A fractal image coding scheme using classified range regions is proposed. Two classes of range regions, shade and nonshade, are defined here, A shade range region is encoded by the average gray level, while a nonshade range region is encoded by IFS parameters. To obtain classified range regions, the two-stage block merging scheme is proposed. Each range region is produced by merging primitive square blocks. Shade range regions are obtained at the first stage, and from the rest of primitive blocks nonshade range regions are obtained at the second stage. Furthermore, for increasing the variety of region shape, the 8-directional block merging scheme is defined by extension of the 4-directional scheme. Also, two similar schemes for encoding region shapes, each corresponding to the 4-directional block merging scheme and the 8-directional block merging scheme, are proposed. From the results of simulation by using a test image, it was demonstrated that the variety of region shape allows large shade range regions to be extracted efficiently, and these large shade range regions are more effective in reduction of total amount of codebits with less increase of degradation of reconstructed image quality than large nonshade range regions. The 8-directional merging and coding scheme and the 4-directional scheme reveal almost the same coding performance, which is improved than that of the quad-tree partitioning scheme. Also, these two schemes achieve almost the same reconstructed image quality.

A novel image improving algorithm for compressed image sequence by merging a reference image is presented. A high quality still image of the same scene is used as a reference image. The degraded images are improved by merging reference image with them. Merging amount is controlled by the resemblance between the reference image and compressed image after applying motion compensation. Experiments conducted on sequences of JPEG images are given. This technique does not need a prior knowledge of compression technique so it can be applied to other techniques as well.

A Pragmatic Trellis Coded MPSK on a Rayleigh fading channel is analyzed. This scheme allows bandwidth expansion ratio to be varied aiming at an optimization between complexity of the system design and improvement of coding gain. In order to vary the bandwidth expansion ratio, a punctured convolutional code is used. The performance of the proposed TC-2^mPSK on a Rayleigh fading channel is theoretically analyzed. In the test examples, the BER performances of TC-QPSK and TC-8PSK are evaluated by theoretical analyses and computer simulations at the encoder parameters of K3 and r3/4. The results show that the proposed scheme can attain better performance not only over the uncoded scheme but over the conventional Pragmatic TCM.

For transmission of video signals, it is important that the system allows a certain degree of flexibility in bit rate as well as quality, depending upon the requirements of media and channel conditions. In this paper, we discuss the hierarchical transmission of Huffman code using multi-code/multi-rate DS/SS system to realize flexible transmission. We first discuss and show that the structure of Huffman code tree directly expresses hierarchical structure, and that parallel transmission of Huffman code can achieve hierarchical transmission. By assigning different transmission data rate to the bits in each stratum, it is possible to transmit different amount of information from each stratum. Further, we show the quality of each of the stratum can easily controlled by an appropriate power distribution to each parallel transmission branch.

This paper proposes a new wireless multimedia communication system based on hierarchical modulation, which gives unequal transmission reliability corresponding to the sensitivity to the transmission errors. In order to achieve high quality multimedia communication in a band-limited and time-variant fading channel, the proposed scheme changes the modulation scheme according to the contents of information. Numerical analysis shows that the proposed system is an effective high-quality and high-speed multimedia transmission technique in fading channel.

This paper proposes the radio ATM entrance network with radio links connecting between access points and wired backbone ATM networks for wireless ATM access. In order to reduce the interference power among the radio entrance links, the Power and Modulation Level Controlled Radio method is newly proposed, the method controls not only modulation level but also the transmission power according to the ATM cell traffic intensity. Theoretical analysis shows that the proposed method can increase the carrier to noise plus interference power ratio and can reduce the average cell loss rate compared with the conventional Modulation Level Controlled Radio method in case that there is the interference power among the radio ATM entrance links.

Conventional approach for frequency estimation usually assume a single tone without data modulation. In many applications such an assumption, realized by using either a separate pilot beacon or synchronization preamble is not feasible. This paper deals with frequency estimation of phase-modulated carriers in the absence of timing information and known data pattern. We introduce new frequency estimators that are based on the generalized maximum likelihood principle. The communication channels under consideration include both additive white Gaussian noise (AWGN) channels and correlated Rician fading channels. For the latter class, we distinguish between the case when the fading (amplitude) process is tracked and that when it is not tracked.

