This letter mainly focuses on improving current noise reduction methods to solve the critical speech distortion problems with robust noise reduction in noisy speech signals for speech enhancement over IP networks. For robust noise reduction with packet loss recovery, we propose a novel optimized Wiener filtering technique that uses the estimated SNR (Signal-to-Noise Ratio) with packet loss recovery method which is applied as post-filtering over IP-networks. Simulation results demonstrate that the proposed scheme provides better reduction and recovery rates with considering packet loss and SNR environment than other methods.

FeedForward (FF-) Blind Source Separation (BSS) systems have some degree of freedom in the solution space. Therefore, signal distortion is likely to occur. First, a criterion for the signal distortion is discussed. Properties of conventional methods proposed to suppress the signal distortion are analyzed. Next, a general condition for complete separation and distortion-free is derived for multi-channel FF-BSS systems. This condition is incorporated in learning algorithms as a distortion-free constraint. Computer simulations using speech signals and stationary colored signals are performed for the conventional methods and for the new learning algorithms employing the proposed distortion-free constraint. The proposed method can well suppress signal distortion, while maintaining a high source separation performance.

This letter deals with adaptive array beamforming based on a minimum variance distortionless response (MVDR) technique with robust capabilities for code-division multiple access signals. It has been shown that the MVDR beamformer suffers from the drawback of being very sensitive to pointing error over the eigenspace-based beamformers. For the purpose of efficient estimation and calibration, a highly efficient approach has been proposed that is implemented on polynomial rooting rather than spectral searching. However, this rooting method is suboptimal in the presence of the noise and multiple access interference (MAI). In this letter, we propose an improved polynomial rooting calibration method that is robust in both of the low signal-to-noise ratio and large MAI scenarios. Several computer simulations are provided for illustrating the effectiveness of the proposed method.

Recently, a lot of UWB antennas have been reported by many research groups. Most of the reported antennas have omnidirectional radiation characteristics. The disadvantage of using omnidirectional antennas is that the antenna performance can be degraded by adjacent walls or metals. If unidirectional UWB antennas are utilized, the degradation on the antenna performance due to omnidirectionality can be avoided. Another important topic in UWB antennas is the waveform distortion caused by antennas' transmission characteristics. In impulse-based UWB communications, waveform distortions of transmitted and received pulses caused by antennas deteriorate the communication performance. Therefore, the development of UWB antennas having small waveform distortions is highly desirable. In this paper, we propose a novel bowtie antenna using leaf-shaped radiating elements and a flat reflector. This antenna has unidirectional radiation patterns over the frequency range of 3.0 to 10.5 GHz. The actual gain in the maximum radiation direction is 6.0-9.0 dBi in the frequency range of 4.5-9.4 GHz (relative bandwidth of 71%). The cross-correlations between source pulse and received pulse waveforms are 0.89-0.94, and hence the waveform distortion caused by this antenna is relatively small. As a result, the proposed antenna is useful for impulse-based UWB communication systems using correlation detection.

This paper proposes a new binary motion estimation algorithm that improves the motion vector accuracy by using a hybrid distortion measure. Unlike conventional binary motion estimation algorithms, the proposed algorithm considers the sum of absolute difference (SAD) as well as the sum of bit-wise difference (SBD) as a block-matching criterion. In order to reduce the computational complexity and remove additional memory accesses, a new scheme is used for SAD calculation. This scheme uses 8-bit data of the lowest layer already moved into the local buffer to calculate the SAD of other higher binary layer. Experimental results show that the proposed algorithm finds more accurate motion vectors and removes the blockishness of the reconstructed video effectively. We applied this algorithm to existing video encoder and obtained noticeable visual quality enhancement.

3D video, which consists of a sequence of mesh models, can reproduce dynamic scenes containing 3D information. To summarize 3D video, a key frame extraction method is developed using rate-distortion (R-D) trade-off. For this purpose, an effective feature vector is extracted for each frame. Shot detection is performed using the feature vectors as a preprocessing followed by key frame extraction. Simple but reasonable definitions of rate and distortion are presented. Based on an assumption of linearity, an R-D curve is generated in each shot, where the locations of the key frames are optimized. Finally, R-D trade-off can be achieved by optimizing a cost function using a Lagrange multiplier, where the number of key frames is optimized in each shot. Therefore, our system will automatically determine the best locations and the number of key frames in the sense of R-D trade-off. Our experimental results show the extracted key frames are compact and faithful to the original 3D video.

