Supporting real time video over ATM networks

Includes bibliographical references.In this project, we propose and evaluate an approach to delimit and tag such independent video slice at the ATM layer for early discard. This involves the use of a tag cell differentiated from the rest of the data by its PTI value and a modified tag switch to facilitate the selective discarding of affected cells within each video slice as opposed to dropping of cells at random from multiple video frames

Flow control of real-time unicast multimedia applications in best-effort networks

One of the fastest growing segments of Internet applications are real-time mul- timedia applications, like Voice over Internet Protocol (VoIP). Real-time multimedia applications use the User Datagram Protocol (UDP) as the transport protocol because of the inherent conservative nature of the congestion avoidance schemes of Transmis- sion Control Protocol (TCP). The e®ects of uncontrolled °ows on the Internet have not yet been felt because UDP tra±c frequently constitutes only » 20% of the total Internet tra±c. It is pertinent that real-time multimedia applications become better citizens of the Internet, while at the same time deliver acceptable Quality of Service (QoS). Traditionally, packet losses and the increase in the end-to-end delay experienced by some of the packets characterizes congestion in the network. These two signals have been used to develop most known °ow control schemes. The current research considers the °ow accumulation in the network as the signal for use in °ow control. The most signi¯cant contribution of the current research is to propose novel end- to-end °ow control schemes for unicast real-time multimedia °ows transmitting over best-e®ort networks. These control schemes are based on predictive control of the accumulation signal. The end-to-end control schemes available in the literature are based on reactive control that do not take into account the feedback delay existing between the sender and the receiver nor the forward delay in the °ow dynamics. The performance of the proposed control schemes has been evaluated using the ns-2 simulation environment. The research concludes that active control of hard real- time °ows delivers the same or somewhat better QoS as High Bit Rate (HBR, no control), but with a lower average bit rate. Consequently, it helps reduce bandwidth use of controlled real-time °ows by anywhere between 31:43% to 43:96%. Proposed reactive control schemes deliver good QoS. However, they do not scale up as well as the predictive control schemes. Proposed predictive control schemes are e®ective in delivering good quality QoS while using up less bandwidth than even the reactive con- trol schemes. They scale up well as more real-time multimedia °ows start employing them

Enhanced quality reconstruction of erroneous video streams using packet filtering based on non-desynchronizing bits and UDP checksum-filtered list decoding

