High Band Signals Research Articles

Early Detection of Pediatric Seizures in the High Gamma Band

Using data from the MIT Physionet EEG database collected at the Children's Hospital Boston, we identify a method of detecting seizures in ten pediatric patients at least thirty seconds before seizure onset by identifying significant preictal locations and their respective frequencies within the high gamma band of 30 through 100 Hz. We analyze the potential predictive performance of event-related potential analysis in this high gamma band, provide evidence that detection algorithms should take into account the varying strength of a patient's common frequency extremes, and provide evidence that patient-specific approaches to machine learning algorithms may be more successful in the detection of pediatric seizures as they are more difficult to detect than adult seizures. Using these results, machine learning detection algorithm performance on pediatric patient data, which is prone to issues from limitations within the algorithms, may be significantly improved by incorporating high gamma band signal processing at the locations identified by this process.

High‐band feature extraction for artificial bandwidth extension using deep neural network andH∞optimisation

This work aims to enhance the quality of narrowband (0–4 kHz) voice signal in terms of frequency components, i.e. missing high-frequency components in a range of 4–8 kHz. The proposed artificial bandwidth extension framework uses the H ∞ optimisation. In this context, a signal model is used to get a better representation of wideband (0–8 kHz) information of a signal. The H ∞ optimisation is used to obtain the synthesis filter for a given signal model, which is used to synthesise the high-band (4–8 kHz) signal. The discrete Fourier transform addition is performed to add the narrowband signal and estimated high-band signal for removing the leaked information from the synthesis filter and non-ideal low pass filter. Gain adjustment is performed on the estimated high-band signal to make its energy equal to the true high-band signal. Non-stationary characteristics of speech signals generate an assorted variety in synthesis filters and corresponding gain. For this, a deep neural network (DNN) is used to estimate the synthesis filter and gain by using the given narrowband information. The authors analyse the performances of the DNN model on two data sets. Objective and subjective analyses are carried out on these data sets.

Sex differences in high-level appreciation of automobile design-evoked gamma broadband synchronization

The present study was conducted to provide neuroimaging correlates for neurodesign of automobile for marketing aesthetics, using event-related spectral perturbations (ERSPs) and participant reports. Thirty men and women aged 22–27 years were presented with various 3-dimensional automobile modelling shapes (rectangular, streamlined, and round), which were cross-matched with various interior colour tones (pure hue/vivid, light, and dark tones) in the experimental conditions, i.e., rectangular exterior with a vivid tone interior. The stimuli pairs were to be rated by participants to facilitate our understanding of the emotional dimensions of automotive design qualities. Significant differences were observed in the high gamma band of 80–100 Hz in the left temporal area between the two sexes. Men elicited a stronger high gamma band signals for dark colour tone interiors and rectangular or round automobile modelling designs because of the meaningful and comprehensible signals associated with the mechanisms of working memory. In contrast, women had fewer reactions than men, and elicited higher beta-band dynamics in the anterior cingulate cortex for rectangular automobile modelling design, and higher gamma-band dynamics for light colour tone interiors, which might relate to their higher self-awareness of positive emotional reward.

Performance analysis of dual connectivity in control/user-plane split heterogeneous networks

To ensure seamless mobility, logical separation between control plane and data plane has been evolved as a promising solution for the forthcoming fifth generation (5G) cellular networks. Here, macro-cells provide control coverage using a low frequency band signal, whereas small-cells provide high data rates to the mobile terminals (MTs) over high frequency band signals. While performing frequent handovers over small-cells, throughput perceived by an MT may fall below the requested data rate. To deal with the high data rate demand of the forthcoming 5G networks, recently dual connectivity (DC) technology has been proposed for long term evolution (LTE) networks in Release 12. However, analyzing the performance of DC in control/user-plane (C/U) split LTE heterogeneous networks is quite limited in the preceding literature. In this work, we propose an analytical framework to compare the performances of DC with hard handover in C/U split architecture in terms of system throughput and saturation probability, i.e., the probability that the total demand for resources exceeds the total capacity of the serving cells. Our proposed framework explicitly considers the data rate demands of the MTs, traffic arrival pattern and channel conditions. The analytical results have also been validated against simulation results. Our analyses reveal that the performance gain of DC over traditional hard handover is actually conditional on underlying traffic load density and call arrival rates.