In this paper, we propose a statistical approach to improve the performance of spectral quantization of speech coders. The proposed techniques compensate for the distortion in a decoded line spectrum pair (LSP) vector based on a statistical mapping function between a decoded LSP vector and its corresponding original LSP vector. We first develop two codebook-based probabilistic matching (CBPM) methods by investigating the distribution of LSP vectors. In addition, we propose an iterative procedure for the two CBPMs. Next, the proposed techniques are applied to the predictive vector quantizer (PVQ) used for the IS-641 speech coder. The experimental results show that the proposed techniques reduce average spectral distortion by around 0.064 dB and the percentage of outliers compared with the PVQ without any compensation, resulting in transparent quality of spectral quantization. Finally, the comparison of speech quality using the perceptual evaluation of speech quality (PESQ) measure is performed and it is shown that the IS-641 speech coder employing the proposed techniques has better decoded speech quality than the standard IS-641 speech coder.

This paper proposes a design method of a Doherty amplifier, which can determine the most efficient backed-off point of the amplifier by adjusting a load modulation parameter. The parameter is defined through the design of output transmission line of a carrier and a peak amplifier using a virtual open stub technique. This paper describes the design results using the technique to optimize efficiency of a Doherty amplifier for an orthogonal frequency division multiplexing (OFDM) signal, and parameter adjustment for a linearized Doherty amplifier using an adaptive digital predistortion (ADPD). Applying this method, the developed 250 W ADPD Doherty amplifier has achieved drain efficiency of 43.4% and intermodulation (IM) distortion of -48.3 dBc with output power of 44.1 dBm (10.1 dB output backed-off) at 563 MHz using an OFDM signal for integrated services digital broadcasting-terrestrial (ISDB-T).

A new method for data hiding in binary images based on block pattern coding and dynamic programming with distortion-minimizing capabilities is proposed. Up to three message data bits can be embedded into each 22 block in an input image by changing the block's pixel pattern into another, which represents the value of the message data bits as a code according to a block pattern encoding table. And extraction of hidden message data is accomplished by block pattern decoding. To minimize the resulting image distortion, two optimization techniques are proposed. The first is to use multiple block pattern encoding tables, from which an optimal one is selected specifically for each input image, and the second is to use a dynamic programming algorithm to divide the message data into bit segments for optimal embedding in a sense of minimizing the number of binary bit flippings. Accordingly, not only more data bits can be embedded in an image block on the average, but the resulting image distortion is also reduced in an optimal way. Experimental results are also included to show the effectiveness of the proposed approach.

In this paper, we propose a low complexity realization method for compensating for nonlinear distortion. Generally, nonlinear distortion is compensated for by a linearization system using a Volterra kernel. However, this method has a problem of requiring a huge computational complexity for the convolution needed between an input signal and the 2nd-order Volterra kernel. The Simplified Volterra Filter (SVF), which removes the lines along the main diagonal of the 2nd-order Volterra kernel, has been previously proposed as a way to reduce the computational complexity while maintaining the compensation performance for the nonlinear distortion. However, this method cannot greatly reduce the computational complexity. Hence, we propose a subband linearization system which consists of a subband parallel cascade realization method for the 2nd-order Volterra kernel and subband linear inverse filter. Experimental results show that this proposed linearization system can produce the same compensation ability as the conventional method while reducing the computational complexity.

This paper describes two different types of GaAs-HBT compatible, base-collector diode 0/20-dB step attenuators--diode-linearizer type and harmonics-trap type--for 3.5-GHz-band wireless applications. The two attenuators use an AC-coupled, stacked type diode switch topology featuring high power handling capability with low bias current operation. Compared to a conventional diode switch topology, this topology can improve the capability of more than 6 dB with the same bias current. In addition, successful incorporation of a shunt diode linearizer and second- and third-harmonic traps into the attenuators gives the IM3 distortion improvement of more than 7 dB in the high power ranging from 16 dBm to 18 dBm even in the 20-dB attenuation mode when IM3 distortion levels are basically easy to degrade. Measurement results show that both the attenuators are capable of delivering power handling capability (P0.2 dB) of more than 18 dBm with IM3 levels of less than -35 dBc at an 18-dBm input power while drawing low bias currents of 3.8 mA and 6.8 mA in the thru and attenuation modes from 0/5-V complementary supplies. Measured insertion losses of the linearizer-type and harmonics-trap type attenuators in the thru mode are as low as 1.4 dB and 2.5 dB, respectively.

