This paper proposes a low complexity noise suppressor with hybrid filterbanks and adaptive time-frequency tiling. An analysis hybrid filterbank provides efficient transformation by further decomposing low-frequency bins after a coarse transformation with a short frame size. A synthesis hybrid filterbank also reduces computational complexity in a similar fashion to the analysis hybrid filterbank. Adaptive time-frequency tiling reduces the number of spectral gain calculations. It adaptively generates tiling information in the time-frequency plane based on the signal characteristics. The average number of instructions on a typical DSP chip has been reduced by 30% to 7.5 MIPS in case of mono signals sampled at 44.1 kHz. A Subjective test result shows that the sound quality of the proposed method is comparable to that of the conventional one.

In this paper, a sidelobe suppression technique for orthogonal frequency division multiplexing (OFDM)-based cognitive radios (CR) is proposed. In the OFDM-based CR systems, after the CR terminal executes spectrum sensing, it transmits a CR packet by activating the subcarriers in the frequency bands where no signals are detected (hereinafter, these subcarriers are called "active subcarrier") and by disabling (nulling) the subcarriers in the frequency bands where the signals are detected. In this situation, a problem arises in that the signals that leak from the active subcarriers to the null subcarriers may interfere with the primary systems. Therefore, this signal leakage has to be minimized. In many OFDM-based wireless communication systems, one packet or frame consists of multiple OFDM symbols and the discontinuity between the consecutive OFDM symbols causes the signal leakage to the null subcarriers. In the proposed method, signal leakage to the null subcarriers is suppressed by regenerating null subcarriers in the frequency-domain signal of the whole packet as follows. One CR packet consisting of multiple OFDM symbols having null subcarriers and guard interval (GI) is buffered and oversampled, and then the oversampled signal is Fourier transformed at once and consequently the frequency-domain signal of the packet is obtained. The null subcarriers in the frequency-domain signal are zeroed again, and then the signal is inverse Fourier transformed and transmitted. The proposed method significantly suppresses the signal leakage. The spectral power density, the peak-to-average power ratio (PAPR) and the packet error rate (PER) performances of the proposed method are evaluated by computer simulations and the effectiveness of the proposed method is shown.

This paper proposes an enhanced method for estimating the a priori Signal-to-Disturbance Ratio (SDR) to be employed in the Acoustic Echo and Noise Suppression (AENS) system for full-duplex hands-free communications. The proposed a priori SDR estimation technique is modified based upon the Two-Step Noise Reduction (TSNR) algorithm to suppress the background noise while preserving speech spectral components. In addition, a practical approach to determine accurately the Echo Spectrum Variance (ESV) is presented based upon the linear relationship assumption between the power spectrum of far-end speech and acoustic echo signals. The ESV estimation technique is then employed to alleviate the acoustic echo problem. The performance of the AENS system that employs these two proposed estimation techniques is evaluated through the Echo Attenuation (EA), Noise Attenuation (NA), and two speech distortion measures. Simulation results based upon real speech signals guarantee that our improved AENS system is able to mitigate efficiently the problem of acoustic echo and background noise, while preserving the speech quality and speech intelligibility.

The beamforming weights which can suppress the interfering signal toward out-of-cell mobile stations in downlink are designed for a time division duplexing based OFDMA system when the channel information is not perfect. The derived beamforming weights do not improve the average SINR performance monotonously with the increased transmit SNR if the inverse of the transmit SNR is used as the regularization factor of the beamforming weights and the channel information obtained by the BS to design the BF weights is not perfect. Therefore, we suggest a simple scheme to select the regularization factor. The proposed beamforming weights improve the performance monotonously with the increased transmit SNR and achieve near-optimal performance. The performance achieved by applying the beamforming weights used in uplink to downlink beamforming is also investigated.

This letter deals with robust interference suppression based on eigenanalysis interference canceller (EIC) with the joint code-aid and noise subspace-based correcting approach. It has been shown that the EIC is very sensitive to pointing error, especially when the interference number is overestimated. Based on the corrected steering angle, a proper blocking matrix of the EIC can be obtained to suppress the leakage of desired signal. Therefore, desired signal cancellation does not occur; even if the interference number is overestimated in constructing the interference subspace. Several computer simulations are provided to demonstrate the effectiveness of the proposed approach.