This paper proposes a novel signal transmission scheme for helicopter satellite communications. The proposed scheme is based on time diversity, and combined with a novel algorithm to suppress an influence of carrier phase slip. In the proposed scheme, carrier phase slip is detected in cross correlation processing of the received signal, and is effectively suppressed. The proposed scheme thus makes it possible to employ coherent phase shift keying modulation to achieve bit error rate performance superior to that of differential phase shift keying modulation even in the low carrier-to-noise power ratio environment.

A novel cumulant based MUSIC like DOA estimation algorithm for multicarrier modulation has been proposed in this paper. While the conventional MUSIC algorithm is not applicable to a correlation matrix calculated from received signals transmitted over the different carriers, the proposed algorithm can estimate the DOA of the signals with multicarrier modulation. The proposed algorithm does not require the sensor array responses for the frequency range of the interest and the initial phases of the carriers. With the proposed algorithm the number of signals whose DOA are estimated can be increased and the accuracy of the DOA estimation can be improved by employing larger number of carriers.

In this paper, we suggest the interference cancellation (IC) technique suitable for turbo coded code division multiple access (CDMA) systems, that merges IC processes into turbo decoding processes to improve system performance and reduce system complexity. To ensure the reliability of the temporary decision bits for cancellation, we use cyclic redundancy code (CRC) check as a measure. Prior to design turbo coded CDMA system, we first derive the optimized polynomials of low-rate turbo codes appropriate to CDMA systems. According to the simulation results with setting the processing gain (PG) to 120, the turbo coded CDMA system with the proposed IC technique can accommodate 60 users over additive white Gaussian noise (AWGN) channel when signal to noise ratio (SNR) is about 2. 5 dB and required frame error ratio (FER) is 10^-2. Compared this result with the performance of single user's system, it requires only additional 1 dB SNR.

A computational cost reduction scheme for a post-distortion type nonlinear distortion compensator of OFDM signals is proposed, and compared with the conventional sub-optimum detection scheme. The proposed scheme utilizes the principle that a complex OFDM signal can be demodulated with not only both I-phase (real part) and Q-phase (imaginary part) components, but also either of them. Usually each phase of an OFDM signal exhibits different signal envelope and they are distorted differently by the nonlinearity of a power amplifier. Consequently, three output sequence patterns can be obtained at the receiver. By comparing these outputs, we can know the erroneous positions of these sequences to some extent. By the aid of this comparison, we need to evaluate only a limited number of replicas for the compensation process of the post-distortion type nonlinear distortion compensator, which results in the computational cost reduction. We have proposed four new compensation schemes based on this idea and derived their performance in terms of the bit error rate and the average number of calculations.

In recent years, several global network systems using non-stationary satellites have been proposed. Some of them are announced to start services within years. We also have several experimental systems with stratospheric aircrafts. In the future, the radio communication system using stratospheric aircrafts will be one of the promising media for personal communications. The question of how to establish the optimal communication under such circumstance seems to be still open. In this paper, performance evaluations of wireless communication systems using LEO satellites and stratospheric aircrafts are proposed. We will show some proper communication parameters to improve competence of mobile communication in the such systems as well.

In this paper, a transceiver VLSI architecture is proposed for high speed digital CATV modems, which can perform both the QAM and the VSB transmissions. The proposed architecture of all-digital dual-mode QAM/VSB receiver consists of digital AGC, digital demodulator, fractionally spaced blind equalizer and DFE, digital carrier recovery, and symbol timing recovery. Finite word-length simulation results show that the proposed architecture can achieve an SNR 29 dB for both the 64-QAM mode and 8-VSB mode when the 10 bit ADC input signal SNR is 36 dB, and there are 6 kHz of carrier frequency offset, 110 ppm of symbol rate offset, and -82 dBc carrier phase-jitter at 10 kHz away from the nominal carrier frequency.

This paper experimentally confirms the throughput characteristics of TCP and Network Direct Memory Access (Network DMA), our proposed data transfer protocol, over a large delay-band-width link. The experiments clarify that the various problems of TCP over large delay-bandwidth links include limited window size, data retransmission mechanism, and protocol processing overhead. The test results suggest that we must improve not only protocol itself but also the protocol processing architecture to realize Gbit/s class throughput over such links. To avoid these problems, Network DMA realizes high speed memory copy across a network by labeling each packet with its memory address without host CPU intervention; protocol processing is done by firmware on the network interface card. Moreover, it realizes selective retransmission by using the memory addresses. Test results show that Network DMA achieves the sustained throughput of 535Mbit/s over a 10,000km 622Mbit/s ATM link and over 400Mbit/s effective throughput even when the cell loss ratio is 10^-4.

