrix. In this work, we improve upon finite-alphabet equalization by performing unbiased estimation and soft-output computation for coded systems. By simulating a massive MU-MIMO system that uses orthogonal frequency-division multiplexing and per-user convolutional coding, we show that soft-output finite-alphabet equalization delivers competitive error-rate performance using only 1 to 3 bits per entry of the equalization matrix, even for challenging mmWave channels.
17.	Chen, Ming, 1972, et al. (författare) Adaptive Channel Prediction Based on Polynomial Phase Signals 2008 Ingår i: 2008 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP; Las Vegas, NV; United States; 31 March 2008 through 4 April 2008. - 1520-6149. - 9781424414840 ; , s. 2881-2884 Konferensbidrag (refereegranskat)abstract Motivated by recently published physics based scattering SISO and MIMO channel models, a new adaptivechannel prediction using Kalman filter based on non-stationary polynomial phase signals with time-varying amplitudes is proposed. To mitigate the influence of the time-varying amplitudes on parameter estimation, an iterative estimation using the Non-linear instantaneous LS criterion is proposed, where the number of signal components and model orders are known. The new predictoroutperforms the classical Linear Prediction and stationarysinusoidal modeling based prediction in Monte Carlo simulations.
18.	Cheng, Jie, et al. (författare) Optimal Carrier Frequency Design for Frequency Diverse Array Mimo Radar 2023 Ingår i: ICASSP 2023 - 2023 IEEE International Conference on Acoustics, Speech and Signal Processing, Proceedings. - 1520-6149. - 9781728163277 ; 2023-June Konferensbidrag (refereegranskat)abstract In this work, we introduce a novel approach for designing the transmit frequency offset scheme based on Cramér-Rao lower bound (CRLB) minimization for a frequency diverse array multiple-input multiple-output (FDA-MIMO) radar. The problem originates in non-uniform FDA radar where each frequency offset scheme derives from a specific mathematical model, but where no optimization is conducted with respect to the frequency offset design. We propose two frequency selection schemes for FDA-MIMO radar based on A-optimal minimization, both incorporating a priori knowlegdge. The worst case optimal FDA-MIMO (OFDA-MIMOW) radar forms the frequency selection by minimizing the CRLB over a grid of the unknown parameters, whereas the Bayesian optimal FDA-MIMO (OFDA-MIMO-B) radar instead minimizes the Bayesian CRLB (BCRLB). The performances of the methods are evaluated using simulated data and compared to other common frequency selection schemes.
19.	Daniel, Ondrej, et al. (författare) Blind sub-Nyquist GNSS signal detection 2016 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2016-May, s. 6575-6579 Konferensbidrag (refereegranskat)abstract A satellite navigation receiver traditionally searches for positioning signals using an acquisition procedure. In situations, in which the required information is only a binary decision whether at least one positioning signal is present or absent, the procedure represents an unnecessarily complex solution. This paper presents a different approach for the binary detection problem with significantly reduced computational complexity. The approach is based on a novel decision metric which is utilized to design two binary detectors. The first detector operates under the theoretical assumption of additive white Gaussian noise and is evaluated by means of Receiver Operating Characteristics. The second one considers also additional interferences and is suitable to operate in a real environment. Its performance is verified using a signal captured by a receiver front-end.
20.	Dimitrakakis, Christos, 1975, et al. (författare) Boosting word error rates 2005 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; V, s. V501-V504 Konferensbidrag (refereegranskat)abstract We apply boosting techniques to the problem of word error rate minimisation in speech recognition. This is achieved through a new definition of sample error for boosting and a training procedure for hidden Markov models. For this purpose we define a sample error for sentence examples related to the word error rate. Furthermore, for each sentence example we define a probability distribution in time that represents our belief that an error has been made at that particular frame. This is used to weigh the frames of each sentence in the boosting framework. We present preliminary results on the well-known Numbers 95 database that indicate the importance of this temporal probability distribution. © 2005 IEEE.
