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Content-adaptive bitstream-layer model for coding distortion assessment of H.264/AVC networked video

JOURNAL OF VISUAL COMMUNICATION AND IMAGE REPRESENTATION 2014年第5期25卷 1199-1208页

作者： Su, Honglei Yang, Fuzheng Xidian Univ State Key Lab Integrated Serv Networks Xian Peoples R China

Bitstream-layer models are designed to use the information extracted from both packet headers and payload for real-time and non-intrusive quality monitoring of networked video. This paper proposes a content-adaptive bitstream-layer (CABL) model for coding distortion assessment of H.264/AVC networked video. Firstly, the fundamental relationship between perceived coding distortion and quantization parameter (QP) is established. Then, considering the fact that the perceived coding distortion of a networked video significantly relies on both the spatial and temporal characteristics of video content, spatial and temporal complexities are incorporated in the proposed model. Assuming that the residuals before Discrete Cosine Transform (DCT) keep to the Laplace distribution, the scale parameters of the Laplace distribution are estimated utilizing QP and quantized coefficients on the basis of the Parseval theorem firstly. Then the spatial complexity is evaluated using QP and the scale parameters. Meanwhile, the temporal complexity is obtained using the weighted motion vectors (MV) considering the variations in temporal masking extent for high motion regions and low motion regions, respectively. Both the two characteristics of video content are extracted from the compressed bitstream without resorting to a complete decoding. Using content related information, the proposed model is able to adapt to different video contents. Experimental results show that the overall performance of CABL model significantly outperforms that of the P.1202.1 model and other coding distortion assessment models in terms of widely used performance criteria, including the Pearson Correlation Coefficient (PCC), the Spearman Rank Order Correlation Coefficient (SROCC), the Root-Mean-Squared Error (RMSE) and the Outlier Ratio (OR). (c) 2014 Elsevier Inc. All rights reserved.

关键词： No reference Video quality assessment Bitstream-layer model coding distortion H.264/AVC Networked video Spatial complexity Temporal complexity

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Modeling, estimating, and compensating low-bit rate coding distortion in speech recognition

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IEEE TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE PROCESSING 2006年第1期14卷 246-255页

作者： Yoma, NB Molina, C Silva, J Busso, C Univ Chile Dept Elect Engn Santiago Chile

A solution to the problem of speech recognition with signals distorted by low-bit rate coders is presented in this paper. A model for the coding-decoding distortion, a HMM compensation method to include this model, and an EM-based adaptation algorithm to estimate this distortion are proposed here. Medium vocabulary continuous-speech speaker-independent recognition experiments with 8 kbps G.729(CS-CELP), 13 kbps RPE-LTP (GSM), 5.3 kbps G723.1, 4.8 kbps FS- 1016 and 32 kbps G.726(ADPCM) coders show that the approach described in this paper is able to dramatically reduce the effect of the coding distortion and, in some cases, gives a word accuracy higher than the baseline system with uncoded speech. Finally, the EM estimation algorithm requires only one adapting utterance and the approach described is certainly suitable for dialogue systems where just a few adapting utterances are available.

关键词： coding distortion EM estimation algorithm HMM compensation low-bit rate coders speech recognition

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A Novel Analytic Quantization-distortion Model for Hybrid Video coding

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IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMS FOR VIDEO TECHNOLOGY 2009年第5期19卷 627-641页

作者： Guo, Liwei Au, Oscar C. Ma, Mengyao Liang, Zhiqin Wong, Peter H. W. Multimedia Technol Res Ctr Hong Kong Hong Kong Peoples R China Hong Kong Univ Sci & Technol Dept Comp Sci Hong Kong Hong Kong Peoples R China Harmonic Inc Video Demand Dept Hong Kong Hong Kong Peoples R China Mobile Ltd VP Dynam Labs Hong Kong Hong Kong Peoples R China

A proper theoretical quantization-distortion model for hybrid video coding is always desirable, since this allows us to explain the behavior of existing codecs and to design better ones. However, due to the existence of motion-compensated prediction, hybrid video coding introduces interframe dependency into the encoded video, which makes its quantization-distortion characteristics difficult to analyze. In this paper, a joint analysis of quantization and motion-compensated prediction is presented. For a complete analysis, we investigate not only the distortion that quantization introduces into video signal, but also its effect on motion-compensated prediction. Based on the joint analysis, a quantization-distortion model of hybrid video coding is proposed. Our extensive experimental results show that the proposed model can estimate the quantization-distortion curve of hybrid video coding with high accuracy. Furthermore, the estimation accuracy remains high for various video sequences and encoder configurations.

