LSP-based multiple-description coding for real-time low bit-rate voice over IP

Postfilters for suppressing coding noises in low bit rate speech coders usually operate in the time domain with high complexity and also introduce nonlinear phase distortion to the speech signals because they are recursive filters. In this paper, a frequency-domain postfiltering technique for multiband excited linear predictive (MBELP) coders is proposed. The idea is to firstly sample the postfilter magnitude spectrum at the pitch harmonics and then scale the signal band magnitudes by the sampled postfilter magnitudes. This approach does not need to specifically perform the filtering process, and the signal energies before and after postfiltering can be easily equalized by simply re-scaling the signal band magnitudes. Another obvious advantage of using this frequency-domain postfiltering technique is that the phase information utilized in multiband synthesis will be retained because only the magnitude spectra are modified. In this paper, a fast method is given to sample the postfilter magnitude spectrum which is characterized by line spectral pair (LSP) parameters. The proposed frequency-domain postfilter has a much lower implementation complexity than its time-domain counterpart.

In view of the limited bandwidth of underwater video image transmission, a low bit rate underwater video compression coding method is proposed. Based on the preprocessing process of wavelet transform and coefficient down-sampling, the visual redundancy of underwater image is removed and the computational coefficients and coding bits are reduced. At the same time, combined with multi-level wavelet decomposition, inter frame motion compensation, entropy coding and other methods, according to the characteristics of different types of frame image data, reduce the number of calculations and improve the coding efficiency. The experimental results show that the reconstructed image quality can meet the visual requirements, and the average compression ratio of underwater video can meet the requirements of underwater acoustic channel transmission rate.

With the emergence of medical imaging, the compression of volumetric medical images is essential. For this purpose, we propose a novel Multiplicative Multiresolution Decomposition (MMD) wavelet coding scheme for lossless compression of volumetric medical images. The MMD is used in speckle reduction technique but offers some proprieties which can be exploited in compression. Thus, as the wavelet transform the MMD provides a hierarchical representation and offers a possibility to realize lossless compression. We integrate in proposed scheme an inter slice filter based on wavelet transform and motion compensation to reduce data energy efficiently. We compare lossless results of classical wavelet coders such as 3D SPIHT and JP3D to the proposed scheme. This scheme incorporates MMD in lossless compression technique by applying MMD/wavelet or MMD transform to each slice, after inter slice filter is employed and the resulting sub-bands are coded by the 3D zero-tree algorithm SPIHT. Lossless experimental results show that the proposed scheme with the MMD can achieve lowest bit rates compared to 3D SPIHT and JP3D.

An optical packet switch design using optical switching and shared electronic buffering is described and analyzed. The electronic buffering and add/drop unit allows random memory access, variable packet length, and aggregation/segregation of low bit rate streams. The design is scalable to a large number of wavelengths, and employs contention resolution by using the wavelength dimension combined with electronic buffering. The number of buffer inputs, i.e. the number of optoelectronic conversions, is minimized. The packet loss and delay performance when using fixed packet length and electronic buffering are investigated by simulation. If the link utilization is moderate, the buffering requirements becomes minimal. If number of wavelengths in the link is increased, the number of buffer inputs can be reduced. With a sufficiently high number of wavelengths, the buffer requirements is minimal, even when the link utilization is high.

This paper proposes a novel error-resilient packet video coding method which is designed to operate on unreliable IP networks suffering from large bursts of packet losses. The main novelties of the proposed system are: (1) A Bayesian packet loss predictor is employed to dynamically predict the packet loss probabilities; (2) A harmonic frame-expansion is applied to the video stream to add error-resilience. This includes dynamically choosing the size of the expansions. (3) Error-propagation is reduced by filtering out error-prone high-frequency components from reference images. This means that the proposed video codec method does not need to switch between the inter and intra coding modes. (4) Theoretical analysis of the video performance is given. Our preliminary simulations have shown that the frame expansion method can provide excellent resilience to video packet losses. The system has generated high quality videos at low bit-rates (64kbs) on simulated networks with the packet loss probability ranging from 5% to 20%. Visual inspection has shown that under similar PSNRs, the resulting videos have noticeable less visual artifacts as compared with those from the conventional codecs

Modern mobile communications require optimum bandwidth utilization with minimum loss, delay and good quality of speech transmission. This triggered the usage of low bit rate voice codecs, among which CELP codecs inherit the merits of both waveform and source codecs, like toll quality, low bit rate etc. In this paper several hardware and software optimization techniques are presented for efficient implementation of ITU G.729 standard (CS-ACELP, conjugate structure algebraic code excited linear prediction) of 8 Kbit/s bit rate on a real time digital signal processor (DSP), with the aim of overcoming the limitation of computational burden and also scaling this application for enhanced speed to process more channels. These techniques are in general applicable to any speech codec and DSP processor platform.