21.	Elvander, Filip, et al. (författare) On Harmonic Approximations of Inharmonic Signals 2020 Ingår i: 2020 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2020 - Proceedings. - 1520-6149. - 9781509066315 ; 2020-May, s. 5360-5364 Konferensbidrag (refereegranskat)abstract In this work, we present the misspecified Gaussian Cramér-Rao lower bound for the parameters of a harmonic signal, or pitch, when signal measurements are collected from an almost, but not quite, harmonic model. For the asymptotic case of large sample sizes, we present a closed-form expression for the bound corresponding to the pseduo-true fundamental frequency. Using simulation studies, it is shown that the bound is sharp and is attained by maximum likelihood estimators derived under the misspecified harmonic assumption. It is shown that misspecified harmonic models achieve a lower mean squared error than correctly specified unstructured models for moderately inharmonic signals. Examining voices from a speech database, we conclude that human speech belongs to this class of signals, verifying that the use of a harmonic model for voiced speech is preferable.
22.	Eriksson, Thomas, 1964, et al. (författare) Digital predistortion of concurrent multiband communication systems 2014 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781479928927 ; , s. 3918-3922 Konferensbidrag (refereegranskat)abstract In this semi-tutorial paper, we derive the general form of a predistorter for a concurrent multiband system. Further, we derive complexity-reduced forms, and we compare the resulting algorithms with previously proposed algorithms in the literature.
23.	Fascista, Alessio, et al. (författare) Low-Complexity Accurate Mmwave Positioning for Single-Antenna Users Based on Angle-of-Departure and Adaptive Beamforming 2020 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2020-May, s. 4866-4870 Konferensbidrag (refereegranskat)abstract The problem of position estimation of a mobile user equipped with a single antenna receiver using downlink transmissions is addressed. The advantages of this setup compared to the classical MIMO and uplink scenarios are analyzed in terms of achievable theoretical performance (Cramér-Rao bounds) considering a realistic power budget. Based on this analysis, a low-complexity two-step algorithm with improved localization performance is proposed, which first performs a (coarse) angle of departure estimation and then precodes the down-link signal to introduce beamforming towards the user direction. Results demonstrate that position estimation in downlink can be potentially much more accurate than in uplink, even in presence of multiple users in the system.
24.	Fascista, Alessio, et al. (författare) RIS-aided joint localization and synchronization with a single-antenna mmwave receiver 2021 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2021-June, s. 4455-4459 Konferensbidrag (refereegranskat)abstract MmWave multiple-input single-output (MISO) systems using a single-antenna receiver are regarded as a promising solution for the near future, before the full-fledged 5G MIMO will be widespread. However, for MISO systems synchronization cannot be performed jointly with user localization unless two-way transmissions are used. In this paper we show that thanks to the use of a reconfigurable intelligent surface (RIS), joint localization and synchronization is possible with only downlink MISO transmissions. The direct maximum likelihood (ML) estimator for the position and clock offset is derived. To obtain a good initialization for the ML optimization, a decoupled, relaxed estimator of position and delays is also devised, which does not require knowledge of the clock offset. Results show that the proposed approach attains the Cramér-Rao lower bound even for moderate values of the system parameters.
25.	Ferranti, Luca, et al. (författare) Multiple Offsets Multilateration: : A New Paradigm for Sensor Network Calibration with Unsynchronized Reference Nodes 2022 Ingår i: 2022 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2022 - Proceedings. - 1520-6149. - 9781665405409 ; 2022-May, s. 4958-4962 Konferensbidrag (refereegranskat)abstract Positioning using wave signal measurements is used in several applications, such as GPS systems, structure from sound and Wifi based positioning. Mathematically, such problems require the computation of the positions of receivers and/or transmitters as well as time offsets if the devices are unsynchronized. In this paper, we expand the previous state-of-the-art on positioning formulations by introducing Multiple Offsets Multilateration (MOM), a new mathematical framework to compute the receivers positions with pseudoranges from unsynchronized reference transmitters at known positions. This could be applied in several scenarios, for example structure from sound and positioning with LEO satellites. We mathematically describe MOM, determining how many receivers and transmitters are needed for the network to be solvable, a study on the number of possible distinct solutions is presented and stable solvers based on homotopy continuation are derived. The solvers are shown to be efficient and robust to noise both for synthetic and real audio data.