This paper proposes a frequency domain nulling filter and Turbo equalizer to suppress interference in the uplink of one-cell reuse single-carrier time division multiple access (TDMA) systems. In the proposed system, the desired signal in a reference cell is interfered by interference signals including adjacent-channel interference (ACI), co-channel interference (CCI), and intersymbol interference (ISI). At the transmitter, after a certain amount of spectrum is nulled considering the expected CCI, the suppressed power due to nulling is reallocated to the remaining spectrum components so as to keep the total transmit power constant. In this process, when mitigation of ACI is necessary, after a certain amount of spectrum at both edges is nulled using an edge-removal filter, the aforementioned process is conducted. At the receiver, frequency domain SC/MMSE Turbo equalizer (FDTE) is employed to suppress ISI due to spectrum nulling process in the transmitter as well as the multipath fading. Computer simulations confirm that the proposed scheme is effective in suppression of CCI, ACI and ISI in one-cell reuse single-carrier TDMA systems.

A Recessed-Channel Dual-Gate Single Electron Transistor (RCDG-SET) which has the possibility of room temperature operation is proposed. Side gates of a RCDG-SET form electrical tunneling barriers around a recessed channel, which is newly introduced. Not only gate but also a recessed channel is self aligned to source and drain. Characteristics of a RCDG-SET are compared with those of previous DG-SETs through device simulation (SILVACO). Due to a recessed channel and a self aligned structure, MOSFET current which causes low Peak-to-Valley Current Ratio (PVCR) is suppressed. This property of a RCDG-SET is expected to contribute for room temperature operation.

This paper introduces our proposed pre-FFT type MMSE-AAA for an OFDM packet transmission system to suppress sporadic interference. The AAA scheme controls an antenna weight to minimize the mean square error between its output signals of two periods with identical transmitted waveform and iterates the weight updating process in an OFDM symbol to rapidly converge the weight. The average PER performance of the proposed AAA with the presence of a sporadic inter-system/intra-system interference signal is evaluated through computer simulations that assume an exponentially decaying 12-path LOS fading channel and IEEE 802.11a data frame transmission. Simulation results show that the proposed AAA can effectively suppress sporadic inter-system interference that is irrelevant to its arrival timing. Sporadic intra-system interference can also be suppressed by the proposed AAA more efficiently than inter-system interference as long as the interference arrives between 13% and 90% of the OFDM symbol duration after the beginning of an OFDM symbol of the desired signal.

This paper proposes a VoIP (Voice over Internet Protocol) session capacity expansion method that uses periodic packet transmission suppression control for wireless LANs. The proposed method expands the VoIP session capacity of an AP without critically degrading the QoS (Quality of Service) of all stations. Simulation results show the proposed method with 0.5% packet suppression control on each station expands a VoIP session capacity by up to 5% compared to a legacy method while satisfying required QoS for all stations.

We report on the spurious suppression effect in low-microwave power transmitters by high temperature superconducting (HTS) bandpass filters (BPFs) which are promising for devices requiring BPFs with high-frequency selectivity. Some of the major issues on the power BPFs with HTS planar circuits for wireless communication applications are reviewed. As a case study for the HTS filter and its spurious suppression effect, this paper describes an example of the measured power spectrum density (PSD) on the suppression effect by one of our developed power BPFs with YBCO films for the 5 GHz band. It was designed with equivalent cascade resonators of 16 poles. We demonstrated the effect by HTS power filter in a power amplifier for the 5 GHz band.

The downlink beamforming weights which can suppress interfering signals toward out-of-cell mobile stations for a TDD-based OFDMA system are introduced. First, the downlink beamforming weights are optimally designed based on iteration. Then, the downlink beamforming weights are designed in a sub-optimal way. It is shown that the sub-optimally designed downlink beamforming weights have the same structure as that of the uplink beamforming weights which are derived based on MMSE. The performance of these schemes is compared based on the average receive SINR. The performance of a heuristic scheme which exploits uplink beamforming weights for downlink beamforming weights is also investigated.

The present paper describes quality enhancement of speech corrupted by an additive background noise in a single-channel system. The proposed approach is based on the introduction of a perceptual criterion using a frequency-weighting filter in a subtractive-type enhancement process. Although this subtractive-type method is very attractive because of its simplicity, it produces an unnatural and unpleasant residual noise. Thus, it is difficult to select fixed optimized parameters for all speech and noise conditions. A new and effective algorithm is thus developed based on the masking properties of the human ear. This newly developed algorithm allows for an automatic adaptation in the time and frequency of the enhancement system and determines a suitable noise estimate according to the frequency of the noisy input speech. Experimental results demonstrate that the proposed approach can efficiently remove additive noise related to various kinds of noise corruption.

