Source Bits Research Articles

Video Streaming Over Wireless Packet Networks: An Occupancy-Based Rate Adaptation Perspective

The perceived video quality in a wireless streaming application strongly depends on the channel's dynamics and the fluctuations of the source bit rate. In this paper, we introduce two channel-adaptive rate control schemes for slowly and fast varying channels, respectively. Both schemes account for the playback buffer occupancy in the joint optimization of the source rate and channel-code forward error correction parameters. For the first scheme, we assume that the channel state does not change during the transmission of a video frame. We optimize the channel-code parameters and maximize the per-frame source rate subject to satisfying a constraint on the probability of delivering the next video frame within a buffer-occupancy-dependent critical time (Tc). For the second scheme, we allow the channel state to change within the frame delivery period, and we compute the optimal system parameters and maximize the source rate while satisfying a constraint on the mean frame delivery time. Our schemes aim at maintaining the occupancy of the playback buffer around a predefined threshold value, hence ensuring continuous video playback. Simulation and numerical investigations are carried out to study the interactions among various key parameters and verify the adequacy of the analysis.

Cooperative Source and Channel Coding for Wireless Multimedia Communications

Past work on cooperative communications has indicated substantial improvements in channel reliability through cooperative transmission strategies. To exploit cooperation benefits for multimedia transmission over slow fading channels, we propose to jointly allocate bits among source coding, channel coding and cooperation to minimize the expected distortion of the reconstructed signal at the receiver. Recognizing that, not all source bits are equally important in terms of the end-to-end distortion, we further propose to protect the more important bits through user cooperation. We compare four modes of transmission that differ in their compression and error protection strategies (single layer or multiple layer source coding with unequal error protection, with versus without cooperation). Our study includes an i.i.d. Gaussian source as well as a video source employing an H.263+ codec. We present an information theoretic analysis for the Gaussian source to investigate the effects of the modulation scheme, bandwidth ratio (number of channel uses per source sample), and average link signal-to-noise ratios on the end-to-end distortion of the four modes studied. The information theoretic observations are validated using practical channel coding simulations. Our study for video considers error propagation in decoded video due to temporal prediction and jointly optimizes a source coding parameter that controls error propagation, in addition to bits for source coding, channel coding and cooperation. The results show that cooperation can significantly reduce the expected end-to-end distortion for both types of source and that layered cooperation provides further improvements and extends the benefits to a wider range of channel qualities.

Joint Source-Channel Rate Allocation in Parallel Channels

A fast rate-optimal rate allocation algorithm is proposed for parallel transmission of scalable images in multichannel systems. Scalable images are transmitted via fixed-length packets. The proposed algorithm selects a subchannel, as well as a channel code rate for each packet, based on the signal-to-noise ratios (SNRs) of the subchannels. The resulting scheme provides unequal error protection of source bits and significant gains are obtained over equal error protection schemes. An application of the proposed algorithm to JPEG2000 transmission shows the advantages of exploiting differences in SNRs between subchannels. Multiplexing of multiple sources is also considered, and additional gains are achieved by exploiting information diversity among the sources.

Rate Control for H.264 Video With Enhanced Rate and Distortion Models

A new rate control scheme for H.264 video encoding with enhanced rate and distortion models is proposed in this work. Compared with existing H.264 rate control schemes, our scheme has offered several new features. First, the inter-dependency between rate-distortion optimization (RDO) and rate control in H.264 is resolved via quantization parameter estimation and update. Second, since the bits of the header information may occupy a larger portion of the total bit budget, which is especially true when being coded at low bit rates, a rate model for the header information is developed to estimate header bits more accurately. The number of header bits is modeled as a function of the number of nonzero motion vector (MV) elements and the number of MVs. Third, a new source rate model and a distortion model are proposed. For this purpose, coded 4 times 4 blocks are identified and the number of source bits and distortion are modeled as functions of the quantization stepsize and the complexity of coded 4 times 4 blocks. Finally, a R-D optimized bit allocation scheme among macroblocks (MBs) is proposed to improve picture quality. Built upon the above ideas, a rate control algorithm is developed for the H.264 baseline-profile encoder under the constant bit rate constraint. It is shown by experimental results that the new algorithm can control bit rates accurately with the R-D performance significantly better than that of the rate control algorithm implemented in the H.264 software encoder JM8.1a

Cycle-Based Rate Control for One-Way and Interactive Video Communications Over Wireless Channels