关键词： coding distortion hybrid video coding motion-compensated prediction quantization video modeling

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A comparison of front-ends for bitstream-based ASR over IP

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SIGNAL PROCESSING 2006年第7期86卷 1502-1508页

作者： Pelaez-Moreno, Carmen Gallardo-Antolin, Ascensin Gomez-Cajas, Diego F. Diaz-De-Maria, Fernando EPS Univ Carlos III Madrid Dept Teor Senal & Comunicac Madrid 28911 Spain

Automatic speech recognition (ASR) is. called to play a relevant role in the provision of spoken interfaces for IP-based applications. However, as a consequence of the transit of the speech signal over these particular networks, ASR systems need to face two new challenges: the impoverishment of the speech quality due to the compression needed to fit the channel capacity and the inevitable occurrence of packet losses. In this framework, bitstream-based approaches that obtain the ASR feature vectors directly from the coded bitstream, avoiding the speech decoding process, have been proposed ([S.H. Choi, H.K. Kim, H.S. Lee, Speech recognition using quantized LSP parameters and their transformations in digital communications, Speech Commun. 30 (4) (2000) 223-233. A. Gallardo-Antolin, C. Pelaez-Moreno, F. Diaz-de-Maria, Recognizing GSM digital speech, IEEE Trans. Speech Audio Process., to appear. H.K. Kim, R.V. Cox, R.C. Rose, Performance improvement of a bitstream-based front-end for wireless speech recognition in adverse environments, IEEE Trans. Speech Audio Process. 10 (8) (2002) 591-604. C. Pelaez-Moreno, A. Gallardo-Antolin, F. Diaz-de-Maria, Recognizing voice over IP networks: a robust front-end for speech recognition on the WWW, IEEE Trans. Multimedia 3(2) (2001) 209-218], among others) to improve the robustness of ASR systems. LSP (Line Spectral Pairs) are the preferred set of parameters for the description of the speech spectral envelope in most of the modern speech coders. Nevertheless, LSP have proved to be unsuitable for ASR, and they must be transformed into cepstrum-type parameters. In this paper we comparatively evaluate the robustness of the most significant LSP to cepstrum transformations in a simulated VoIP (voice over IP) environment which includes two of the most popular codecs used in that network (G.723.1 and G.729) and several network conditions. In particular, we compare 'pseudocepstrum' [H.K. Kim, S.H. Choi, H.S. Lee, On approximating Line Spectr

关键词： robust speech recognition speech coding IP networks coding distortion packet loss LSP

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No-Reference Quality Assessment for Networked Video via Primary Analysis of Bit Stream

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IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMS FOR VIDEO TECHNOLOGY 2010年第11期20卷 1544-1554页

作者： Yang, Fuzheng Wan, Shuai Xie, Qingpeng Wu, Hong Ren Xidian Univ State Key Lab Integrated Serv Networks Xian 710071 Peoples R China Northwestern Polytech Univ Sch Elect & Informat Xian 710129 Peoples R China Huawei Technol Co Ltd Shenzhen 518129 Peoples R China RMIT Univ Sch Elect & Comp Engn Melbourne Vic 3001 Australia

A no-reference (NR) quality measure for networked video is introduced using information extracted from the compressed bit stream without resorting to complete video decoding. This NR video quality assessment measure accounts for three key factors which affect the overall perceived picture quality of networked video, namely, picture distortion caused by quantization, quality degradation due to packet loss and error propagation, and temporal effects of the human visual system. First, the picture quality in the spatial domain is measured, for each frame, relative to quantization under an error-free transmission condition. Second, picture quality is evaluated with respect to packet loss and the subsequent error propagation. The video frame quality in the spatial domain is, therefore, jointly determined by coding distortion and packet loss. Third, a pooling scheme is devised as the last step of the proposed quality measure to capture the perceived quality degradation in the temporal domain. The results obtained by performance evaluations using MPEG-4 coded video streams have demonstrated the effectiveness of the proposed NR video quality metric.