We introduce the Rank M-type L (RM L)-filter to remove impulsive and speckle noise from corrupted images by means of use of DSP TMS320C6701.

We propose a noise suppression method based on multi-model compositions and multi-pass search. In real environments, input speech for speech recognition includes many kinds of noise signals. To obtain good recognized candidates, suppressing many kinds of noise signals at once and finding target speech is important. Before noise suppression, to find speech and noise label sequences, we introduce multi-pass search with acoustic models including many kinds of noise models and their compositions, their n-gram models, and their lexicon. Noise suppression is frame-synchronously performed using the multiple models selected by recognized label sequences with time alignments. We evaluated this method using the E-Nightingale task, which contains voice memoranda spoken by nurses during actual work at hospitals. The proposed method obtained higher performance than the conventional method.

We propose a noise suppression algorithm with the Kalman filter theory. The algorithm aims to achieve robust noise suppression for the additive white and colored disturbance from the canonical state space models with (i) a state equation composed of the speech signal and (ii) an observation equation composed of the speech signal and additive noise. The remarkable features of the proposed algorithm are (1) applied to adaptive white and colored noises where the additive colored noise uses babble noise, (2) realization of high performance noise suppression without sacrificing high quality of the speech signal despite simple noise suppression using only the Kalman filter algorithm, while many conventional methods based on the Kalman filter theory usually perform the noise suppression using the parameter estimation algorithm of AR (auto-regressive) system and the Kalman filter algorithm. We show the effectiveness of the proposed method, which utilizes the Kalman filter theory for the proposed canonical state space model with the colored driving source, using numerical results and subjective evaluation results.

To achieve high recognition performance for a wide variety of noise and for a wide range of signal-to-noise ratio, this paper presents methods for integration of four noise reduction algorithms: spectral subtraction with smoothing of time direction, temporal domain SVD-based speech enhancement, GMM-based speech estimation and KLT-based comb-filtering. In this paper, we proposed two types of combination methods of noise suppression algorithms: selection of front-end processor and combination of results from multiple recognition processes. Recognition results on the CENSREC-1 task showed the effectiveness of our proposed methods.

A method for extracting a sound signal from a particular area that is surrounded by multiple ambient noise sources is proposed. This method performs several fixed beamformings on a pair of small microphone arrays separated from each other to estimate the signal and noise power spectra. Noise suppression is achieved by applying spectrum emphasis to the output of fixed beamforming in the frequency domain, which is derived from the estimated power spectra. In experiments performed in a room with reverberation, this method succeeded in suppressing the ambient noise, giving an SNR improvement of more than 10 dB, which is better than the performance of the conventional fixed and adaptive beamforming methods using a large-aperture microphone array. We also confirmed that this method keeps its performance even if the noise source location changes continuously or abruptly.

In this paper, we propose an adaptive guard symbol insertion method for one-cell reuse TDMA cellular systems in which co-channel interference is reduced by adaptively selecting the best transmit-pulse waveform with different guard (null-) symbols according to the average error power (AEP) corresponding to signal-to-interference and noise power ratio (SINR), even though the same frequency channel is used at all base stations. Using the proposed system, current TDMA-based systems are readily extensible to one-cell reuse systems, which achieves higher spectrum efficiency. The system capacity is enhanced using the proposed method; moreover, the required qualities such as blocking probability and outage probability are retained.

This paper proposes a co-channel interference (CCI) suppression scheme employing a frequency-domain nulling filter and turbo equalizer for single-carrier uplink time division multiple access (TDMA) systems. In the proposed scheme, after the received signal is transformed into a frequency-domain signal via fast Fourier transform (FFT), CCI from an adjacent cell is suppressed by the nulling filter. Moreover, the proposed scheme employs a soft canceller and minimum mean square error (SC/MMSE) based turbo equalizer to suppress the performance degradation due to inter-symbol interference (ISI) caused by the nulling filter as well as the ISI induced by fading channel. Computer simulation confirms that the proposed scheme is effective in suppression of CCI compared to the conventional linear frequency-domain equalizer.

This letter presents a 900 MHz ZigBee RF transmitter front-end with on-chip LO suppression circuit at the output. To suppress the LO leakage at the RF output, a novel LO suppression circuit is adopted at the up-conversion mixer. The RF transmitter implemented in 0.18 µm CMOS shows more than 28 dB of LO suppression over a wide range of the baseband signal power variation.