26.	Ferranti, Luca, et al. (författare) Sensor Networks Tdoa Self-Calibration : 2d Complexity Analysis and Solutions 2021 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781728176062 - 9781728176055 ; 2021-June, s. 4635-4639 Konferensbidrag (refereegranskat)abstract Given a network of receivers and transmitters, the process of determining their positions from measured pseudoranges is known as network self-calibration. In this paper we consider 2D networks with synchronized receivers but unsynchronized transmitters and the corresponding calibration techniques, known as Time-Difference-Of-Arrival (TDOA) techniques. Despite previous work, TDOA self-calibration is computationally challenging. Iterative algorithms are very sensitive to the initialization, causing convergence issues. In this paper, we present a novel approach, which gives an algebraic solution to two previously unsolved scenarios. We also demonstrate that our solvers produce an excellent initial value for non-linear optimisation algorithms, leading to a full pipeline robust to noise.
27.	Fröhle, Markus, 1984, et al. (författare) Channel gain prediction for multi-agent networks in the presence of location uncertainty 2016 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2016-May, s. 3911-3915 Konferensbidrag (refereegranskat)abstract Coordination among mobile agents relies on communication over a wireless channel and can thus be improved by channel prediction. We present a Gaussian process framework to learn channel parameters and predict the channel between arbitrary transmitter and receiver locations. We explicitly incorporate location uncertainty in both learning and prediction phases. Simulation results show that if location uncertainty is not modeled appropriately, it has a degenerative effect on the prediction quality.
28.	Fu, Keren, 1988, et al. (författare) Effective Small Dim Target Detection by Local Connectedness Constraint 2014 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781479928927 ; , s. 8110-8114 Konferensbidrag (refereegranskat)abstract The main drawback of conventional filtering based methods for small dim target (SDT) detection is they could not guarantee sufficient suppression ability towards trivial high frequency component which belongs to background, such as strong corners and edges. To overcome this bottleneck, this paper proposes an effective SDT detection algorithm by using local connectedness constraint. Our method provides direct control for target size, ensure high accuracy and could be easily embedded into the classical sliding-window based framework. The effectiveness of the proposed method is validated using images with cluttered background.
29.	Fu, Keren, 1988, et al. (författare) Learning full-range affinity for diffusion-based saliency detection 2016 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781479999880 ; 2016-May, s. 1926-1930 Konferensbidrag (refereegranskat)abstract In this paper we address the issue of enhancing salient object detection through diffusion-based techniques. For reliably diffusing the energy from labeled seeds, we propose a novel graph-based diffusion scheme called affinity learning-based diffusion (ALD), which is based on learning full-range affinity between two arbitrary graph nodes. The method differs from the previous existing work where implicit diffusion was formulated as a ranking problem on a graph. In the proposed method, the affinity learning is achieved in a unified graph-based semi-supervised manner, whose outcome is leveraged for global propagation. By properly selecting an affinity learning model, the proposed ALD outperforms the ranking-based diffusion in terms of accurately detecting salient objects and enhancing the correct salient objects under a range of background scenarios. By utilizing the ALD, we propose an enhanced saliency detector that outperforms 7 recent state-of-the-art saliency models on 3 benchmark datasets.