Contact resistance is caused by constriction resistance and film resistance through contact layers. It is well known that a surface film causes non-linear voltage and current characteristics. The origin of non-linearity is caused by tunneling electron through thin insulation barrier or jumping over the thick barrier (Shottky barrier) on the contact surface. In this paper, a new idea causing nonlinear property by only current constriction which flows through very small contact spot area, if there is no film layer, is proposed by the two dimensional contact model. The contact model, used in this paper, is a two dimensional type narrow path of contact area (short bridge) made by thin copper foil of 0.035 mm on a glass epoxy resin board. The contact part is made by scraping with an electric drill as a single bridge shape of 0.1 mm wide and 0.3 mm long on the centre of a board (100 mm100 mm). The 3rd harmonic distortion voltage was measured by using a Component Linearity Test Equipment (Type CLT1 made by Radiometer Electronics Company) which the system supplies a pure sine wave current of 10 kHz and detects a distortion voltage of 30 kHz by a narrow band pass filter circuit. The sensitivity of the Component Linearity Test Equipment (CLT1) is under a 10-9 volt. Four bridge samples were examined for the comparison of nonlinear distortion voltage. The distortion voltage of a sample (A) (0.1 mm wide, 0.3 mm long) is too larger than the one of the sample (B) (0.2 mm wide, 0.3 mm long) at the same applied voltage which resistance is not so different each other. It seems that current constriction to the spot (A) may heat up higher and cool down lower than (B). It would be also guessed that the power dissipation of 20 kHz cause temperature oscillation of 20 kHz, then it causes a component of contact resistance of 20 kHz, and therefore the product of 10 kHz current and 20 kHz resistance component cause 30 kHz component distortion voltage.

This paper presents a novel technique to compensate intermodulation distortion of a self-heterodyne direct conversion OFDM system in multipath propagation environments. A self-heterodyne direct conversion system has an advantage that simple receivers can be built that are completely immune to any phase noise or frequency offset. This system, however, has a disadvantage that the nonlinear square-law detector at the receiver of the self-heterodyne direct conversion system gives rise to second order intermodulation distortion. In this study, channel estimation is performed using a training sequence and then the predistortion coefficients with regard to estimated channel parameters are derived to compensate the receiver nonlinearity. Transmit power distribution is employed to overcome multipath fading channels as well. Computer simulation demonstrates that the proposed approach improves the BER performance of the self-heterodyne direct conversion OFDM system in a multipath fading channel. This scheme gives advantage to multi-carrier systems that are much more sensitive to frequency and phase error than single-carrier systems.

A new method of cancellation of IM3 using current feedback has been proposed for a multi-stage RFIC amplifier. In order to cancel the IM3 present in an output signal of the amplifier, the IIP3 level and IM3 phase of the amplifier are adjusted by means of feedback circuit techniques, so that the target specification is satisfied. By estimating the IIP3 level and IM3 phase variations for two states in situations with and without feedback possessing linear factors, the parameters of a feedback circuit can be calculated. To confirm the validity of the method, we have investigated two approaches; one including an analytical approach to designing a two-stage feedback amplifier, achieving an IIP3 level improvement of 14.8 dB. The other method involves the fabrication of single-stage amplifiers with and without feedback, operating at 850 MHz, both of which were designed as an integrated circuit using a 0.18 µm SiGe BiCMOS process. The fabricated IC's were tested using a load-pull measurement system, and a good agreement between the estimated and measured IIP3 level and IM3 phase variations has been achieved. Further studies show that the error in these variations, as estimated by the method, has been found to be less than 1.5 dB and 15 degrees, respectively, when the load admittance at 1701 MHz was greater than 1/50 S.

