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This paper describes a new measurement method (VS model) for determining the quality of online video services relying on modern H.265/HEVC and VP9 codecs. The said method has been developed on the basis of VQuad-HD curves (according to ITU-T J.341). This model does not refer to signal analysis, but protocol analysis instead. The parameters used are: type of video codec, encoding rate, transport technique, packet loss and burst size. The method may be implemented quickly and easily, which is one of the great advantages when using this method to measure QoS. [ABSTRACT FROM AUTHOR]

Efficient video compression algorithms in advanced multimedia broadcasting systems are in high demand. In the last decades, different video compression tools have been developed which can influence the final Quality of Experience in different ways, his paper has two goals. The first goal is to present a study of different compression algorithms available for stereoscopic 3D videos. The second goal is to present the possibilities in the creation of new stereoscopic models. The well-established video codecs (AVC, MVC, HEVC and MV-HEVC) are considered as encoders. Generic objective video quality metrics are used to analyze the compression efficiencies of the considered codecs, extended with results from subjective tests. The correlations between the objective and subjective scores are analyzed statistically. Due to unsatisfactory results of generic 2D metrics for the stereoscopic sequences used in the test, new objective models are presented. Such models show improved correlation with subjective stereoscopic video quality. The validation, verification and a description of models are presented in detail. [ABSTRACT FROM AUTHOR]

Adaptive Multi-Rate Codec is one of the codecs which is used for making voice calls. It helps to connect people who are scattered in various geographical areas. It adjusts its bit-rate according to the user's channel conditions. It plays a vital role in providing an improved speech quality of voice connection in Long Term Evolution (LTE). There are some constraints which need to be addressed in providing this service profitably. Quality of Service (QoS) is the dominant mechanism which determines the quality of the speech in communication. On several occasions, number of users are trying to access the same channel simultaneously by standing in a particular region for a longer period of time. It refers to Multi-user channel sharing problem which leads to resource loss very often. The main aim of this paper is to develop a novel RSA -- QoS scheduling algorithm for reducing the Resource Loss Ratio. Eventually, it increases the throughput.The simulation result shows that the RSA -- QoS increases the number of users for accessing the resources better than the existing algorithms in terms of resource loss and throughput. Ultimately, it enhances the QoS in Mobile Networks. [ABSTRACT FROM AUTHOR]

This paper describes the scalable extension of High Efficiency Video Coding (HEVC) to provide flexible highquality digital video content services. The proposed scalable codec is designed on multi-loop decoding architecture to support inter-layer sample prediction and inter-layer motion parameter prediction. Inter-layer sample prediction is enabled by inserting the reconstructed picture of the reference layer (RL) into the decoded picture buffer of the enhancement layer (EL). To reduce the motion parameter redundancies between layers, the motion parameter of the RL is used as one of the candidates in merge mode and motion vector prediction in the EL. The proposed scalable extension can support scalabilities with minimum changes to the HEVC and provide average Bjøntegaard delta bitrate gains of about 24% for spatial scalability and of about 21% for SNR scalability compared to simulcast coding with HEVC. [ABSTRACT FROM AUTHOR]

The perceived audio quality of a digital broadcasting system such as DAB+ is dependent on what type of coding and bit rates are applied. Due to bandwidth constraints, audio quality is prone to be in conflict with other service demands such as the number of channels and the transfer of ancillary data. Compared to DAB+, several other audio services have superior bit rates that challenge the audio quality of DAB+. This paper reviews audio-quality criteria and investigates how the perceived audio quality of different broadcasting systems complies with the criteria. Two studies were conducted: Test 1 where DAB+ codecs were tested at bit rates between 96 and 192 kbit/s and Test 2 where DAB+ systems between 48 and 192 kbit/s as well as FM systems were tested. The systems in Test 2 were designed to as far as possible model a realistic broadcasting signal chain including commonly used dynamic processors. The studies were realized through two listening experiments using the ITU-R recommendations BS. 1116 and B S. 1534 (MUSHRA) followed by a closing interview. The results showed that the currently highest available subchannel bit rate for DAB+ (192 kbit/s) was insufficient for attaining perceptually transparent quality for critical items, whereas it enabled a quality comparable to or in some instances better than a modern FM system. Extrapolation of data indicates that critical items may need to be coded at even higher bit rates to reach perceptually transparent quality. From the interviews, auditory features important for the subjects' assessment of quality were observed. This study concludes that when making decisions on broadcasting systems, it is important to have well-founded and clearly defined criteria for minimum acceptable quality and/or perceptually transparent quality. [ABSTRACT FROM AUTHOR]

Non-real-time delivery of stereoscopic video has been considered as a service scenario for 3DTV to overcome the limited bandwidth in the terrestrial digital television system. A hybrid codec combining MPEG-2 and H.264/AVC has been suggested for the compression of stereoscopic video for 3DTV. In this paper, we propose a stereoscopic video coding scheme using adaptive pre-/post-filters (APPF) to improve the quality of 3D video while retaining compatibility with legacy video coding standards. The APPF are applied adaptively to blocks of various sizes determined by the macroblock coding mode and reference frame index. Experiment results show that the proposed method achieves up to 24.86% bit rate savings relative to a hybrid codec of MPEG-2 and H.264/AVC including the inter-view prediction. [ABSTRACT FROM AUTHOR]