442 research outputs found
Fuzzy Logic Control of Adaptive ARQ for Video Distribution over a Bluetooth Wireless Link
Bluetooth's default automatic repeat request (ARQ) scheme is not suited to video distribution resulting in missed display and decoded deadlines. Adaptive ARQ with active discard of expired packets from the send buffer is an alternative approach. However, even with the addition of cross-layer adaptation to picture-type packet importance, ARQ is not ideal in conditions of a deteriorating RF channel. The paper presents fuzzy logic control of ARQ, based on send buffer fullness and the head-of-line packet's deadline. The advantage of the fuzzy logic approach, which also scales its output according to picture type importance, is that the impact of delay can be directly introduced to the model, causing retransmissions to be reduced compared to all other schemes. The scheme considers both the delay constraints of the video stream and at the same time avoids send buffer overflow. Tests explore a variety of Bluetooth send buffer sizes and channel conditions. For adverse channel conditions and buffer size, the tests show an improvement of at least 4 dB in video quality compared to nonfuzzy schemes. The scheme can be applied to any codec with I-, P-, and (possibly) B-slices by inspection of packet headers without the need for encoder intervention.</jats:p
Maximum Average Service Rate and Optimal Queue Scheduling of Delay-Constrained Hybrid Cognitive Radio in Nakagami Fading Channels
As a promising technique to improve achievable bandwidth efficiency, cognitive radio (CR) has attracted substantial research attention from both the academic and industrial communities. To improve the performance attained by the secondary user (SU), a novel hybrid CR system is proposed, which combines the conventional interweave and underlay paradigms to enhance the chance of the SU to access the spectrum. Queuing theory is invoked in this paper to analyze the impact of the primary user’s maximum tolerable delay on the performance of the SU. Multiple queues are assumed for the SU, which is engaged in video communication. Apart from the Poisson traffic generation,we also model the classic Nakagami-m fading channel as a Poisson service process by utilizing the outage probability in the presence of cochannel interference. We optimize both the hybrid interweave/underlay procedure to maximize the average service rate μ_S,max of the SU, as well as the queue’s scheduling scheme, for the sake of minimizing the overall average delay (OAD). As a result, the OAD of the SU is reduced by up to 27% and 20%, compared with the proportion and round-robin schemes, respectively
Error resilient packet switched H.264 video telephony over third generation networks.
Real-time video communication over wireless networks is a challenging problem because
wireless channels suffer from fading, additive noise and interference, which translate
into packet loss and delay. Since modern video encoders deliver video packets with
decoding dependencies, packet loss and delay can significantly degrade the video quality
at the receiver. Many error resilience mechanisms have been proposed to combat packet
loss in wireless networks, but only a few were specifically designed for packet switched
video telephony over Third Generation (3G) networks.
The first part of the thesis presents an error resilience technique for packet switched
video telephony that combines application layer Forward Error Correction (FEC) with
rateless codes, Reference Picture Selection (RPS) and cross layer optimization. Rateless
codes have lower encoding and decoding computational complexity compared to traditional
error correcting codes. One can use them on complexity constrained hand-held
devices. Also, their redundancy does not need to be fixed in advance and any number of
encoded symbols can be generated on the fly. Reference picture selection is used to limit
the effect of spatio-temporal error propagation. Limiting the effect of spatio-temporal
error propagation results in better video quality. Cross layer optimization is used to
minimize the data loss at the application layer when data is lost at the data link layer.
Experimental results on a High Speed Packet Access (HSPA) network simulator for
H.264 compressed standard video sequences show that the proposed technique achieves
significant Peak Signal to Noise Ratio (PSNR) and Percentage Degraded Video Duration
(PDVD) improvements over a state of the art error resilience technique known as
Interactive Error Control (IEC), which is a combination of Error Tracking and feedback
based Reference Picture Selection. The improvement is obtained at a cost of higher
end-to-end delay.
The proposed technique is improved by making the FEC (Rateless code) redundancy
channel adaptive. Automatic Repeat Request (ARQ) is used to adjust the redundancy
of the Rateless codes according to the channel conditions. Experimental results show
that the channel adaptive scheme achieves significant PSNR and PDVD improvements
over the static scheme for a simulated Long Term Evolution (LTE) network.
In the third part of the thesis, the performance of the previous two schemes is
improved by making the transmitter predict when rateless decoding will fail. In this
case, reference picture selection is invoked early and transmission of encoded symbols
for that source block is aborted. Simulations for an LTE network show that this results
in video quality improvement and bandwidth savings.
In the last part of the thesis, the performance of the adaptive technique is improved
by exploiting the history of the wireless channel. In a Rayleigh fading wireless channel,
the RLC-PDU losses are correlated under certain conditions. This correlation is
exploited to adjust the redundancy of the Rateless code and results in higher Rateless
code decoding success rate and higher video quality. Simulations for an LTE network
show that the improvement was significant when the packet loss rate in the two wireless
links was 10%.