In this paper, a new operational amplifier is presented that improves the specifications such as dc gain, common mode rejection ratio. To obtain these improvements, we have used the two important concepts of feedback and bulk amplification.

This paper describes a high power GaN-FET amplifier which is developed for wideband code division multiple access (W-CDMA) base stations. We design a bias network which is symmetrically arranged to the RF line (two way bias network) in order to reduce impedance at a baseband frequency of the multi-carrier W-CDMA signal. As a result, the amplifier with the two way bias network successfully suppressed memory effects. Therefore, the application of a DPD technique to the GaN-FET amplifier with the two way bias network demonstrates almost 20 dB linearity improvement in IM3 and considerable improvement in higher order IMD, resulting in low IMD of less than -50 dBc at the highest ever reported W-CDMA average output power of 76 W.

In this paper, theoretical properties of deinterlacer banks are analyzed. Deinterlacer banks are novel filter banks in the sense that a progressive video sequence is separated into two progressive video sequences of a half frame rate and, furthermore, interlaced sequences are produced as intermediate data. Unlike the conventional filter banks, our deinterlacer banks are constructed in a way unique to multidimensional systems by using invertible deinterlacers, which the authors have proposed before. The system is a kind of shift-varying filter banks and it was impossible to derive the optimal bit-allocation control without any equivalent parallel filter banks. This paper derives an equivalent polyphase matrix representation of the whole system and its equivalent parallel structure, and then shows the optimal rate allocation for the deinterlacer banks. Some experimental results justify the effectiveness of the optimal rate allocation through our theoretical analysis.

Three distortion reduction filters for radio-on-fiber systems are proposed and evaluated from the standpoint of improvements in in-band third order intermodulation (IM3) components (spurious components), insertion loss, temperature stability and so on. The basic filter configuration includes optical comb filter, RF (radiowave frequency) comb filter, and RF dual band rejection filter (DBRF). Experiments are conducted at 2 GHz band for frequency separation Δf=5 MHz and 100 MHz in the temperature range of -10 to +50. These filters can reduce IM3 components even in the saturation region, unlike conventional linearizers. An optical comb filter can reduce IM3 components more than 20 dB and noise level around 10 dB if its polarization controller is properly adjusted, but its insertion loss is large and stability against vibration is very poor. The proposed RF comb filter and RF-DBRF can reduce IM3 components by more than 20 dB and noise level by more than 3 dB. Their stability against vibration and temperature change is good, and insertion losses are 1-2 dB for Δf=100 MHz.

This paper presents two hardware-friendly low-power oriented fast motion estimation (ME) algorithms and their VLSI implementations. The basic idea of the proposed partial distortion sorting (PDS) algorithm is to disable the search points which have larger partial distortions during the ME process, and only keep those search points with smaller ones. To further reduce the computation overhead, a simplified local PDS (LPDS) algorithm is also presented. Experiments show that the PDS and LPDS algorithms can provide almost the same image quality as full search only with 36.7% computation complexity. The proposed two algorithms can be integrated into different FSBMA architectures to save power consumption. In this paper, the 1-D inter ME architecture [12] is used as an detailed example. Under the worst working conditions (1.62 V, 125) and 166 MHz clock frequency, the PDS algorithm can reduce 33.3% power consumption with 4.05 K gates extra hardware cost, and the LPDS can reduce 37.8% power consumption with 1.73 K gates overhead.

We derived explicit formulas for evaluating the error vector magnitude (EVM) from the amplitude distortion (AM-AM) and phase distortion (AM-PM) of power amplifiers (PAs) in orthogonal frequency-division multiplexing (OFDM) systems, such as the IEEE 802.11a/g wireless local area networks (WLANs) standards. We demonstrated that the developed formulas allowed EVM simulation of a memoryless PA using only a single-tone response (i.e. without OFDM modulation and demodulation), thus enabling us to easily simulate the EVM using a harmonic-balance (HB) simulator. This HB simulation technique reduced the processing time required to simulate the EVM of a PA for the IEEE 802.11a standard by a factor of ten compared to a system-level (SL) simulation. We also demonstrated that the measured EVM of a PA module for the IEEE 802.11g could accurately be predicted by applying the measured static AM-AM and AM-PM characteristics to the derived formulas.