A Wide Tuning Range, Low Phase Noise, and Area Efficient Dual-Band Millimeter-Wave CMOS VCO Based on Switching Cores

This paper presents a millimeter-wave wide tuning range voltage-controlled oscillator (VCO) incorporating two switchable decoupled VCO cores. When the first core is switched on producing the low frequency band (LFB) signal and the second core is off, the inductors of the second core are reused to create additional buffers that pass the LFB signal to the output buffers. The generated high frequency band (HFB) signals by the second core when turned on, are directly fed to the output buffers. Producing the outputs of both VCO cores across same terminals without utilizing active/passive combiners and coupled inductors will enhance the phase noise performance of the VCO, increase its output power, and reduce the chip size. Fabricated in a 65-nm CMOS process, the VCO achieves a measured wide tuning range of 26.2% from 54.1 to 70.4 GHz while consuming 7.4-11.2-mA current from 1-V power supply. The peak measured phase noise at 10-MHz offset is -116.3 dBc/Hz and the corresponding FOMT and FOM varies from -180.96 to -191.86 dB and -172.6 to -183.5 dB, respectively. The VCO core area occupies only 0.1 ×0.395 μm <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sup> .

Fiber optic sensors for environmental monitoring

Sensors discussed are designed to register acoustic waves and micro deformations in ground. Sensor technology is based on phase modulation that occurs in optical fiber when it is influenced by outside mechanical disturbance. In our paper we consider two possible ways of registration this phase modulation: fiber optic interferometer and mode to mode interference. These methods can be incorporated with various signal processing technique to obtain maximum efficiency of fiber optic sensor under the certain conditions. It is well known that optical fibers are widely used to transmit high band signals for long distances. In this case fiber sensitivity to environmental disturbances is a bad thing. But from other side the disturbances change propagating light properties (intensity, phase, polarization, etc.). This changes can be registered in the output light and after appropriate signal processing will give information about the parameters of the outside influence on the fiber. This is an idea in brief how optical fibers can be used as physical parameters sensors.

Improved ETSI advanced front-end for automatic speech recognition using quadrature mirror filter synthesis based on robust complex speech analysis

An automatic speech recognition (ASR) is commonly used in these days. Current ASR systems perform well in ideal environment; however, it does not perform well in realistic noisy environment. As a robust ASR, ETSI has standardized Advanced Front-End (AFE) that adopts two-stage of iterative Wiener filter (IWF) to realize a speech enhancement as the front-end of ASR. In the ETSI AFE, 16 kHz speech is divided uniformly by QMF (Quadrature Mirror Filter) into lower-band and higher-band signal and only lower-band signal is emphasized by the IWF and MFCC (Mel Frequency Cepstral Coefficient) is extracted from both the second-stage of emphasized lower-band signal and non-emphasized higher-band signal. FFT is used to estimate speech spectrum that designs the Wiener filter. On the other hand, we have already proposed robust complex speech analysis for an analytic signal. It can estimate more robust and more accurate speech spectrum due to the introduced robust criterion and nature of analytic signal. This paper proposes an improved AFE using wide-band robust ELS (Extended Least Square) complex analysis and real-valued analysis instead of FFT. Moreover, the QMF synthesis is also introduced for second-stage of wide-band analysis. The experimental results using the CENSREC-2 speech database demonstrate that the performance is improved.

Bilateral image denoising in the Laplacian subbands

Abstract This paper presents an image denoising algorithm, which applies bilateral filtering (BLF) in the Laplacian subbands. It is noted that the subband images have wider area of photometric similarity than the original, and hence, they can be more benefited by the BLF than the original. Specifically, an image is Gaussian filtered to obtain a low band image, and the low band image is subtracted from the original to have the high band signal, which forms the Laplacian subbands. For the high band image denoising, we derive an adaptive kernel that is dependent on the edge intensity and photometric similarity of subband images. The high band image is convolved with this kernel and then added to the denoised low band signal, which produces the denoised image. We also propose to process the denoised high band signal by the gradient histogram preservation method, for sharpening the edges with less noise amplification. Experimental results show that the proposed denoising method provides higher PSNR than the original BLF and other multi-resolution denoising algorithms. Since the high band image is also effectively denoised in this process, the sharpened image by high band modification is also visually more pleasing when compared with the results of the conventional sharpening methods.