We propose a joint source-rate/channel-code control scheme for streaming video over a wireless channel. The scheme is designed to maximize the achievable source rate while guaranteeing an upper bound on the probability of starvation at the playback buffer. It can be applied to both one-way and interactive video communications. Rate control is performed adaptively on a per-cycle basis, where a cycle consists of a channel period and the ensuing period. This cycle-based approach has two advantages. First, it reduces the fluctuations in the source bit rate, ensuring smooth variations in video quality. Second, it makes it possible to derive simple expressions for the starvation probability at the playback buffer, which we use to determine the optimal source rate and channel code for the good and bad periods of the subsequent cycle

Encoding and Decoding Binary Product Lattices

A binary product lattice is generated from two binary component lattices of lower dimensions by employing the Kronecker product. This work focuses on codes carved from binary product lattices. Defined as such, an intriguing problem is that of effectively mapping independent data sequences onto a selected subset of lattice points. A novel approach is disclosed yielding an explicit connection between source bits and lattice points. Decoding methods typically used for binary product codes do not apply for product lattices. Several alternative decoding approaches are discussed. In particular, a provably bounded-distance decoder is presented. It relies on the fact that a product lattice code point may be regarded as a two-dimensional array whose rows and columns are points in the component lattices. The obtained results are compared with classical lattices known in the art

A joint source and channel coding algorithm for error-resilient SPIHT-coded video bitstreams

We propose an analytical rate-distortion optimized joint source and channel coding algorithm for error-resilient scalable encoded video for lossy transmission. A video is encoded into multiple independent substreams to avoid error propagation and is assigned forward error correction (FEC) codes and source bits using Lagrange optimization. Our method separates video coding and packetization into different tiers which can be easily incorporated into any coding structure that generates a set of independent compressed bit-streams. To demonstrate the performance, we use the 2-state Markov model to describe the burst loss channel and Reed-Solomon codes as forward error correction codes. Simulation results show that the proposed channel incorporated rate-distortion optimization approach have better performance.

781 Mbit/s photon-counting optical communications using a superconducting nanowire detector

We demonstrate 1550 nm photon-counting optical communications with a NbN-nanowire superconducting single-photon detector. Source data are encoded with a rate-1/2 forward-error correcting code and transmitted by use of 32-ary pulse-position modulation at 5 and 10 GHz slot rates. Error-free performance is obtained with -0.5 detected photon per source bit at a source data rate of 781 Mbits/s. To the best of our knowledge, this is the highest reported data rate for a photon-counting receiver.

Fast algorithm for distortion-based error protection of embedded image codes

We consider a joint source-channel coding system that protects an embedded bitstream using a finite family of channel codes with error detection and error correction capability. The performance of this system may be measured by the expected distortion or by the expected number of correctly decoded source bits. Whereas a rate-based optimal solution can be found in linear time, the computation of a distortion-based optimal solution is prohibitive. Under the assumption of the convexity of the operational distortion-rate function of the source coder, we give a lower bound on the expected distortion of a distortion-based optimal solution that depends only on a rate-based optimal solution. Then, we propose a local search (LS) algorithm that starts from a rate-based optimal solution and converges in linear time to a local minimum of the expected distortion. Experimental results for a binary symmetric channel show that our LS algorithm is near optimal, whereas its complexity is much lower than that of the previous best solution.

Coin flipping from a cosmic source: On error correction of truly random bits

AbstractWe study a problem related to coin flipping, coding theory, and noise sensitivity. Consider a source of truly random bits x ∈ {0, 1}n, and k parties, who have noisy version of the source bits yi ∈ {0, 1}n, when for all i and j, it holds that P[y = xj] = 1 − ϵ, independently for all i and j. That is, each party sees each bit correctly with probability 1 − ϵ, and incorrectly (flipped) with probability ϵ, independently for all bits and all parties. The parties, who cannot communicate, wish to agree beforehand on balanced functions fi: {0, 1}n → {0, 1} such that P[f1(y1) = … = fk(yk)] is maximized. In other words, each party wants to toss a fair coin so that the probability that all parties have the same coin is maximized. The function fi may be thought of as an error correcting procedure for the source x. When k = 2,3, no error correction is possible, as the optimal protocol is given by fi(yi) = y. On the other hand, for large values of k, better protocols exist. We study general properties of the optimal protocols and the asymptotic behavior of the problem with respect to k, n, and ϵ. Our analysis uses tools from probability, discrete Fourier analysis, convexity, and discrete symmetrization. © 2005 Wiley Periodicals, Inc. Random Struct. Alg., 2005

