Function Of Bit Error Rate Research Articles

Research on user pairing algorithm for LTE femtocell uplink

In this article, a new simulated annealing based bit error rate pairing scheduling (SA-BPS) algorithm for LTE femtocell uplink virtual multiple input multiple output (V-MIMO) systems is proposed, which uses Round Robin (RR) criterion to decide the first user, and simulated annealing (SA) algorithm with objective function of bit error rate (BER) is suggested to decide the pairing users. The SA-BPS is considered not only to be used for inner layer user pairing within femtocell user equipments (HUEs) to increase spectrum efficiency, but also to be used for cross layer user pairing between HUEs and macrocell user equipments (MUEs) to cancel the cross layer interference from MUEs to HUEs. As a growth control coefficient is used in SA-BPS to control the growth of average BER of users in the pairing group, rapid user BER deterioration can be prevented by adopting a proper coefficient. Simulation results show that, the SA-BPS outperforms RPS, DPS, and the conventional SA algorithm in terms of BER performance, whether used for inner layer user pairing or cross layer user pairing. Also, the mode switching signal to interference ratio (SIR) threshold between inner and cross layer user pairing are obtained by simulation in three cross layer interference power distribution scenarios.

A new measure of index assignment in dynamic channels

In dynamic noisy communication channels, a channel bit error rate (BER) varies with time. When channel distortion measure D c is a function of bit error rate, an optimal index assignment to minimize D c is not as uniformly optimal as BER varies. The paper presents a new distortion measure called expected channel distortion (ECD). It is independent of the bit error rate by two proposed beta-type distributions in a multiple-error pattern. Additionally, any index assignment algorithm with ECD measure is mismatch-free. In other words, index assignment algorithms can be against channel mismatch even though BER varies with time in dynamic channels. The voice digitization in North American Telephone Systems (CCITT) is adopted to demonstrate the usefulness of the proposed measure.

Spread spectrum multiple-access with DPSK modulation and diversity for image transmission over indoor radio multipath fading channels

As demand for networked multimedia applications is increasing rapidly, it is important to provide the ubiquitous accessibility for these services in wireless communication environment. Such access allows users to share novel multimedia applications without any geographical restrictions. We investigate the application of the well-known antimultipath spread spectrum code-division multiple access (SS-CDMA) techniques to image transmission related to the development of next-generation multimedia wireless local area networks within a building. SS-CDMA is particularly well suited to a subband coding scheme that divides the image information into multiple parallel data streams using an analysis filter bank, each of which is multiplied by its unique spreading code. All the product signals are then transmitted at the same time in the same radio channel, even though the total bandwidth of all the signals may exceed the channel bandwidth. Each received signal is independently recovered at the decoder by multiplying its spreading code and all the recovered subbands are then reassembled by a synthesis filter bank into a close reproduction to the original image. A forward error correction (FEC) scheme based on convolutional codes with interleaving is proposed to minimize the effect of bursty channel errors on the picture quality degradation. Better image quality can be achieved by using the predetection diversity combining in order to combat the indoor multipath distortion. In this paper, the image quality of subband image transmission via SS-CDMA indoor fading channels with differential phase shift keying (DPSK) modulation is evaluated and examined. Simulation results show that the image peak signal-to-noise (PSNR) ratio is inversely related to a logarithmic function of bit error rate when each subband employs equal protection channel coding.

The Analysis of Some Selective-Repeat ARQ Schemes with Finite Receiver Buffer

In high bit rate data transmission systems with ARQ error control, the throughput efficiency is a function of bit error rate, block or packet size, and the effect of significant round trip delays such as may be experienced in satellite communication systems. The selective-repeat ARQ scheme is capable of providing superior throughput performance independent of round trip delay, but requires excessively large receiver buffers; as a result the inferior GoBack <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">N</tex> procedure is commonly adopted. This paper analyzes a class of mixed-mode ARQ protocol models which incorporate a selectiverepeat mode with finite receiver buffer. The protocol models are shown to be amenable to exact throughput analysis, but do assume that the round trip delay is constant and known, blocks are of constant length, and the ACK/NAK signals are returned error free. These assumptions might create difficulties for practical implementation. However, the analytical model results highlight those aspects of ARQ protocols which affect throughput performance as round trip delays increase. The results show that it is desirable for best throughput performance in practical systems that at least the first retransmission of a block following an error should be in the selective-repeat mode to obtain superior performance over GoBack <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">N</tex> schemes. Furthermore, alternative secondary retransmission modes are considered which ensure that reliable transmission can be achieved without receiver buffer overflow, even if the selective-repeat mode retransmissions fail. It is shown that the choice of secondary mode does not have a significant effect on the throughput efficiency but has a bearing on complexity.

Hardware for Detection and Partial Correction of PCM Transmission Errors

In pulse code modulation (PCM) of speech, transmission errors cause changes in the statistical properties of the received samples. Recently published work describes computer simulations of a difference detection and correction (DDC) system which detect errors on the basis of these changes. The system examines the differences between adjacent received samples. If a particular difference exceeds a threshold, which is dependent on the rms value of a block of 64 differences, the quantized sample responsible for this large difference is deemed to be erroneous. A nonlinear filter of the median type is then introduced as a corrector. We now present a comprehensive description of the hardware realization of this system, and present measured results of signal-to-noise ratio (snr) as a function of bit error rate, input power and the error criterion for narrow-band white noise input signals. An improvement of 6 dB in snr is achieved for error rates between 0.2 and 2.0%. The snr results are consistent with subjective impressions of the quality of speech processed by the DDC system.