关键词： coding distortion networked video no-reference video quality assessment packet loss temporal pooling

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Recognizing voice aver IP:: A robust front-end for speech recognition on the World Wide Web

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IEEE TRANSACTIONS ON MULTIMEDIA 2001年第2期3卷 209-218页

作者： Peláez-Moreno, C Gallardo-Antolín, A Díaz-De-María, F Univ Carlos III Madrid Dept Tecnol Commun Leganes 28911 Spain

The Internet Protocol (IP) environment poses two relevant sources of distortion to the speech recognition problem: lossy speech coding and packet loss. In this paper, we propose a new front-end for speech recognition over IP networks. Specifically;we suggest extracting the recognition feature vectors directly from the encoded speech (i.e., the bit stream) instead of decoding it and subsequently extracting the feature vectors. This approach offers two significant benefits. First, the recognition system is only affected by the quantization distortion of the spectral envelope. Thus, we are avoiding the influence of other sources of distortion due the encoding-decoding process. Second, when packet loss occurs, our front-end becomes more effective since it is not constrained to the error handling mechanism of the codec. Vile have considered the ITU G.723.1 standard codec, which is one of the most preponderant coding algorithms in voice over IP (VoIP) and compared the proposed front-end with the conventional approach in two automatic speech recognition (ASR) tasks, namely, speaker-independent isolated digit recognition and speaker-independent continuous speech recognition. In general, our approach outperforms the conventional procedure, for a variety of simulated packet loss rates. Furthermore, the improvement is higher as network conditions worsen.

关键词： coding distortion G.723.1 IP networks IP telephony packet loss speech recognition voice over IP (VoIP)

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Content-Adaptive Packet-Layer Model for Quality Assessment of Networked Video Services

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IEEE JOURNAL OF SELECTED TOPICS IN SIGNAL PROCESSING 2012年第6期6卷 672-683页

作者： Yang, Fuzheng Song, Jiarun Wan, Shuai Wu, Hong Ren Xidian Univ State Key Lab Integrated Serv Networks Xian 710071 Peoples R China Northwestern Polytech Univ Sch Elect & Informat Xian 710129 Peoples R China RMIT Univ Sch Elect & Comp Engn Melbourne Vic 3001 Australia

Packet-layer models are designed to use only the information provided by packet headers for real-time and non-intrusive quality monitoring of networked video services. This paper proposes a content-adaptive packet-layer (CAPL) model for networked video quality assessment. Considering the fact that the quality degradation of a networked video significantly relies on the temporal as well as the spatial characteristics of the video content, temporal complexity is incorporated in the proposed model. Due to very limited information directly available from packet headers, a simple and adaptive method for frame type detection is adopted in the CAPL model. The temporal complexity is estimated using the ratio of the number of bits for coding P and I frames. The estimated temporal complexity and frame type are incorporated in the CAPL model together with the information about the number of bits and positions of lost packets to obtain the quality estimate for each frame, by evaluating the distortions induced by both compression and packet loss. A two-level temporal pooling is employed to obtain the video quality given the frame quality. Using content related information, the proposed model is able to adapt to different video contents. Experimental results show that the CAPL model significantly outperforms the G. 1070 model and the DT model in terms of widely used performance criteria, including the Root-Mean-Squared Error (RMSE), the Pearson Correlation Coefficient (PCC), the Spearman Rank Order Correlation Coefficient (SCC), and the Outlier Ratio (OR).