The explosive increase of traffic in computer communications is a clear sign that we have entered the multimedia information age. To cope with this ever increasing need, economical optical access networks that support burst traffic such as in the Internet are expected to be developed. The ATM-PON is considered to be a promising candidate for such a network, and vigorous efforts in this direction are being promoted worldwide. This paper focuses on accommodating burst traffic in the ATM-PON. In order to do this, a mechanism to transport bandwidth requests from the ONU to the OLT and an algorithm to support dynamic bandwidth allocations based on ONU requests are needed. We have performed a comparative study on bandwidth request methods and bandwidth allocation algorithms, including bandwidth request dependence on time interval and correlation and/or impact between system design parameters. The results of computer simulations are useful in determining how to accommodate burst traffic efficiently in the ATM-PON.

Major problems of traffic control in ATM networks include how to decide whether a network accepts a new call or not in real time and how to select the best set of Dual Leaky Bucket (DLB) parameter values. To solve these problems, it is necessary to determine the amount of network bandwidth required by the call. In this paper, we present an analysis based on bounding technique to derive an upper bound on bandwidth requirement when the call is characterized by a set of DLB parameters. Consequently, a new definition of the upper bound on bandwidth requirement and simple formulae used for computing the upper bound have been obtained. To clarify the advantages of the derived upper bound, we demonstrate its two applications, one to select the best set of DLB parameter values from candidates for minimizing the amount of bandwidth to be allocated to the call and the other to establish a Connection Admission Control (CAC) scheme. The upper bound-based CAC scheme is fast enough to process in real time due to its simplicity and provides a significant improvement of network utilization compared to the peak rate-based CAC scheme.

In this paper we present two traffic control approaches, a circuit emulation traffic control (CETC) and an adaptive priority traffic control (APTC) for supporting voice services in ATM networks. Most voice services can be handled as CBR traffic, this causes a lot of wasted bandwidth. Sending voice through VBR (variable bit rate) may be a better alternative, because it allows the network to allocate voice bandwidth on demand. In CETC, the service discipline guarantees the quality of service (QOS) for voice circuits. Through mathematical analysis, we show that CETC features an adequate performance in delay-jitter. Moreover, it is feasible in implementation. We also present an APTC approach which uses a dynamic buffer allocation scheme to adjust the buffer size based on the real traffic need, as well as employs an adaptive priority queuing technique to handle various delay requirements for VBR voice traffic. It provides an adequate QOS for voice circuits in addition to improving the multiplexing gain. Simulation results show that voice traffic get satisfied delay performance using our approaches. It may fulfill the emerging needs of voice service over ATM networks.

This paper explores virtual destination (VD)/virtual source (VS)-based available bit rate (ABR) flow control performance, targeting wireless asynchronous transfer mode (WATM) application that can incur long link-delays because of employing radio-medium sharing and/or radio-specific data link control schemes. As this paper reveals, the conventional VD/VS scheme has difficulty in sustaining satisfactory ABR performance, when it is applied to long-delay-causing WATM; it suffers from significant increase in the necessary buffer capacity. To ensure the ABR performance in WATM, this paper proposes a new VD/VS coupling scheme using a feed-forward congestion indication. The proposed scheme controls the allowed cell rate of a source end system in a feed-forward manner by predicting the queue length at the time the WATM-associated-round-trip ahead. Simulation results show that the proposed scheme exhibits excellent ABR performance with a long delay of the divided loop on the radio-link side. It is also verified that the proposed scheme is rather robust against uncertainty and/or time-variation regarding the predetermined radio link delay.

In this work, we propose an End-to-End Rate Control Approach (EERCA) for congestion avoidance in Available Bit Rate (ABR) service on Asynchronous Transmit Mode (ATM) networks. In our approach, the network estimates the number of cells stored in the switch for each VC. The source generates a specific traffic pattern, then a proper explicit rate can be derived based on the received traffic pattern at the destination. This approach is designed to reduce the rate calculation effort in the switch as well as to avoid the complexity in setting the monitoring-interval. EERCA features higher efficiency, higher utilization, more stable queue occupancy, shorter transient response time, and better fairness compared with existed schemes.