Speech Bandwidth Extension Using Temporal Envelope Modeling

Speech bandwidth extension (SBE) assumes that high-frequency components of a speech signal, e.g., the frequency band of 4-7 kHz, can be estimated by parameters extracted from the narrowband signal (0-4 kHz). Therefore, it is very important to understand the characteristics of the highband signal as well as perceptual cues to represent the highband signal. This letter proposes a new SBE algorithm using a temporal envelope model. The temporal envelope model considers band-limited temporal envelopes as the perceptual cue of the 4-7 kHz band signal while it deemphasizes the importance of rapidly varying components. To implement the SBE with no additional bits, the proposed method adopts a Gaussian mixture model (GMM) to estimate the temporal envelope of the highband signal from that of the narrowband one. Simulation results confirm that the proposed SBE algorithm shows better perceptual quality than a conventional source-filter model-based approach.

A 0.9 V 96 $\mu$W Fully Operational Digital Hearing Aid Chip

A 0.9 V 96 muW fully operational low-power digital hearing aid chip is proposed and implemented. An internal status controller is introduced to achieve full operation of the adaptive-SNR analog front end. Dedicated DSP with an additional volume control parameter eliminates any internal overflow and enables the hearing aid to be customized for each individual user. When the input audio band is split into a low band and a high band, the audio signal can be processed coarsely. In addition, fine processing of the high-band signal can be obtained with a low-power automatic gain control (AGC) comprising a digital comparator and a subtraction unit. A heterogeneous Sigma-Delta DAC reduces the power consumption of the interpolation filter without degrading performance by allowing different frequencies between the input signal and the sampling clock of the Sigma-Delta modulator. Compared with a conventional Sigma-Delta DAC, the heterogeneous Sigma-Delta DAC reduces the power dissipation by 40.4% and the area occupation by 40.5%, and it has a reported error rate of only 0.16%. The fabricated chip achieves a 79 dB peak SNR with 4.1 muVrms of input-referred noise voltage. The core area is 2.8 mm x 1.1 mm in a 0.18 mum standard CMOS process.

Perceptual relevance of the temporal envelope to the speech signal in the 4–7kHz band

The perceptual relevance of adopting the temporal envelope to model the frequency band of 4-7 kHz (highband) in wideband speech signal is described in this letter. Based on theoretical work in psychoacoustics, we find out that the temporal envelope can indeed be a perceptual cue for the high-band signal, i.e., a noiseless sound can be obtained if the temporal envelope is roughly preserved. Subjective listening tests verify that transparent quality can be obtained if the model is used for the 4.5-7 kHz band. The proposed model has the benefits of offering flexible scalability and reducing the cost for quantization in coding applications.

Analysis and design of a fully integrated CMOS low-noise amplifier for concurrent dual-band receivers

This article thoroughly analyzes a concurrent dual-band low-noise amplifier (LNA) and carefully examines the effects of both active and passive elements on the performance of the dual-band LNA. As an example of the analysis, a fully integrated dual-band LNA is designed in a standard 0.18-μm 6M1P CMOS technology from the system viewpoint for the first time to provide a higher gain at the high band in order to compensate the high-band signal's extra loss over the air transmission. The LNA drains 6.21 mA of current from a 1.5-V supply voltage and achieves voltage gains of 14 and 22 dB, input S11 of 15 and 18 dB, and noise figures of 2.45 and 2.51 dB at 2.4 and 5.2 GHz, respectively. © 2006 Wiley Periodicals, Inc. Int J RF and Microwave CAE, 2006.

Apparatus and method for outputting audio signal of laptop computer coupled with docking station

An apparatus and a method for outputting an audio signal of a laptop computer to a first speaker in the laptop computer and a to second speaker in a docking station are provided. The apparatus includes a first controller implemented in the laptop computer for filtering a high frequency band component and a low frequency band component from the audio signal, for mixing the filtered signals, and for outputting the mixed signal to the first speaker when the laptop computer is not coupled with the docking station, and otherwise, filtering only a high frequency band component from the audio signal and outputting the filtered signal to the first speaker. A second controller is implemented in the docking station for receiving the audio signal from the laptop computer, for filtering a low frequency band component from the received audio signal, and for outputting the filtered signal to the second speaker when the laptop computer is coupled with the docking station. When the laptop computer is coupled with the docking station, the audio signal generated in the laptop computer is divided into high and low frequency band signals, and each divided signal is output to different speakers according to the characteristics of the divided signals. Thus, efficiency in use of the speakers and sound quality are improved.