30.	Georgescu, R., et al. (författare) Particle PHD forward filter-backward simulator for targets in close proximity 2013 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781479903566 ; , s. 6387-6391 Konferensbidrag (refereegranskat)abstract In this work, we introduce the particle PHD forward filter - backward simulator (PHD-FFBSi) capable of dealing with uncertainties in the labeling of tracks that appear when tracking two targets in close proximity with measurements that do not discriminate between them. The Forward Filter Backward Simulator is a smoothing technique based on rejection sampling for the calculation of the probabilities of association between targets and tracks. The forward filter is a particle implementation of the Probability Hypothesis Density (PHD) filter that presents advantages over an SIR filter. Difficulties that arise due to the presence of target birth and death processes are addressed through modifications to the fast FFBSi. Simulations show the new particle filter of asymptotically linear complexity in the number of particles calculates correct target label probabilities at varying levels of measurement noise.
31.	Gholami, Mohammad Reza, 1976, et al. (författare) Range Based Sensor Node Localization in the Presence of Unknown Clock Skews 2013 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781479903566 ; , s. 4046-4050 Konferensbidrag (refereegranskat)abstract We deal with the positioning problem based on two-way time-ofarrival (TW-TOA) measurements in asynchronous wireless sensor networks. The optimal estimator for this problem poses a difficult global optimization problem. To avoid the drawbacks in solving the optimal estimator, we use approximations and derive linear models,which facilitate efficient solutions. In particular, we employ the least squares method and solve a general trust region subproblem to find a coarse estimate. To further refine the estimate, we linearize the measurementsand obtain a linear model which can be solved using regularized least squares. Simulation results illustrate that the proposed approaches asymptotically attain the Cram´er-Rao lower bound.
32.	Granström, Karl, 1981, et al. (författare) Detectability prediction of hidden Markov models with cluttered observation sequences 2016 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781479999880 ; 2016-May, s. 4269-4273 Konferensbidrag (refereegranskat)abstract There is good reason to model an asymmetric threat (a structured action such as a terrorist attack) as an hmm whose observations are cluttered. Recently a Bernoulli filter was presented that can process cluttered observations («transactions») and is capable of detecting if there is an hmm present, and if so, estimate the state of the HMM. An important question in this context is: when is the HMM-in-clutter problem feasible? In other words, what system properties allow for a solvable problem? In this paper we show that, given a Gaussian approximation of the pdf of the log-likelihood, approximate detection error bounds can be derived. These error bounds allow a prediction of the detection performance, i.e. a prediction of the probability of detection given an «operating point» of transaction-level false alarm rate and miss probability. Simulations show that our analysis accurately predicts detectability of such threats. Our purpose here is to make statements about what sort of threats can be detected, and what quality of observations are necessary that this be accomplished.
33.	Hanilci, C., et al. (författare) Comparing spectrum estimators in speaker verification under additive noise degradation 2012 Ingår i: Acoustics, Speech and Signal Processing (ICASSP), 2012 IEEE International Conference on. - 1520-6149. - 9781467300445 - 9781467300452 ; , s. 4769-4772 Konferensbidrag (refereegranskat)abstract Different short-term spectrum estimators for speaker verification under additive noise are considered. Conventionally, mel-frequency cepstral coefficients (MFCCs) are computed from discrete Fourier transform (DFT) spectra of windowed speech frames. Recently, linear prediction (LP) and its temporally weighted variants have been substituted as the spectrum analysis method in speech and speaker recognition. In this paper, 12 different short-term spectrum estimation methods are compared for speaker verification under additive noise contamination. Experimental results conducted on NIST 2002 SRE show that the spectrum estimation method has a large effect on recognition performance and stabilized weighted LP (SWLP) and minimum variance distortionless response (MVDR) methods yield approximately 7 % and 8 % relative improvements over the standard DFT method at -10 dB SNR level of factory and babble noises, respectively in terms of equal error rate (EER).