The project of interconnecting CATV in Hyogo Prefecture, Japan has started since March, 1998. In this project, there are three CATV companies in Hanshin area; Kobe, Nishinomiya and Amagasaki. An ATM switch is equipped in each company and these CATVs are connected serially in the above order. Each company provides the video service to the rest of companies using the MPEG2 over ATM. Each MPEG2 stream is sent to the other two CATV companies according to the function of multicast implemented in ATM switch. In the coverage of each CATV, subscribers utilize Internet connection using cable modems as well as standard CATV broadcasting service. In this paper, we present the outline of the research project in Hyogo Prefecture, Japan, and examine the jitter processes of MPEG streams of the testbed network by the simulation. In our testbed network, cells with two types of requirement for QoS are multiplexed; cells for MPEG2 which require the real-time transmission and those for Internet packets which are much more sensitive for the cell loss ratio. We investigate the jitter processes under some scenarios and show how the jitter process is affected by the Internet traffic and the other cell streams of MPEG2. Furthermore, we study the effect of the number of ATM switches on the jitter process when more CATV networks are added serially.

This paper studies the application of a media synchronization mechanism to the interleaved transmission of video and audio specified by the H.223 Annex in PHS. The media synchronization problem due to network delay jitters in the interleaved transmission has not been discussed in either the Annex or any related standards. The slide control scheme, which has been proposed by the authors, is applied to live media. We also propose a QOS control scheme to control both quality of the media synchronization and that of the transmission delay. Through simulation we confirm the effectiveness of the slide control scheme and the QOS control scheme in the interleaved transmission.

In this paper, we propose and evaluate the performance of Wireless ATM MAC layer protocol to support efficiently various ATM traffics, such as CBR, VBR, ABR and UBR, in wireless ATM network environments for reverse and forward link. The proposed MAC protocol could extend efficiently the service discipline of ATM traffics from wired network to wireless ATM network environments. Thus, available bandwidth, which is remained except the bandwidth for CBR and VBR traffics, could be effectively allocated to ABR and UBR traffics. Especially, in view of reverse link, two-phase scheduling algorithm supports successfully variable characteristics of VBR traffic. And, in view of forward link, 'Wireless Dynamic Weighted Earliest Deadline First' scheduling algorithm minimizes the mean cell delay and required buffer size. Simulation results show that proposed method provides effective performance in wireless ATM environments.

This paper proposes an intelligent and autonomous radio resource management scheme for a multi-layered cellular system with different assigned bandwidths to achieve flexible and high capacity wireless communication systems under any traffic conditions, especially under nonuniform traffic conditions. In the proposed system, terminals with lower mobility are connected to the wideband microcell systems to achieve higher system capacity, and terminals with higher mobility are connected to the narrowband macrocell systems to reduce intercell hand-off frequency. To flexibly cope with variations of traffic conditions, radio spectrum is adaptively and autonomously shared by both systems, and its control is conducted by each microcell base station. Moreover, at the existence of nonuniform traffic conditions , the proposed system introduces downlink power control for the microcells and graceful degradation thereby achieving high system capacity even under such extraordinary traffic situations . Computer simulation confirms that the proposed scheme can achieve lower blocking probability than the centralized scheme especially under nonuniform traffic conditions.

An extended desktop multimedia conference system named FREDERIC (File Retrieval Engineering on Distributed EnviRonment and Interactive Communication system) has been developed for international cooperative work by sharing CAD and image data among multi-point users. This paper describes the basic network design concept of utilizing the Internet as a best-effort service and ISDN as a high-speed guaranteed service. Service system requirements and designs were developed to access common databases and collaborative work of multimedia information those are shared by customers with desktop computers and to allow remote offices to use a plant walkthrough system. The performance of the prototype system especially focused on the file transmission time which is the key factor in developing and constructing the system. By applying the image compression technology of multi-tone entropy coding, it is shown that the short time CAD data transfer to meet the requirements can be achieved.