A fully integrated concurrent dual‐band CMOS low‐noise amplifier

AbstractThis paper presents a fully integrated concurrent dual‐band CMOS low‐noise amplifier (LNA). The LNA is implemented in a standard 0.18‐μm 6M1P CMOS process and is designed from the system viewpoint to provide higher gain at the higher band, for the first time, to compensate the higher‐band signal's extra loss over the air transmission. The LNA drains 5.3 mA of current from a 1.8‐V supply voltage and achieves voltage gains of 16 and 22.5 dB, input return losses of 28 and 27 dB, and noise figures of 3.0 and 3.1 dB at 2.4 and 5.2 GHz, respectively. © 2003 Wiley Periodicals, Inc. Microwave Opt Technol Lett 39: 52–53, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/mop.11124

Audio processing apparatus

An audio processing apparatus is arranged to be capable of attenuating only an unnecessary wind noise without cutting off the low frequency part of sounds picked up. In the apparatus, a first low frequency band signal and a first high frequency band signal are extracted from a signal obtained by subtracting a delayed right-side audio signal from a received left-side audio signal. A second low frequency band signal and a second high frequency band signal are extracted from a signal obtained by subtracting a delayed left-side audio signal from a received right-side audio signal. An addition signal is formed by adding together the extracted first low frequency band signal and the extracted second low frequency band signal. The extracted first low frequency band signal and the addition signal are added together in an arbitrary variable addition ratio to obtain a first variable addition signal. The extracted second low frequency band signal and the addition signal are also added together in an arbitrary variable addition ratio to obtain a second variable addition signal. The extracted first high frequency band signal is outputted with the first variable addition signal added thereto, while the extracted second high frequency band signal is outputted with the second variable addition signal added thereto.

Design and implementation of a parametric speech coder

Speech coders based on linear predictive coding are good and accurate in modeling speech utterances. They not only model speech utterance in an accurate manner but have the following properties: i) they provide a good estimate of the vocal tract spectral envelope; ii) they are analytically tractable; iii) they can be implemented in either software or hardware; iv) they use less data storage than many other approaches of speech coders. In this paper, a hybrid speech coder based on code excited linear predictive coding (CELPC) and voice excited linear predictive coding (VELPC) is presented. CELPC is one of the main techniques for producing high quality speech at around 4.8 kbps. However, the computation requirement for the original CELPC was too demanding and not suitable for practical use. Therefore, a hybrid approach to produce speech signals with high quality and with a bit rate of 3.478 kbps has been proposed. This method splits the speech signal into two portions, the base-band signal and the high-band signal in the frequency domain. The base-band and the high-band signal are then coded using the CELPC and VELPC techniques, respectively.

Decomposition of high-band signals in a reversible subband coding by using similarity of bands

Decomposition of high-band signals in a reversible subband coding by using similarity of bands

Generation of wideband speech from bandlimited speech by use of an autocorrelation function

A high quality of speech sound is requested in speech communication circuits. This requirement, however, is not satisfied in telephone and mobile communication circuits. This paper proposes a new method to generate a lower speech signal from bandlimited speech (300–3400 Hz). A short-time autocorrelation function (ACF) is calculated on a speech waveform where the amplitude is compressed with a square root. A period is determined from the ACF. The period of the ACF is employed as an output signal. This signal is low-pass filtered and added to the original bandlimited signal. On the other hand, the higher band signal is synthesized by a pitch excitation formant synthesizer with fixed formant frequencies. The synthesized signal is filtered with BPF (3.4–7 kHz) and added to the original. Those processes are repeated with every pitch period. Short sentences uttered by two males and two females are assessed with a preference test. Synthesized speech is compared with bandlimited natural speech where the lowest frequency is varied between 0 and 300 Hz. The hearing test reveals that the proposed method provides a speech signal that is equivalent to the bandlimited signal where the limit of low frequency is 100–200 Hz.

Speech bandwidth extension method and apparatus

A speech bandwidth extension method and apparatus analyzes narrowband speech sampled at 8 kHz using LPC analysis to determine its spectral shape and inverse filtering to extract its excitation signal. The excitation signal is interpolated to a sampling rate of 16 kHz and analyzed for pitch control and power level. A white noise generated wideband signal is then filtered to provide a synthesized wideband excitation signal. The narrowband shape is determined and compared to templates in respective vector quantizer codebooks, to select respective highband shape and gain. The synthesized wideband excitation signal is then filtered to provide a highband signal which is, in turn, added to the narrowband signal, interpolated to the 16 kHz sample rate, to produce an artificial wideband signal. The apparatus may be implemented on a digital signal processor chip.

Method and apparatus for extracting isolated speech word

An isolated speech word is extracted from an input speech signal. First, the speech signal is divided into a high frequency band signal and a low frequency band signal. Next, the power levels of each of the high and low frequency band signals are independently compared with respective threshold levels. Finally, the front and the end of the speech word are detected from these independent comparisons.