34.	He, Jiguang, et al. (författare) Compressed-Sensing-Based 3D Localization with Distributed Passive Reconfigurable Intelligent Surfaces 2023 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2023-June Konferensbidrag (refereegranskat)abstract In this paper, the programmable signal propagation paradigm, enabled by Reconfigurable Intelligent Surfaces (RISs), is exploited for high accuracy 3-Dimensional (3D) user localization with a single multi-antenna base station. Capitalizing on the tunable reflection capability of passive RISs, we present a two-stage user localization method leveraging the multi-reflection wireless environment. In the first stage, we deploy an off-grid Compressive Sensing (CS) approach, which is based on the atomic norm minimization, for estimating the angles of arrival associated with each RIS, which is followed, in the second stage, by a maximum likelihood location estimation initialized with a least-squares line intersection technique. The presented numerical results showcase the high accuracy of the proposed 3D localization method, verifying our theoretical Cramér Rao lower bound analysis.
35.	Helmholz, Hannes, 1990, et al. (författare) Evaluation of Sensor Self-Noise in Binaural Rendering of Spherical Microphone Array Signals 2020 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2020-May, s. 161-165 Konferensbidrag (refereegranskat)abstract Spherical microphone arrays are used to capture spatial sound fields, which can then be rendered via headphones. We use the Real-Time Spherical Array Renderer (ReTiSAR) to analyze and auralize the propagation of sensor self-noise through the processing pipeline. An instrumental evaluation confirms a strong global influence of different array and rendering parameters on the spectral balance and the overall level of the rendered noise. The character of the noise is direction independent in the case of spatially uniformly distributed noise. However, timbre of the rendered self-noise changes with head orientation in the case of spatially non-uniform noise. We determine audibility thresholds of the coloration artifact during head rotations for different array configurations in a perceptual user study.
36.	Hohnerlein, Christoph, et al. (författare) Perceptual Evaluation Of A Multiband Acoustic Crosstalk Canceler Using A Linear Loudspeaker Array 2017 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781509041176 ; , s. 96-100 Konferensbidrag (refereegranskat)abstract We propose and evaluate an acoustic crosstalk canceler for binaural sound presentation based on an 8-channel linear loudspeaker array. The system uses traditional LSFI beamforming and path estimation in the mid and low frequency range, respectively. Simulations show a channel separation of around 40 dB over the vast part of the considered frequency range. The channel separation in a real-world implementation was measured to be 10 dB - 25 dB over a broadband spectrum of 250 Hz - 9000 Hz. When comparing localization precision of virtual binaural signals to ground-truth presentation with headphones, a 21 subject study revealed significantly higher localization precision (F(1) = 23.56, p 0.001); task difficulty, timbre and spaciousness where rated the same accross both presentation methods. Robustness of the method was high and no fixation of the subjects' heads was applied.
37.	Hu, Sha, et al. (författare) Spherical Large Intelligent Surfaces 2020 Ingår i: 2020 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2020 - Proceedings. - 1520-6149. - 9781509066315 ; 2020-May, s. 8673-8677 Konferensbidrag (refereegranskat)abstract As an emerging technology and evolution that goes beyond massive multi-input multi-output (MIMO), large intelligent surface (LIS) has gained much interest. LIS acts as an electromagnetic surface that can transmit, redirect, and receive radiating signals across its entire contiguous surface. It allows for unprecedented energy-focusing, data-transmission and terminal-positioning, and can fulfill the most grand visions for future communication systems. Earlier proposed LISs are in two-dimensional (2D), i.e., planar shapes. In this paper, we extend LISs to be three-dimensional (3D) and deployed as spherical surfaces. Compared to 2D LIS, a spherical LIS has advantages in wide coverage, simple positioning technique, and flexible deployment as reflecting surface.