With the introduction of ATM technology, service providers around the world have actively engaged in offering high bandwidth services. Currently, services, such as T1/E1, T3/E3 circuit emulation, are made available to large-volume account users. However, more advanced services, such as multimedia applications, have demanded not just high bandwidth but also flexible rate adaptation with quality-of-service (QoS) guarantee. To support the above service requirements, sophisticated network planning and engineering procedures should be taken. In the past few years, we have conducted various researches on developing the engineering strategies for resource control and management to support multi-rate service offering. We have also looked into the design details of connection control and management for achieving the QoS requirement. We considered the service quality of the underlying transport in regard with the QoS management. In this paper, we will outline those results and give an overview description about the proposed framework.

Network traffic contains many symptoms of various network faults. Symptoms of faults aggregate and are manifested in the aggregate traffic characteristics generally observed by a traffic monitor. It is very difficult for a manager or an NMS (Network Management Station) to isolate the symptoms manifested in the aggregate traffic characteristics. Especially, transit networks, like a backbone network, deal with many types of traffic. So, symptom isolation must be efficient. In this paper, we propose a powerful algorithm for symptom isolation. This algorithm is based on the popular SNMP-based RMON technology. Using dynamically constructed aggregate, fresh symptoms can be isolated efficiently. We apply the algorithm to two operational transit networks which connects some LANs and WANs, and evaluate it using trace data collected from these networks. The results show a significant improvement in the fault management capability and accuracy. Furthermore, the characteristics of fault symptoms and the various factors for effective system configuration are discussed.

Call tracking data contains a calling address, called address, service type, and other useful attributes to predict a customer's calling activity. Call tracking data is becoming a target of data mining for telecommunication carriers. Conventional data-mining programs control the number of association rules found with two types of thresholds (minimum confidence and minimum support), however, often they generate too many association rules because of the wide variety of patterns found in call tracking data. This paper proposes a new method to reduce the number of generated rules. The method proposed tests each generated rule based on Akaike Information Criteria (AIC) without using conventional thresholds. Experiments with artificial call tracking data show the high performance of the proposed method.

A method is proposed for deriving a traffic characteristics model that can be used to forecast the traffic volume for intelligent telecommunication services. A sort of regression analysis with dummy variables is used to represent the service quantitatively and to construct the traffic characteristics model. Recursive least squares estimation, which is a special case of the Kalman filter, is applied to the traffic characteristics model to forecast the traffic volume. In the proposed modeling and forecasting, qualitative factors representing a certain service attribute are selected and using an information criterion, the model with the best fit is identified as the most suitable forecasting model. Numerical results using practical observation data showed that the proposed method produces an accurate forecast and is thus effective for practical use.

In this paper, we describe a routing method for path on SDH Network with digital cross-connect control, which is implemented in an automated path provisioning function. Excessive concentration of assigned time slots at particular links results in longer restoration time, which is needed to switch or reroute paths on failure link. We propose an optimization method to provision the shortest route considering deconcentration of time slots assigned on each link. After defining LP-based formulation for path routing, we carried out computer simulation study for restoration performance on sample networks, assuming each restoration process for paths on failure link is executed one after another. Mean restoration time by our proposed method has reduced to a great extent compared to a basic routing method. It has been proven that the proposed method can realize effective use of resources and faster restoration time, and can be utilized in commercial systems.

This paper proposes a novel ultra high-speed file server based on a personal computer (PC) to provide the instantaneous delivery of huge files, like movie files, graphic images and computer programs, over high-speed networks. In order to improve the sustained sequential read speed from arrays of hard drives to host memory in the server, two key techniques are proposed: "multi-stage striping (MSS)" and the "sequential file system (SFS)." An experimental file server based on a general-purpose PC is constructed and its performance is measured. The results show that the server offers ultra high read speeds, up to 105Mbytes/s, with just 8 hard drives.

Voice activity detection (VAD) is to determine whether a short time speech frame is voice or silence. VAD is useful in reducing the mean speech coding rate by suppressing transmission during silence periods, and is effective in transmitting speech and other data simultaneously. This letter describes a VAD system that uses a neural network. The neural network gets several parameters by analyzing slices of the speech wave form, and outputs only one scalar value related to voice activity. This output is compared to a threshold to determine whether the slice is voice or silence. The mean code transfer rate can be reduced to less than 50% by using the proposed VAD system.

