Very low rate DCT-based video coding using dynamic VQ

A new feedforward control method to control encod-ing parameters is proposed for low bit rate video encoding. At low transmission bit rates, all input video frames cannot be transmitted and an encoding frame interval varies with the information amount in each encoded frame. Subjective video quality is evaluated from both picture degradation and encoding frame intervals. Because picture degradation and encoding frame intervals are traded off, encoding parameter control is very important. The proposed method is based on a hybrid transform predictive coding scheme with adaptive coefficient selection. The method estimates the amount of information for some encoding parameters and selects parameters so that given control characteristics can be achieved. Computer simulations for both the proposed method and a conventional feedback method show the effectiveness of the new method. This method will be applied to an NETEC series low bit rate video codec.

There are many high performance speech coding algorithms, but most of them using bit rates higher than 5kbps. Expected quality of low bit rate speech coding has not been achieved yet, which is an area of interest for military and security applications. There is still room for improvement for low bit rates. Low bit rate speech coding using sinusoidal transform coding is discussed in this paper. The sinusoidal transform coding (STC) is a frequency domain speech compression technique, which can produce speech of high quality at bit rates below 5kbps.In STC, speech signal are represented by linear combination of sine waves with time-vary amplitudes, phase, and frequencies. STC can produce high quality synthesized speech at low bit rates. This paper focuses on the development of a fully speech coder at 2.4 kbps and 4.8 kbps.

Motion estimation and motion compensation (MEMC) has been applied inside the wavelet-transformed domain for video compression at low bit rates. However, pixel-based motion estimation in the wavelet domain is not as straightforward as in the spatial domain due to the shift variance property caused by decimation. The paper finds a new wavelet-based motion estimation and compensation method, named as the sub-decimation decomposition method, to perform shift invariant motion estimation inside the wavelet transformed domain, so that better matching blocks are compensated to achieve more efficient video compression.

Low bit rate audio coding often relies on Fourier representation despite its limitations for transient signal modeling. This study proposes alternative decompositions and expansion strategies that lead to more accurate modeling. Two classes of methods are considered, subspace decomposition methods, and atomic decomposition methods and their performances are compiled to propose an audio modeling scheme amenable to low bit rate coding.

The development of low bit rate terminals for use with wide bandwidth Ka band transponders, such as ACTS, has forced a reconsideration of phase noise and how it effects low bit rate systems. In this paper we initially calculate the cumulative effect of the phase noise contributions from the oscillators in the ground based transmitter, satellite transponder, and the ground based receiver. We will discuss the cases where the carrier power-to-phase noise power ratio is much larger and much smaller than the carrier power-to-noise/interference power ratio. We extend our results to the case of non-white phase noise spectrum and compare with actual oscillator spectra. We also will discuss complications arising from the fact that one cannot change the phase noise characteristics of the spacecraft. We conclude by illustrating how these analyses can be used to predict the increase in error probability due to phase noise.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

The influence of noise on the performance of a low bit rate parametric speech coder is addressed in this paper. MELP vocoder is used to estimate three parameters: the fundamental frequency, voicing and linear prediction coefficients. Influence of different noises under various acoustic environments on the MELP vocoder's parameters is studied. Pitch accuracy rate, voicing decision error rate and average spectral distortion are used to quantitatively evaluate the corrupted speech quality and intelligibility. The results show that the pitch accuracy and voicing decision are susceptible to the noise even at high SNR and illustrate that the presence of noise in unvoiced frames has great impact on the accuracy of voicing decision. The research will be helpful in designing specific speech enhancement algorithm for low bit rate coding.

Current rate control schemes in video coding standards do not have efficient frame-level bit allocation because of the inherent constraints in real-time encoding. In this paper, we assume an offline video encoding environment and proposed a rate control scheme based on optimal bit allocation for low bit rate streaming applications. Specifically, we apply a /spl rho/-domain rate-distortion (R-D) model, originally applied at macroblock (MB) level, to frame-level. Based on this frame-level R-D model and a two-pass encoding method, we are able to allocate bits among video frames in an optimal way so that video sequences can be coded at low bit rate with an improved quality. Experimental results demonstrate the proposed scheme is able to achieve not only noticeable reduction in average distortion but also a more consistent and smoother visual quality.

We utilize the inter-frame redundancy with the larger-size super-frame structure to realize ultra low bit rate speech encoding. A new clustering model of speech characteristics is proposed to process effectively the parameters of large super-frames. Based on the model, we present algorithms for ultra low bit rate speech encoding at 600 bps and 300 bps for applications in acoustically harsh environments. At the decoder, a close-loop excitation signal magnitude estimation model is employed to improve the naturalness of synthesized speech. Two prototypes have been realized and evaluated using the DRT tests based on the national standard of China. Both prototypes are able to synthesize high quality of speech with DRT score 88.85 and 81.78 respectively.