38.	Huang, Huiping, 1991, et al. (författare) Joint DOA Estimation and Distorted Sensor Detection Under Entangled Low-Rank and Row-Sparse Constraints 2024 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; , s. 12851-12855 Konferensbidrag (refereegranskat)abstract The problem of joint direction-of-arrival estimation and distorted sensor detection has received a lot of attention in recent decades. Most state-of-the-art work formulated such a problem via low-rank and row-sparse decomposition, where the low-rank and row-sparse components were treated in an isolated manner. Such a formulation results in a performance loss. Differently, in this paper, we entangle the low-rank and row-sparse components by exploring their inherent connection. Furthermore, we take into account the maximal distortion level of the sensors. An alternating optimization scheme is proposed to solve the low-rank component and the sparse component, where a closed-form solution is derived for the low-rank component and a quadratic programming is developed for the sparse component. Numerical results exhibit the effectiveness and superiority of the proposed method.
39.	Huang, Jinliang, et al. (författare) Impact of channel estimation error on performance of adaptive MIMO systems 2008 Ingår i: Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing. - 1520-6149. ; , s. 2865-2868 Tidskriftsartikel (refereegranskat)abstract Adaptive modulation scheme has been widely used in multiple-input multiple-output (MIMO) systems to enhance the spectral efficiency while maintaining the bit-error-rate (BER) under a target level. In this work, we investigate the performance of adaptive modulation in the presence of imperfect channel estimation and the impact of estimation noise on the spectral efficiency. The closed-form expressions for the average spectral efficiency are derived. Two MIMO schemes are considered, i.e., orthogonal space-time block codes (OSTBC) and spatial multiplexing with zero-forcing receiver (SM-ZF), and a low complexity method to enable the transmitter to switch between OSTBC and SM-ZF is utilized to achieve higher spectral efficiency than adaptive OSTBC and adaptive SM-ZF.
40.	Jacobsson, Sven, 1990, et al. (författare) MSE-Optimal 1-Bit Precoding for Multiuser MIMO Via Branch and Bound 2018 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2018-April, s. 3589-3593 Konferensbidrag (refereegranskat)abstract In this paper, we solve the sum mean-squared error (MSE)-optimal 1-bit quantized precoding problem exactly for small-to-moderate sized multiuser multiple-input multiple-output (MU-MIMO) systems via branch and bound. To this end, we reformulate the original NP-hard precoding problem as a tree search and deploy a number of strategies that improve the pruning efficiency without sacrificing optimality. We evaluate the error-rate performance and the complexity of the resulting 1-bit branch-and-bound (BB-1) precoder, and compare its efficacy to that of existing, suboptimal algorithms for 1-bit precoding in MU-MIMO systems.
41.	Jaramillo, Alfredo Esquivel, et al. (författare) Robust Fundamental Frequency Estimation in Coloured Noise 2020 Ingår i: 2020 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2020 - Proceedings. - 1520-6149. - 9781509066315 ; 2020-May, s. 741-745 Konferensbidrag (refereegranskat)abstract Most parametric fundamental frequency estimators make the implicit assumption that any corrupting noise is additive, white Gaus-sian. Under this assumption, the maximum likelihood (ML) and the least squares estimators are the same, and statistically efficient. However, in the coloured noise case, the estimators differ, and the spectral shape of the corrupting noise should be taken into account. To allow for this, we here propose two schemes that refine the noise statistics and parameter estimates in an iterative manner, one of them based on an approximate ML solution and the other one based on removing the periodic signal obtained from a linearly constrained minimum variance (LCMV) filter. Evaluations on real speech data indicate that the iteration steps improve the estimation accuracy, therefore offering improvement over traditional non-parametric fundamental frequency methods in most of the evaluated scenarios.