Globally Optimal Uneven Error-Protected Packetization of Scalable Code Streams

In this paper, we present a family of new algorithms for rate-fidelity optimal packetization of scalable source bit streams with uneven error protection. In the most general setting where no assumption is made on the probability function of packet loss or on the rate-fidelity function of the scalable code stream, one of our algorithms can find the globally optimal solution to the problem in O(N/sup 2/L/sup 2/) time, compared to a previously obtained O(N/sup 3/L/sup 2/) complexity, where N is the number of packets and L is the packet payload size. If the rate-fidelity function of the input is convex, the time complexity can be reduced to O(NL/sup 2/) for a class of erasure channels, including channels for which the probability function of losing n packets is monotonically decreasing in n and independent erasure channels with packet erasure rate no larger than N/2(N + 1). Furthermore, our O(NL/sup 2/) algorithm for the convex case can be modified to rind an approximation solution for the general case. All of our algorithms do away with the expediency of fractional bit allocation, a limitation of some existing algorithms.

Qos enhancement for adaptive streaming services over wcdma

This paper presents a new method to adapt the quality-of-service (QoS) in a wideband code-division multiple-access (WCDMA) system, called radio network feedback (RNF). The concept of RNF is in general applicable to all the services requiring a minimum quality (i.e., non best effort) and in this paper, the focus is on streaming. In this context, RNF makes it possible for a streaming server to adapt its source bit rate to a WCDMA radio link, whose bandwidth may vary in time due, for example, to decongestion/congestion situations over the air interface or to handover. This is very beneficial for the end user as it allows to increase the bandwidth (i.e., the quality) when possible and to decrease it (instead of just dropping the service) when needed. RNF was compared with Client-based adaptation solutions. Simulation results show that RNF is fast and accurate and performs better than Client-based adaptation. For example, rebuffering occurs only with Client-based method. Moreover, with RNF the up-switch is performed only when needed, whereas the up-switch performed by means of the Client-based method is often erroneous and may annoy the end user.

Real-time unequal error protection algorithms for progressive image transmission

We consider unequal error protection strategies for the efficient progressive transmission of embedded image codes over noisy channels. In progressive transmission, the reconstruction quality is important not only at the target transmission rate but also at the intermediate rates. An adequate error protection strategy may, thus, consist of optimizing the average performance over the set of intermediate rates. The performance can be the expected number of correctly decoded source bits or the expected distortion. For the rate-based performance, we prove some interesting properties of an optimal solution and give an optimal linear-time algorithm to compute it. For the distortion-based performance, we propose an efficient linear-time local search algorithm. For a binary symmetric channel, two state-of-the-art source coders (SPIHT and JPEG2000), we compare the progressive ability of our proposed solutions to that of the strategies that optimize the end-to-end performance of the system. Experimental results showed that the proposed solutions had a slightly worse performance at the target transmission rate and a better performance at most of the intermediate rates, especially at the lowest ones.

Fast algorithm for rate-based optimal error protection of embedded codes

Embedded image codes are very sensitive to channel noise because a single bit error can lead to an irreversible loss of synchronization between the encoder and the decoder. P.G. Sherwood and K. Zeger (see IEEE Signal Processing Lett., vol.4, p.191-8, 1997) introduced a powerful system that protects an embedded wavelet image code with a concatenation of a cyclic redundancy check coder for error detection and a rate-compatible punctured convolutional coder for error correction. For such systems, V. Chande and N. Farvardin (see IEEE J. Select. Areas Commun., vol.18, p.850-60, 2000) proposed an unequal error protection strategy that maximizes the expected number of correctly received source bits subject to a target transmission rate. Noting that an optimal strategy protects successive source blocks with the same channel code, we give an algorithm that accelerates the computation of the optimal strategy of Chande and Farvardin by finding an explicit formula for the number of occurrences of the same channel code. Experimental results with two competitive channel coders and a binary symmetric channel showed that the speed-up factor over the approach of Chande and Farvardin ranged from 2.82 to 44.76 for transmission rates between 0.25 and 2 bits per pixel.