We present an efficient computation constrained block-based motion vector estimation algorithm for low bit rate video coding that offers good tradeoffs between motion estimation distortion and number of computations. A reliable predictor determines the search origin. An efficient search pattern exploits structural constraints within the motion field. A flexible cost measure used to terminate the search allows simultaneous control of the motion estimation distortion and the computational cost. Experimental results demonstrate the viability of the proposed algorithm in low bit rate video coding applications, achieving essentially the same levels of rate-distortion performance and subjective quality as that of the full search algorithm when used by the UBC H.263+ video coding reference software. However the proposed motion estimation algorithm provides substantially higher encoding speed as well as graceful computational degradation capabilities.

We propose a new real-time rate-distortion (R-D) optimized dynamic frame-skipping (DFS) transcoder for low bit rate video transmission. The proposed transcoder can adjust the number of skipped frames according to the target frame rate as well as the channel rate. Experimental results indicate that the proposed R-D optimized transcoder can not only perform frame-skipping effectively but also improve visual quality in the low bit rate video transcoding.

Forecasts predict that Internet traffic will continue to grow in the near future. A huge share of this traffic is caused by multimedia streaming. The quality of experience (QoE) of such streaming services is an important aspect and in most cases the goal is to maximize the bit rate which-in some cases-conflicts with the requirements of both consumers and providers. For example, in mobile environments users may prefer a lower bit rate to come along with their data plan. Likewise, providers aim at minimizing bandwidth usage in order to reduce costs by transmitting less data to users while maintaining a high QoE. Today's adaptive video streaming services try to serve users with the highest bit rates that consequently results in high QoE. In practice, however, some of these high bit rate representations may not differ significantly in terms of perceived video quality compared to lower bit rate representations. In this paper, we present a novel approach to determine the statistically indifferent quality variation of adjacent video representations for adaptive video streaming services by adopting standard objective quality metrics and existing QoE models. In particular, whenever the quality variation between adjacent representations is imperceptible from a statistical point of view, the representation with higher bit rate can be substituted with a lower bit rate representation. As expected, this approach results in savings with respect to bandwidth consumption while still providing a high QoE for users. The approach is evaluated subjectively with a crowdsourcing study. Additionally, we highlight the benefits of our approach, by providing a case study that extrapolates possible savings for providers.

Recently, ITU-T established a draft standard, H.263 to he used as a possible standard for very low bit rate video coding at less than 64 kb/s. It is based on the ITU-T H.261 standard for videoconferencing. This paper provides an analysis of the performance of H.263 coder with the aim of establishing its operational parameters and limitations when operating at very low bit rates. Its performance is then optimized by selecting the desired set of operational parameters. The H.263 coder suffers from "blocking" effects when operating at very low bit rates. To improve the subjective quality of the coded images, the discrete cosine transform (DCT) coefficients are weighted using a visual quantization matrix before quantization. Simulation results showed that the proposed method increases the peak signal-to-noise ratio (PSNR) values at all bit rates tested, but more importantly, the perceived visual quality is improved significantly by reducing the annoying "blocking" effects.

Scalable Video Coding (SVC) now is becoming one extension part of the H.264/AVC standard. The bandwidth of video transmission systems is different, and the terminal units of video communication are diverse, so the SVC will be more popular because of its scalability in coding different picture size, frame rate and bit rate. However, the flexibility of SVC can’t solve the conflict between the bit rate and the video quality, especially in the very low bit rate condition. In this paper, we describe a new approach to deal with this problem. We downsample the layer with high resolution, and then code the different layers with SVC. Finally, we upsample the high layer data in the decoder. Experiments show that this method can achieve better subjective quality in the low bit rate condition.

Many lossy compression algorithms have been proposed that perform well for low bit rates, but are computationally intense. On the other hand, there are many simple algorithms that perform poorly at high compression ratios. An algorithm that incorporates both computational simplicity and acceptable performance at low bit rates would have many potential applications. In wireless communications, for example, where computing power and transmission speed are important factors, the time required to compress and send image data would need to be reduced significantly. A low bit rate compression algorithm that is conceptually simple as well as computationally efficient is presented in this paper. Theoretically, it can be shown to be computationally superior to JPEG in the compression phase, and comparable in the decompression phase. In practice, the quality of the results, as measured by the PSNR, is found to be better than JPEG results at bit rates lower than about 0.2-0.3 bpp depending on the input image.

In this paper, we introduce a new implementation of motion-compensated temporal filter (MCTF) based video coder which utilizes rate-distortion optimal wavelet packet transform (RDWP) for some low bit rate video coding application scenarios. Although enhanced MC-EZBC video coder provides good scalability and high coding efficiency at high bit rates, reconstructed video quality will drop rapidly with increased motion at low bit rates. The proposed video coder applies optimal wavelet packet transform to each temporal frame and rate-distortion optimization to each GOP. Experimental results show this codec provides better performance than MC-EZBC with PSNR improved 0.3–1.2dB at low bit rates.