In this paper, we consider the applicability of turbo code for future third generation (3G) mobile telecommunication systems. Futhermore, we propose a simple method of estimating the channel variance which is necessary for the MAP (Maximum A Posteriori) decoding algorithm. We compare the performance of turbo code with a known channel variance, conventional variance estimate and variance estimated by our proposed technique. We show that our variance estimation scheme is adequate for 3G WB-CDMA mobile systems without degradation of turbo code performance.

The productivity of developing software for switching systems and the effects of using advanced software development methods were evaluated and analyzed. Productivity was found to be improved by using automatic code generation, simulator debugging, a hierarchical object-oriented software structure, and software-development-support tools. The evaluation showed that the total productivity was improved by about 20%, compared with a case where these efforts were not introduced. It also showed each effect of these methods and tools by evaluating their manpower saving ratios. These results are expected to benefit the development of various types of communication-switching and multimedia service systems. Also, our development-support tools and methods are expected to be the basis for attaining higher software development productivity.

A zerotree image-coding scheme is introduced that effectively exploits the inter-scale self-similarities found in the octave decomposition by a wavelet transform. A zerotree is useful for efficiently coding wavelet coefficients; its efficiency was proved by Shapiro's EZW. In the EZW coder, wavelet coefficients are symbolized, then entropy-coded for further compression. In this paper, we analyze the symbols produced by the EZW coder and discuss the entropy for a symbol. We modify the procedure used for symbol-stream generation to produce lower entropy. First, we modify the fixed relation between a parent and children used in the EZW coder to raise the probability that a significant parent has significant children. The modified relation is flexibly modified again based on the observation that a significant coefficient is more likely to have significant coefficients in its neighborhood. The three relations are compared in terms of the number of symbols they produce.

A unified source coding method is highly desired for many systems that deal with images diversifying from 1 bit/pel bi-level documents to SHD (Super High Definition) images of 12 bit/pel for each color component, and progressive coding that allows images to be reconstructed with increasing pixel accuracy or spatial resolution is essential for many applications including World Wide Web, medical images archive, digital library, pre-press and quick look applications. In this paper, we propose a unified continuous-tone and bi-level image coding method with pyramidal and progressive transmission feature. Hierarchical structure is constructed by interlacing subsampling, and each hierarchy is encoded by DPCM combined with reduced Markov model. Simulation results show that the proposed method is a little inferior than JBIG for bi-level image coding but can achieve better lossless compression ratio for gray-level image coding than CREW, in which wavelet transform is exploited to construct hierarchical structure.

A novel zero-voltage-switched half-bridge converter is proposed. This converter achieves the zero-voltage switching while maintaining a constant frequency PWM control. Then the power conversion of high efficiency and low noise is realized at a higher switching frequency. In the experiment, a high efficiency of 83% is achieved for a low output voltage of 3.3 V, an output current of 30 A, and an input-voltage range of 200 to 400 V at the switching frequency of 400 kHz.

A fuel cell energy system is under development for supply of generated electrical energy to telecommunications equipment. It is a cogeneration system; the heat energy recovered is used to cool the telecommunications equipment. For this system, a method is described for controlling a new DC interconnection converter. Its DC interconnection characteristics are also discussed. The new converter controls its input current to the fuel cell rated current at maximum and can operate stably even when the fuel cell voltage decreases. This allows good DC interconnection characteristics to be obtained in both the steady state and the transient state.

In this paper, millimeter-wave radar cross section (RCS) characteristics for rough surface is investigated by means of an approximation method of the magnetic field integral equation and the feasibility of road condition sensing is discussed. The RCS measurement at 94 GHz is carried out in order to verify the numerical result, thereby the numerical results are in good agreement with the measured RCS. The dependence of RCS on the radar incidence angle and surface roughness is investigated where the cross-polarized RCS characteristic is also considered.

A decision-directed carrier phase recovery scheme for high-speed satellite communications is proposed. Since the estimation is performed in complex domain from the baseband signal, the scheme has fast acquisition performance, unlike the conventional PLL. This merit makes it applicable for various wireless systems such as wireless local area networks (LANs), wireless asynchronous transfer modes (ATMs) and local multipoint distribution systems (LMDSs) that need high-speed burst signal communications. Also, this scheme can be implemented easily because low pass filters (LPFs) are utilized in filtering the estimates in order to suppress the noise within the carrier recovery loop. Moreover it does not require any divider or voltage-controlled oscillator (VCO). The performance is analyzed through analytical methods and simulation.