| 1. |
- Berggren, Magnus, et al.
(författare)
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Audio Processing Solution for Video Conference Based Aerobics
- 2010
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Ingår i: 2010 Digest of Technical Papers, International Conference on Consumer Electronics. - Las Vegas : IEEE. - 9781424443147 - 9781424443161 ; , s. 407-408
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Konferensbidrag (refereegranskat)abstract
- In this paper an audio processing solution for video conference based aerobics is presented. The proposed solution leaves the workout music unaltered by separating it from the speech and processing each signal separately. The speech signal processing is also performed at a lower sample rate, which saves computational power. Real time evaluation of the system shows that high quality music as well as a good two-way communication is maintained during the aerobic session.
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| 2. |
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| 3. |
- Nordholm, Sven, et al.
(författare)
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Performance Limits in Subband Beamforming
- 2003
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Ingår i: IEEE transactions on speech and audio processing. - : IEEE. - 1063-6676 .- 1558-2353 .- 2329-9290. ; 11:3, s. 193-203
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Tidskriftsartikel (refereegranskat)abstract
- This paper analyzes subband beamforming schemes mainly aimed at speech enhancement and acoustic echo suppression applications such as hands-free telephony for both mobile and office environments, internet telephony and video conferencing. Analytical descriptions of both causal finite-length and noncausal infinite-length subband microphone array structures are given. More specifically, this paper compares finite Wiener filter performance with the noncausal Wiener solution, giving a comprehensive theoretical suppression limit. It is shown that even short filters will yield a good approximation of the infinite solution, provided that the element spacing and temporal sampling is matched to the frequency band of interest. Typically, 10-20 FIR taps are sufficient in each subband.
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| 4. |
- Schüldt, Christian, et al.
(författare)
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Adaptive filter length selection for acoustic echo cancellation
- 2009
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Ingår i: Signal Processing. - AMSTERDAM : Elsevier. - 0165-1684 .- 1872-7557. ; 89:6, s. 1185-1194
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Tidskriftsartikel (refereegranskat)abstract
- The number of coefficients in an adaptive finite impulse response filter-based acoustic echo cancellation setup is an important parameter, affecting the overall performance of the echo cancellation. Too few coefficients give undermodelling and too many cause slow convergence and an additional echo due to the mismatch of the extra coefficients. This paper proposes a method to adaptively determine the filter length, based on estimation of the mean square deviation. The method is primarily intended for identifying long non-sparse systems, such as a typical impulse response from an acoustic setup. Simulations with band limited flat spectrum signals are used for verification, showing the behavior and benefits of the proposed algorithm. Furthermore, off-line calculation using recorded speech signals show the behavior in real situations and comparison with another state-of-the-art variable filter length algorithm shows the advantages of the proposed method.
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| 5. |
- Tegborg, Viktoria, et al.
(författare)
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Experimental results of passive imaging of moving continuous broadband sound sources within a sensor field
- 2011
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Ingår i: IEEE Journal of Oceanic Engineering. - : IEEE. - 0364-9059 .- 1558-1691. ; 36:1, s. 26-37
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Tidskriftsartikel (refereegranskat)abstract
- Sparsely distributed sensors or sparse arrays can be associated with high positional accuracy and capability for large area surveillance. This paper shows, both through simulations and with real measurement data, that sparse sensor systems can be used to accomplish high-performance underwater surveillance. The paper focuses on measurement setups with several passive single hydrophones placed hundreds of meters apart in a water depth of dozen of meters. By such a setup, a sound source is more likely to move inside the array, and thus near-field processing can be considered. The sensor pairs will also be located in different directions in relation to the sound source. These two prerequisites give the possibilities to high spatial resolution. Images of the coherent sound activity for different sensor pairs are formed, and every sensor pair map is summarized into a resulting map. In this way, an arbitrary number of sound sources of a target can be resolved. Good correlation results are shown even when the sound sources are at the same distance as the baseline of the sensor pairs. This is achieved by Doppler compensating for target movement and also by whitening of the cross spectra. The analyses of the measurements also show that baseline-dependent bandwidth can improve the results. In this paper, it is indicated from experimental data that two propeller sound sources could be resolved, and in another measurement setup, the engine could be separated from the propellers.
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| 6. |
- Yermeche, Zohra, et al.