To facilitate the implementation of the proposed error resilience techniques in practical
scenarios, RTP/UDP/IP level packetization schemes are also proposed for each
error resilience technique.
Compared to existing work, the proposed error resilience techniques provide better
video quality. Also, more emphasis is given to implementation issues in 3G networks
Joint source channel coding for progressive image transmission
Recent wavelet-based image compression algorithms achieve best ever performances with fully embedded bit streams. However, those embedded bit streams are very sensitive to channel noise and protections from channel coding are necessary. Typical error correcting capability of channel codes varies according to different channel conditions. Thus, separate design leads to performance degradation relative to what could be achieved through joint design. In joint source-channel coding schemes, the choice of source coding parameters may vary over time and channel conditions. In this research, we proposed a general approach for the evaluation of such joint source-channel coding scheme. Instead of using the average peak signal to noise ratio (PSNR) or distortion as the performance metric, we represent the system performance by its average error-free source coding rate, which is further shown to be an equivalent metric in the optimization problems.
The transmissions of embedded image bit streams over memory channels and binary symmetric channels (BSCs) are investigated in this dissertation. Mathematical models were obtained in closed-form by error sequence analysis (ESA). Not surprisingly, models for BSCs are just special cases for those of memory channels. It is also discovered that existing techniques for performance evaluation on memory channels are special cases of this new approach. We further extend the idea to the unequal error protection (UEP) of embedded images sources in BSCs. The optimization problems are completely defined and solved. Compared to the equal error protection (EEP) schemes, about 0.3 dB performance gain is achieved by UEP for typical BSCs. For some memory channel conditions, the performance improvements can be up to 3 dB. Transmission of embedded image bit streams in channels with feedback are also investigated based on the model for memory channels. Compared to the best possible performance achieved on feed forward transmission, feedback leads to about 1.7 dB performance improvement
Multimedia streaming over wireless channels
The improvements in mobile communication systems have accelerated the development of new multimedia streaming techniques to increase the quality of streaming data over time varying wireless channels. In order to increase multimedia quality, error control schemes are indispensable due to time-varying and erroneous nature of the channel. However, relatively low channel capacity of wireless channels, and dependency structure in multimedia limit the eectiveness of existing error control schemes and require more sophisticated techniques to provide quality improvement on the streaming data. In this thesis, we propose sender driven multimedia streaming algorithms that incorporate error control schemes of FEC, ARQ, and packet scheduling by considering media and channel parameters such as packet importance, packet dependencies, decoding deadlines, channel state information, and channel capacity. Initially, we have proposed a multi-rate distortion optimization framework so as to jointly optimize FEC rate and packet selection by minimizing end-to-end distortion to satisfy a specified Quality of Service under channel capacity constraint. Minimization of end-to-end distortion causes computational complexity in the rate distortion optimization framework due to dependency in encoded multimedia. Therefore, we propose multimedia streaming algorithms that select packet and FEC rate with reduced computational complexity and high quality as compared with multi-rate distortion optimization framework. Additionally, protocol stack of a UMTS cellular network system with W-CDMA air interface is presented in order to clarify the relation between proposed multimedia streaming algorithms and UMTS system that is used in simulations. Finally, proposed algorithms are simulated and results demonstrate that proposed algorithms improve multimedia quality significantly as compared to existing methods
Video QoS/QoE over IEEE802.11n/ac: A Contemporary Survey
The demand for video applications over wireless networks has tremendously increased, and IEEE 802.11 standards have provided higher support for video transmission. However, providing Quality of Service (QoS) and Quality of Experience (QoE) for video over WLAN is still a challenge due to the error sensitivity of compressed video and dynamic channels. This thesis presents a contemporary survey study on video QoS/QoE over WLAN issues and solutions. The objective of the study is to provide an overview of the issues by conducting a background study on the video codecs and their features and characteristics, followed by studying QoS and QoE support in IEEE 802.11 standards. Since IEEE 802.11n is the current standard that is mostly deployed worldwide and IEEE 802.11ac is the upcoming standard, this survey study aims to investigate the most recent video QoS/QoE solutions based on these two standards. The solutions are divided into two broad categories, academic solutions, and vendor solutions. Academic solutions are mostly based on three main layers, namely Application, Media Access Control (MAC) and Physical (PHY) which are further divided into two major categories, single-layer solutions, and cross-layer solutions. Single-layer solutions are those which focus on a single layer to enhance the video transmission performance over WLAN. Cross-layer solutions involve two or more layers to provide a single QoS solution for video over WLAN. This thesis has also presented and technically analyzed QoS solutions by three popular vendors. This thesis concludes that single-layer solutions are not directly related to video QoS/QoE, and cross-layer solutions are performing better than single-layer solutions, but they are much more complicated and not easy to be implemented. Most vendors rely on their network infrastructure to provide QoS for multimedia applications. They have their techniques and mechanisms, but the concept of providing QoS/QoE for video is almost the same because they are using the same standards and rely on Wi-Fi Multimedia (WMM) to provide QoS
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