Mobile video traffic is expected to increase explosively because of the proliferating number of Wi-Fi terminals. An overloaded multiple-input multiple-output (MIMO) technique allows the receiver to implement smaller number of antennas than the transmitter in exchange for degradation in video quality and a large amount of computational complexity for postcoding at the receiver side. This paper proposes a novel linear precoder for high-quality video streaming in overloaded multiuser MIMO systems, which protects visually significant portions of a video stream. A low complexity postcoder is also proposed, which detects some of data symbols by linear detection and the others by a prevoting vector cancellation (PVC) approach. It is shown from simulation results that the combination use of the proposed precoder and postcoder achieves higher-quality video streaming to multiple users in a wider range of signal-to-noise ratio (SNR) than a conventional unequal error protection scheme. The proposed precoder attains 40dB in peak signal-to-noise ratio even in poor channel conditions such as the SNR of 12dB. In addition, due to the stepwise acquisition of data symbols by means of linear detection and PVC, the proposed postcoder reduces the number of complex additions by 76% and that of multiplications by 64% compared to the conventional PVC.

High-resolution image and video communication in home networks is highly expected to proliferate with the spread of Wi-Fi devices and the introduction of multiple-input multiple-output (MIMO) systems. This paper proposes a joint transmission and coding scheme for broadcasting high-resolution video streams over multiuser MIMO systems with an eigenbeam-space division multiplexing (E-SDM) technique. Scalable video coding makes it possible to produce the code stream comprised of multiple layers having unequal contribution to image quality. The proposed scheme jointly assigns the data of scalable code streams to subcarriers and spatial streams based on their signal-to-noise ratio (SNR) values in order to transmit visually important data with high reliability. Simulation results show that the proposed scheme surpasses the conventional unequal power allocation (UPA) approach in terms of both peak signal-to-noise ratio (PSNR) of received images and correct decoding probability. PSNR performance of the proposed scheme exceeds 35dB with the probability of over 95% when received SNR is higher than 6dB. The improvement in average PSNR by the proposed scheme compared to the conventional UPA comes up to approx. 20dB at received SNR of 6dB. Furthermore, correct decoding probability reaches 95% when received SNR is greater than 4dB.

Joint source-channel coding (JSCC) is a method to jointly allocate the given total transmission bitrate to the source coding and channel coding to maximize the video quality at the receiving end. In this paper, we propose a practical model for efficiently determining a near-optimal code rate for JSCC in real-time video communications. The conventional code rate decision schemes using analytical source coding distortion model and channel-induced distortion model are usually complex, and typically employ the process of model parameter training which involves potentially high computational complexity and implementation cost. To avoid the complex modeling procedure, we introduce a very simple video quality model based on the playable bitrate which is defined as the total bit amount per unit time that is not affected by the channel loss during transmission including correctly recovered bits by the channel decoder. Because the video quality at the receiving end is clearly commensurate with the playable bitrate, we can easily determine the quality-oriented near-optimal code rate by finding the code rate that maximizes the playable bitrate at the sender side. The proposed playable bitrate model is very simple because it does not require the complex training procedure for obtaining model parameters, which is usually required in the conventional code rate decision method. It is shown by simulations that the proposed code rate decision scheme based on the playable bitrate model can efficiently determine the near-optimal code rate for JSCC in terms of high accuracy on the optimal code rate.

The joint source-channel coding problem is considered. The joint source-channel coding theorem reveals the existence of a code for the pair of the source and the channel under the condition that the error probability is smaller than or equal to ε asymptotically. The separation theorem guarantees that we can achieve the optimal coding performance by using the two-stage coding. In the case that ε = 0, Han showed the joint source-channel coding theorem and the separation theorem for general sources and channels. Furthermore the ε-coding theorem (0 ≤ ε <1) in the similar setting was studied. However, the separation theorem was not revealed since it is difficult in general. So we consider the separation theorem in this setting.

A channel code is constructed using sparse matrices for stationary memoryless channels that do not necessarily have a symmetric property like a binary symmetric channel. It is also shown that the constructed code has the following remarkable properties. 1. Joint source-channel coding: Combining channel code with lossy source code, which is also constructed by sparse matrices, a simpler joint source-channel code can be constructed than that constructed by the ordinary block code. 2. Universal coding: The constructed channel code has a universal property under a specified condition.

When the joint source-channel (JSC) decoder is used for source coding over noisy channels, the JSC decoder may invent errors even though the received data is not corrupted by the channel noise, if the JSC decoder assumes the channel was noisy. A novel encoder algorithm has been recently proposed to improve the performance of the communications system under this situation. In this letter, we propose another algorithm based on conditional entropy-constrained vector quantizer to further improve the encoder. The algorithm proposed in this letter significantly improves the performance of the communications system when the hypothesized channel bit error rate is high.

Improving the overall performance of reliable speech communication in ultrashort wave radios over very noisy channels is of great importance and practical use. An iterative joint source-channel (de-)coding and (de-)modulation (JSCCM) algorithm is proposed for ITU-T Rec.G.729EV by both exploiting the residual redundancy and passing soft information throughout the receiver while introducing a systematic global iteration process. Being fully compatible with existing transmitter structure, the proposed algorithm does not introduce additional bandwidth expansion and transmission delay. Simulations show substantial error correcting performance and synthesized speech quality improvement over conventional separate designed systems in delay and bandwidth constraint channels by using the JSCCM algorithm.

In this paper, Information-Spectrum characterization is derived for the reliable transmission of general correlated sources over the general multiple-access channels. We consider the necessary and sufficient conditions for the transmission of general correlated sources over the general multiple-access channels by using Information-Spectrum methods which are introduced by Han and Verdu.

This paper clarifies a necessary condition and a sufficient condition for transmissibility for a given set of general sources and a given general broadcast channel. The approach is based on the information-spectrum methods introduced by Han and Verdu. Moreover, we consider the capacity region of the general broadcast channel with arbitrarily fixed error probabilities if we send independent private and common messages over the channel. Furthermore, we treat the capacity region for mixed broadcast channel.

This paper proposes a new method, EC-MDC that can detect and correct bit errors in the bitstream generated by multiple description coding. The proposed method generates two sub-bitstreams having a few redundancies as much as conventional multiple description coding. If a sub-bitstream at one side has bit error, the bit error can be corrected by using sub-bitstream of the other side. In BER-SNR experiments, reconstruction quality of the proposed method shows about 11dB higher than that of the conventional MDC at BER < 10-3 when a sub-bitstream is corrupted.