The latest video coding standards, such as H.264 and H.265, are extremely vulnerable in error-prone networks. Due to their sophisticated spatial and temporal prediction tools, the effect of an error is not limited to the erroneous area but it can easily propagate spatially to the neighboring blocks and temporally to the following frames. Thus, reconstructed video packets at the decoder side may exhibit significant visual quality degradation. Error concealment and error corrections are two mechanisms that have been developed to improve the quality of reconstructed frames in the presence of errors. In most existing error concealment approaches, the corrupted packets are ignored and only the correctly received information of the surrounding areas (spatially and/or temporally) is used to recover the erroneous area. This is due to the fact that there is no perfect error detection mechanism to identify correctly received blocks within a corrupted packet, and moreover because of the desynchronization problem caused by the transmission errors on the variable-length code (VLC). But, as many studies have shown, the corrupted packets may contain valuable information that can be used to reconstruct adequately of the lost area (e.g. when the error is located at the end of a slice). On the other hand, error correction approaches, such as list decoding, exploit the corrupted packets to generate several candidate transmitted packets from the corrupted received packet. They then select, among these candidates, the one with the highest likelihood of being the transmitted packet based on the available soft information (e.g. log-likelihood ratio (LLR) of each bit). However, list decoding approaches suffer from a large solution space of candidate transmitted packets. This is worsened when the soft information is not available at the application layer; a more realistic scenario in practice. Indeed, since it is unknown which bits have higher probabilities of having been modified during transmission, the candidate received packets cannot be ranked by likelihood. In this thesis, we propose various strategies to improve the quality of reconstructed packets which have been lightly damaged during transmission (e.g. at most a single error per packet). We first propose a simple but efficient mechanism to filter damaged packets in order to retain those likely to lead to a very good reconstruction and discard the others. This method can be used as a complement to most existing concealment approaches to enhance their performance. The method is based on the novel concept of non-desynchronizing bits (NDBs) defined, in the context of an H.264 context-adaptive variable-length coding (CAVLC) coded sequence, as a bit whose inversion does not cause desynchronization at the bitstream level nor changes the number of decoded macroblocks. We establish that, on typical coded bitstreams, the NDBs constitute about a one-third (about 30%) of a bitstream, and that the effect on visual quality of flipping one of them in a packet is mostly insignificant. In most cases (90%), the quality of the reconstructed packet when modifying an individual NDB is almost the same as the intact one. We thus demonstrate that keeping, under certain conditions, a corrupted packet as a candidate for the lost area can provide better visual quality compared to the concealment approaches. We finally propose a non-desync-based decoding framework, which retains a corrupted packet, under the condition of not causing desynchronization and not altering the number of expected macroblocks. The framework can be combined with most current concealment approaches. The proposed approach is compared to the frame copy (FC) concealment of Joint Model (JM) software (JM-FC) and a state-of-the-art concealment approach using the spatiotemporal boundary matching algorithm (STBMA) mechanism, in the case of one bit in error, and on average, respectively, provides 3.5 dB and 1.42 dB gain over them. We then propose a novel list decoding approach called checksum-filtered list decoding (CFLD) which can correct a packet at the bit stream level by exploiting the receiver side user datagram protocol (UDP) checksum value. The proposed approach is able to identify the possible locations of errors by analyzing the pattern of the calculated UDP checksum on the corrupted packet. This makes it possible to considerably reduce the number of candidate transmitted packets in comparison to conventional list decoding approaches, especially when no soft information is available. When a packet composed of N bits contains a single bit in error, instead of considering N candidate packets, as is the case in conventional list decoding approaches, the proposed approach considers approximately N/32 candidate packets, leading to a 97% reduction in the number of candidates. This reduction can increase to 99.6% in the case of a two-bit error. The method’s performance is evaluated using H.264 and high efficiency video coding (HEVC) test model software. We show that, in the case H.264 coded sequence, on average, the CFLD approach is able to correct the packet 66% of the time. It also offers a 2.74 dB gain over JM-FC and 1.14 dB and 1.42 dB gains over STBMA and hard output maximum likelihood decoding (HO-MLD), respectively. Additionally, in the case of HEVC, the CFLD approach corrects the corrupted packet 91% of the time, and offers 2.35 dB and 4.97 dB gains over our implementation of FC concealment in HEVC test model software (HM-FC) in class B (1920×1080) and C (832×480) sequences, respectively

Emotion Recognition from Speech with Acoustic, Non-Linear and Wavelet-based Features Extracted in Different Acoustic Conditions

ABSTRACT: In the last years, there has a great progress in automatic speech recognition. The challenge now it is not only recognize the semantic content in the speech but also the called "paralinguistic" aspects of the speech, including the emotions, and the personality of the speaker. This research work aims in the development of a methodology for the automatic emotion recognition from speech signals in non-controlled noise conditions. For that purpose, different sets of acoustic, non-linear, and wavelet based features are used to characterize emotions in different databases created for such purpose

Network Factors Influencing Packet Loss in Online Games

In real-time communications it is often vital that data arrive at its destination in a timely fashion. Whether it is the user experience of online games, or the reliability of tele-surgery, a reliable, consistent and predictable communications channel between source and destination is important. However, the Internet as we know it was designed to ensure that data will arrive at the desired destination instead of being designed for predictable, low-latency communication. Data traveling from point to point on the Internet is comprised of smaller packages known as packets. As these packets traverse the Internet, they encounter routers or similar devices that will often queue the packets before sending them toward their destination. Queued packets introduces a delay that depends greatly on the router configuration and the number of other packets that exist on the network. In times of high demand, packets may be discarded by the router or even lost in transmission. Protocols exist that retransmit lost packets, but these protocols introduce additional overhead and delays - costs that may be prohibitive in some applications. Being able to predict when packets may be delayed or lost could allow applications to compensate for unreliable data channels. In this thesis I investigate the effects of cross traffic and router configuration on a low bandwidth traffic stream such as that which is common in games. The experiments investigate the effects of cross traffic packet size, bit-rate, inter-packet timing and protocol used. The experiments also investigate router configurations including queue management type and the number of queues. These experiments are compared to real-world data and a mitigation strategy, where n previous packets are bundled with each new packet, is applied to both the simulated data and the real-world captures. The experiments indicate that most of the parameters explored had an impact on the packet loss. However, the real world data and simulated data differ and would require additional work to attempt to apply the lessons learned to real world applications. The mitigation strategy appeared to work well, allowing 90\% of all runs to complete without data loss. However, the mitigation strategy was implemented analytically and the actual implementation and testing has been left for future work