(författare)
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Bind Subband Beamforming with Time-delay Constraints for Moving Source Speech Enhancement
- 2007
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Ingår i: IEEE Transactions on Audio, Speech, and Language Processing. - : IEEE. - 1558-7916 .- 1558-7924. ; 15:8, s. 2360-2372
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Tidskriftsartikel (refereegranskat)abstract
- A new robust microphone array method to enhance speech signals generated by a moving person in a noisy environment is presented. This blind approach is based on a two-stage scheme. First, a subband time-delay estimation method is used to localize the dominant speech source. The second stage involves speech enhancement, based on the acquired spatial information, by means of a soft-constrained subband beamformer. The novelty of the proposed method involves considering the spatial spreading of the sound source as equivalent to a time-delay spreading, thus, allowing for the estimated intersensor time-delays to be directly used in the beamforming operations. In comparison to previous approaches, this new method requires no special array geometry, knowledge of the array manifold, or acquisition of calibration data to adapt the array weights. Furthermore, such a scheme allows for the beamformer to efficiently adapt to speaker movement. The robustness of the time-delay estimation of speech signals in high noise levels is improved by making use of the non-Gaussian nature of speech trough a subband Kurtosis-weighted structure. Evaluation in a real environment with a moving speaker shows promising results, with suppression levels of up to 16 dB for background noise and interfering (speech) signals, associated to a relatively small effect of speech distortion.
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| 7. |
- Berg, Magnus, et al.
(författare)
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A combined on-line PCA-ICA algorithm for blind source separation
- 2005
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Konferensbidrag (refereegranskat)abstract
- In this paper, we introduce and investigate a method combining principle component analysis (PCA) and independent component analysis (ICA) for blind source separation (BSS). A recursive method for the PCA is applied to meet the demands of a real-time application, and likewise a modified on-line version of the information maximization principle is used. The combined PCA-ICA algorithm, first reduces the dimension of the problem and then separates the signals. Evaluation of the proposed algorithm in a real room shows superior noise suppression capabilities compared to the use of PCA or ICA individually. The proposed algorithm achieves an impressive noise suppression/separation of up to 14 dB with only two microphones. Most importantly this is achieved with negligible distortion of the recovered signal. © 2005 IEEE.
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| 8. |
- Grbic, Nedelko, et al.
(författare)
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A New Pilot-Signal based Space-Time Adaptive Algorithm
- 2001
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Konferensbidrag (refereegranskat)abstract
- In the application of adaptive antenna arrays to wireless communications, a known pilot signal sequence may be used for estimating the array response at the beginning of each data frame. This pilot sequence is usually very short and conventional training methods which estimate the array response, based solely on this training sequence, may incur large estimation errors. In this paper, we propose an online modified weighted recursive least squares type of training algorithm for estimating the optimal array response by exploiting information from the whole frame of the received signal. The benefits of the proposed algorithm is that tracking of coherent noise and interference signals is substantially improved, and the overall performance is increased. Simulation results show that the proposed method offers substantial improvement when compared to the conventional least squares method.
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| 9. |
- Khan, Muhammad Gufran, et al.
(författare)
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A Doublet-Shift Transmitted Reference Scheme for Ultra-Wideband Communication Systems
- 2007
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Konferensbidrag (refereegranskat)abstract
- The transmitted reference (TR) scheme has gained attention for ultra-wideband (UWB) communications as it bypasses the complex task of channel estimation and provides a simple receiver structure. However, the low-complexity detection in this scheme is achieved at the expense of a loss in performance. To this end, this paper proposes a doublet-shift TR (DsTR) signaling and detection scheme to improve the performance of conventional TR impulse radio UWB system. The performance of the proposed scheme is compared with conventional TR signaling scheme in terms of uncoded bit error rate (BER). The simulation results validate that the doublet-shift TR scheme achieves about 1.5-2 dB better performance than the conventional TR scheme. The DsTR scheme does not require extra energy and still has the low-complexity.
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| 10. |
- Khan, Muhammad Gufran, et al.
(författare)
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Detection of Impulse Radio Ultra-Wideband Signals using Recursive Transmitted Reference Receivers
- 2007
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Konferensbidrag (refereegranskat)abstract
- For the detection of impulse radio (IR) UWB signals, complex RAKE receiver or low-complexity transmitted reference (TR) autocorrelation receivers can be used. However, there is a performance loss associated with the low-complexity TR scheme. This paper proposes recursive structures of the conventional TR and averaged TR schemes to improve the detection performance of IR-UWB signals. The performance of proposed schemes is evaluated over the standard IEEE 802.15.4a multipath channels. The performance is compared with conventional TR receivers in terms of uncoded bit-error-rate (BER), assuming that the channel is quasi-static. For averaged and recursive averaged TR schemes, the TR sequence is also slightly modified. The simulation results validate that the proposed schemes has better performance by about 2 dB than the conventional TR and averaged TR receivers.
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