关键词： Video quality assessment packet-layer model coding distortion packet loss networked video

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MEASUREMENT OF SCENE-DEPENDENT QUALITY VARIATIONS IN DIGITALLY CODED TELEVISION PICTURES

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IEE PROCEEDINGS-VISION IMAGE AND SIGNAL PROCESSING 1995年第3期142卷 149-154页

作者： ALDRIDGE, R DAVIDOFF, J GHANBARI, M HANDS, D PEARSON, D UNIV ESSEX DEPT PSYCHOLCOLCHESTER CO4 3SQESSEXENGLAND

Experiments are reported in which the ITU-R Double Stimulus Continuous Quality Scale (DSCQS) method was used to assess scene-dependent, time-varying television picture coding distortions introduced by MPEG-2 coding, at bit rates ranging from 1 Mbit/s to 7.5 Mbit/s. Test sequence lengths of both 10 s and 30 s were employed, the latter in an attempt to present to subjects a more representative range of quality variations, which were found to be large. In forming quality judgements of 30 s sequences, subjects gave significantly more weight to impairments occurring towards the end of the sequence;for identical impairments in the first and in the last 10 s of such a sequence, the difference in rating was typically about half a scale point on a 5-point scale. This recency effect is probably linked to the short-term nature of human working memory.

关键词： BIT RATE coding distortion RECENCY EFFECT TELEVISION PICTURES TESTING METHODOLOGY TEST SEQUENCES

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Recognizing GSM digital speech

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 2005年第6期13卷 1186-1205页

作者： Gallardo-Antolín, A Peláez-Moreno, C Díaz-de-María, F Univ Carlos III Madrid Signal Theory & Commun Dept Leganes 28911 Madrid Spain

The Global System for Mobile (GSM) environment encompasses three main problems for automatic speech cognition (ASR) systems: noisy scenarios, source coding distortion, and transmission errors. The first one has already received much attention;however, source coding distortion and transmission errors must be explicitly addressed. In this paper, we propose an alternative front-end for speech recognition over GSM networks. This front-end is specially conceived to be effective against source coding distortion and transmission errors. Specifically, we suggest extracting the recognition feature vectors directly from the encoded speech (i.e., the bitstream) instead of decoding it and subsequently extracting the feature vectors. This approach offers two significant advantages. First, the recognition system is only affected by the quantization distortion of the spectral envelope. Thus, we are avoiding the influence of other sources of distortion as a result of the encoding-decoding process. Second, when transmission errors occur, our front-end becomes more effective since it is not affected by errors in bits allocated to the excitation signal. We have considered the half and the full-rate standard codecs and compared the proposed front-end with the conventional approach in two ASR tasks, namely, speaker-independent isolated digit recognition and speaker-independent continuous speech recognition. In general, our approach outperforms the conventional procedure, for a variety of simulated channel conditions. Furthermore, the disparity increases as the network conditions worsen.

关键词： coding distortion Global System for Mobile (GSM) networks speech coding speech recognition tandeming transmission errors wireless networks

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Continuous quality assessment of digitally-coded television pictures

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IEE PROCEEDINGS-VISION IMAGE AND SIGNAL PROCESSING 1998年第2期145卷 116-123页

作者： Aldridge, RP Hands, DS Pearson, DE Lodge, NK Univ Essex Dept Elect Syst Engn Colchester CO4 3SQ Essex England Univ Essex Dept Psychol Colchester CO4 3SQ Essex England Independent TV Commiss Winchester SO23 7QA Hants England

Experimental tests are reported of the use of the ITU-R single-stimulus continuous quality evaluation (SSCQE) method to measure the quality of long (24 min) sequences of digitally coded constant bit-rate video, with and without audio. The results indicate that the method is reliable and free from drift and fatigue effects. The relationship between continuously-judged quality and overall quality judgements, made at the end of the sequences, is shown to be disproportionately influenced by negative SSCQE peaks and relatively little by their duration. The implication of these results for digital television coding and transmission is considered.

关键词： coding distortion digital video picture quality evaluation

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