A study of real-time packet video quality using random neural networks

An important and unsolved problem today is that of automatic quantification of the quality of video flows transmitted over packet networks. In particular, the ability to perform this task in real time (typically for streams sent themselves in real time) is especially interesting. The problem is still unsolved because there are many parameters affecting video quality, and their combined effect is not well identified and understood. Among these parameters, we have the source bit rate, the encoded frame type, the frame rate at the source, the packet loss rate in the network, etc. Only subjective evaluations give good results but, by definition, they are not automatic. We have previously explored the possibility of using artificial neural networks (NNs) to automatically quantify the quality of video flows and we showed that they can give results well correlated with human perception. In this paper, our goal is twofold. First, we report on a significant enhancement of our method by means of a new neural approach, the random NN model, and its learning algorithm, both of which offer better performances for our application. Second, we follow our approach to study and analyze the behavior of video quality for wide range variations of a set of selected parameters. This may help in developing control mechanisms in order to deliver the best possible video quality given the current network situation, and in better understanding of QoS aspects in multimedia engineering.

Frame synchronization in guided scrambling line codes

Guided scrambling (GS) ensures the existence of good line code properties in encoded bit sequences through appropriate augmentation of the source bit stream prior to self-synchronizing scrambling. At the receiver, a framing algorithm is required in order to identify the location of the augmenting bits in the unscrambled bit sequence. This paper proposes and analyzes a self-framing algorithm for achieving frame synchronization in GS line codes. This technique is based on the observation of a large number of transitions at codeword boundaries in the encoded sequence, and does not require any additional overhead. Analysis and simulation results confirm the viability of this technique.

A Multimode and Variable‐Rate Voice Communications System with Source‐Matched Error Protection for Mobile Communications

AbstractThis paper presents a performance evaluation of the multimode Fast Variable Rate CELP coder developed by CSELT S.p.A. in the UMTS Code Division Testbed (CODIT) Project [1] for applications involving channels with time‐varying error characteristics. The radio channel is statistically modelled by one Rician or Rayleigh‐distributed desired signal and several uncorrelated Rayleigh plus log‐normally shadowed interfering signals, propagating according to two different laws: a dual path loss law with a turning point (microcellular environment) and a single path loss law (macrocellular environment). For channels with these error characteristics, each mode of the Fast Variable Rate CELP coder has a particular allocation of bits between source and channel coding stages, leading to an unequal error protection code tailored to suit the different error sensitivities of the source bits. It is shown that, on the basis of the sensitivity evaluation of source bits to errors, according to two different error sensitivity measures. Forward Error Correction whose power is matched to the error sensitivity of the source bits can be used. Increased robustness of source bits to errors can thus be obtained while satisfying bandwidth requirements.

An effective model for VBR video with application to an ATM multiplexer

The effective design and engineering of communication networks require accurate mathematical models that should capture the important statistical properties of traffic sources. The ATM Forum has specified some QoS requirements for video traffic, which are essential for the effective management and operation of ATM networks. In this paper, a model for VBR video sources using a modified first-order autoregressive model is presented. Two levels of states with arbitrary average duration are used to represent the source bit-rate. The corresponding ratio of the statistical duration averages in the two levels is used as a new parameter to quantify the activity behaviour of the pertinent sources. A derivation for the resulting model statistical equations is presented and matched to some experimental data. Then, the obtained matched parameters are plugged into the simulation model. The major results are presented for various parameter values, along with an assessment of their effects on the performance of an ATM multiplexer.

On the power spectral density of self-synchronizing scrambled sequences

We derive a closed-form expression for the power spectral density of amplitude/phase-shift keyed bit sequences randomized through self-synchronizing scrambling when the source sequence is a stationary sequence of statistically independent bits. In addition to the dependence on the symbol pulse shape, duration, and the signal space values with which symbols are represented, we show that the power spectral density is dependent only on the probability of logic ones in the source bit stream, the period of the impulse response of the scrambling shift register, and the number of logic ones in this period. Our results confirm that optimum randomization results with use of primitive scrambling polynomials and poorest randomization occurs with two-tap polynomials of the form x/sup D/+1.

映像情報ストレージ技術 ガロア体加算に基づいた直流成分抑圧方式とその性能評価

A DC component suppressing method, called Guided Scrambling (GS), has been proposed. GS constructs a selection set of channel bit streams by subjecting a source bit stream within a data block to several kinds of scrambling and to RLL (Run Length Limited) coding. The one which has the least DC component is then selected. Typically, this technique uses a convolutional operation or GF (Galois field) multiplication. The convolutional GS is good at suppressing the DC component, but has poor performance in the symbol error rate because a bit error propagates to an adjacent codeword before RS decoding. In the GF multiplicative GS, the error rate is lower, but performance in DC component suppression is worse, especially when a simple hardware configuration is used or when a source bit stream is not random. In this paper, we propose the GF additive GS method which generates a better channel bit stream with both these characteristics. We also analyze the spectrum and the average symbol error rate of this method with computer simulations, and show its advantages over other GSs.