42.	Jiang, Fangyuan, et al. (författare) Time Delay Estimation for TDOA Self-Calibration using Truncated Nuclear Norm Regularization 2013 Ingår i: Acoustics, Speech and Signal Processing (ICASSP), 2013 IEEE International Conference on. - 1520-6149. ; , s. 3885-3889 Konferensbidrag (refereegranskat)abstract Measurements with unknown time delays are common in different applications such as microphone array, radio antenna array calibration, where the sources (e.g. sounds) are transmitted in unknown time instants. In this paper, we present a method for estimating unknown time delays from Time-Difference-of-Arrival (TDOA) measurements. We propose a novel rank constraint on a matrix depending on the measurements and the unknown time delays. The time delays are recovered by solving a truncated nuclear norm minimization problem using alternating direction method of multipliers (ADMM). We show in synthetic experiments that the proposed method recovers the time delays with good accuracy for noisy and missing data.
43.	Johansson, Eija, 1982, et al. (författare) Interpolation based on stationary and adaptive AR(1) modeling 2011 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781457705397 ; , s. 4052-4055 Konferensbidrag (refereegranskat)abstract In this paper, we describe a minimal mean square error (MMSE) optimal interpolation filter for discrete random signals. We explicitly derive the interpolation filter for a first-order autoregressive process (AR(1)), and show that the filter depends only on the two adjacent points. The result is extended by developing an algorithm called local AR approximation (LARA), where a random signal is locally estimated as an AR(1) process. Experimental evaluation illustrates that LARA interpolation yields a lower mean square error than other common interpolation techniques, including linear, spline and local polynomial approximation (LPA).
44.	Kang, Jeongwan, et al. (författare) FUNDAMENTAL PERFORMANCE BOUNDS FOR CARRIER PHASE POSITIONING IN LEO-PNT SYSTEMS 2024 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; , s. 13496-13500 Konferensbidrag (refereegranskat)abstract In this paper, we derive the Cramér-Rao bounds (CRBs) on the positioning errors for narrow-band low earth orbit positioning, navigation, and timing (LEO-PNT) systems. Fisher information analysis is performed to characterize the CRBs for errors in Doppler and carrier phase measurements. In addition, we analyze the effect of the carrier phase and Doppler on the CRB for positioning. Numerical simulations were conducted in different settings. To show the relevance of CRB, we compare our results to the maximum likelihood (ML) estimator. Our findings show that using the carrier phase significantly improves the positioning in LEO-PNT systems.
45.	Keskin, Furkan, 1988, et al. (författare) ICI-aware parameter estimation for MIMO-OFDM radar via APES spatial filtering 2021 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2021-June, s. 8248-8252 Konferensbidrag (refereegranskat)abstract We propose a novel three-stage delay-Doppler-angle estimation algorithm for a MIMO-OFDM radar in the presence of inter-carrier interference (ICI). First, leveraging the observation that spatial covariance matrix is independent of target delays and Dopplers, we perform angle estimation via the MUSIC algorithm. For each estimated angle, we next formulate the radar delay-Doppler estimation as a joint carrier frequency offset (CFO) and channel estimation problem via an APES (amplitude and phase estimation) spatial filtering approach by transforming the delay-Doppler parameterized radar channel into an unstructured form. In the final stage, delay and Doppler of each target can be recovered from target-specific channel estimates over time and frequency. Simulation results illustrate the superior performance of the proposed algorithm in high-mobility scenarios.
46.	Khan, Zulfiqar Hasan, 1976, et al. (författare) Adaptive appearance learning for visual object tracking 2011 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781457705397 ; , s. 1413-1416 Konferensbidrag (refereegranskat)abstract This paper addresses online learning of reference object distribution in the context of two hybrid tracking schemes that combine the mean shift with local point feature correspondences, and the mean shift under the Bayesian framework, respectively. The reference object distribution is built up by a kernel-weighted color histogram. The main contributions of the proposed schemes includes: (a) an adaptive learning strategy that seeks to update the reference object distribution when the changes are caused by the intrinsic object dynamic without partial occlusion/ intersection; (b) novel dynamic maintenance of object feature points by exploring both foreground and background sets; (c) integration of adaptive appearance and local point features in joint object appearance similarity and local point features correspondences-based tracker to improve [7]; (d) integration of adaptive appearance in jointappearance similarity and particle filter tracker under theBayesian framework to improve [10]. Experimental results on a range of videos captured by a dynamic/stationary camerademonstrate the effectiveness of the proposed schemes in terms of robustness to partial occlusions, tracking drifts and tightness and accuracy of tracked bounding box. Comparisons are also made with the two hybrid trackers together with 3 existing trackers.
47.	Khanzadi, M Reza, 1983, et al. (författare) Effect of Synchronizing Coordinated Base Stations on Phase Noise Estimation 2013 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781479903566 ; , s. 4938-4942 Konferensbidrag (refereegranskat)abstract In this paper, we study the problem of oscillator phase noise (PN) estimation in coordinated multi-point (CoMP) transmission systems. Specifically, we investigate the effect of phase synchronization between coordinated base stations (BSs) on PN estimation at the user receiver. In this respect, the Bayesian Cramer-Rao bound for PN estimation is derived which is a function of the level of phase synchronization between the coordinated BSs. Results show that quality of BS synchronization has a significant effect on the PN estimation.
48.	Kim, Hyowon, et al. (författare) Low-Complexity 5G Slam with CKF-PHD Filter 2020 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. ; 2020-May, s. 5220-5224 Konferensbidrag (refereegranskat)abstract In 5G mmWave, simultaneous localization and mapping (SLAM) allows devices to exploit map information to improve their position estimate. Even the most basic SLAM filter based on a Rao-Blackwellized particle filter (RBPF) combined with a probability hypothesis density (PHD) map representation exhibits high complexity. This paper proposes a new implementation method for the 5G SLAM using message passing (MP) and the cubature Kalman filter (CKF). We demonstrate that the proposed method significantly reduces the complexity while retaining the SLAM accuracy of the RBPF-PHD approach.
49.	Kim, Samuel, et al. (författare) A pitch synchronous feature extraction method for speaker recognition 2004 Ingår i: Proceedings - IEEE International Conference on Acoustics, Speech, and Signal Processing; Montreal, Que; Canada; 17 May 2004 through 21 May 2004. - 1520-6149. ; 1, s. I405-I408 Konferensbidrag (refereegranskat)abstract This paper presents a novel feature extraction method to improve the performance of speaker identification systems. The proposed feature has a form of a typical conventional feature, mel frequency cepstral coefficients (MFCC), but a flexible segmentation to reduce spectral mismatch between training and testing processes. Specifically, the length and shift size of the analysis frame are determined by a pitch synchronous method, pitch synchronous MFCC (PSMFCC). To verify the performance of the new feature, we measure the cepstral distortion between training and testing and also perform closed set speaker identification tests. With text-independent and text-dependent experiments, the proposed algorithm provides 44.3 % and 26.7 % relative improvement respectively.
50.	Klintberg, Jacob, 1994, et al. (författare) AN IMPROVED METHOD FOR PARAMETRIC SPECTRAL ESTIMATION 2019 Ingår i: ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings. - 1520-6149. - 9781479981311 ; 2019-May, s. 5551-5555 Konferensbidrag (refereegranskat)abstract One important class of problems within spectral estimation is when the signal can be well represented by a parametric model. These kind of problems can be found in many applications such as radar, sonar and wireless communication, and has therefore been extensively investigated. The main problem is to estimate frequencies and their corresponding amplitudes and damping factors from noisy measurements. One approach to this problem is to form a matrix of measurements, and then find an approximation to the range space of the matrix, with requirements that the approximation is of low rank and have a Hankel structure. From the approximation, the signal parameters can be extracted. In this work, we investigate three different methods which follows this methodology. The main contribution will be an illustration of how the problem formulation and rank constraint management affects the accuracy of the estimate. Numerical simulations indicates that a method which formulates a single convex envelope of a least squares fit to the measurement matrix and to the rank constraint jointly is more accurate than the other two investigated methods.

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