With the development of Internet computing techniques, continuous data streams from remote sites are commonly used in scientific and commercial applications. Correspondingly, there is increasing demand of assuring the integrity and authenticity of received data streams. Existing strategies of assuring data integrity and authenticity mainly use message authentication codes (MAC) generated on data blocks and transfer the MAC to the receiver for authentication through either out of band communication or in band communication. Transferring the MAC via out of band communication inevitably introduces communication overhead and additional complexity to synchronize the out of band communication with the data communication. Transferring the MAC via in band channel can be achieved by either appending the MAC to the original data or embedding the MAC into the original data, which would either incur communication overhead or change the original data. It would be desirable to be able to authenticate the stream data without any communication overhead and changing the original data at the same time. To deal with data packet or block loss, many of existing stream data authentication schemes rely on hash chaining, the current usage of which results in uncertainty in authenticating the subsequent data blocks once the first data packet or block loss is detected. In this paper, we propose a novel application layer authentication strategy called DaTA. This authentication scheme requires no change to the original data and causes no additional communication overhead. In addition, it can continue authenticating the rest of data stream even if some data loss has been detected. Our analysis shows that our authentication scheme is robust against packet loss and network jitter. We have implemented a prototype system to evaluate its performance. Our empirical results show that our proposed scheme is efficient and practical under various network conditions

An overview is presented of compression and packet loss effects in speech biometrics. These new problems appear particularly in recent applications of biometrics over mobile or Internet networks. The influence of speech compression on speaker recognition performance in mobile networks is investigated. In a first experiment, it is found that the use of GSM coding degrades the performance. In a second experiment, the features for the speaker recognition system are calculated directly from the information available in the encoded bit stream. It is found that a low LPC order in GSM coding is responsible for most performance degradations. A speaker recognition system was obtained which is equivalent in performance to the original one which decodes and reanalyses speech before performing recognition. The joint packet loss and compression effects over IP networks are also studied. It is experimentally demonstrated that the adverse effects of packet loss alone are negligible, while the encoding of speech, particularly at a low bit rate, coupled with packet loss, can reduce the verification accuracy considerably.

A case study in identifying acceptable bitrates for human face recognition tasks

This paper presents a hierarchical morphological segmentation algorithm based on PSNR for each region. The algorithm follows a top-down procedure. It first takes into account the global information and produces a coarse segmentation, that is, with a small number of regions. Then, each region has different quality in coarse segmented image. Thus, the region with low quality is selected by PSNR and improved by introducing regions corresponding to more local information. It is able to improve the subjective quality of the image while maintaining the bit rates and control the bit rates for certain applications such as mobile audiovisual communications and videotelephone through PSTN, of which the available transmission bandwidth is very limited.

This paper proposes an algorithm to increase the performance of high frequency (HF) wireless networks. The solution and results presented in this paper will contribute to achieve a reliable communication system to support real time video packet communications, encoded at about 8 kbps using the H.264 video coding standard. In addition to allowing civil/amateur communications, HF bands are also used for long distance wireless military communications. Therefore, our work is based on NATO Link and Physical layer standards, STANAG 5066 and STANAG 4539, respectively. At each HF channel, a typical transmission bandwidth is about 3 kHz centered at a carrier around 20 MHz with the resulting throughput of up to 12800 bps. This very low bit rate imposes itself serious challenges for reliable and low delay real time multimedia communications. Thus, this paper discusses and presents a solution to increase the performance of real time HF communication systems using channel diversity with a novel and simple combination technique, which is an alternative to soft combining and maximum ratio combining. With our HF channel diversity scheme, the packet loss, on average, is decreased by 15% in comparison to a single channel.

Multi-view video plus depth is emerging as the most flexible format for 3D video representation, as witnessed by the current standardization efforts by ISO and ITU. The depth information allows synthesizing virtual view points, and for its compression various techniques have been proposed. It is generally recognized that a high quality view rendering at the receiver side is possible only by preserving the contour information since distortions on edges during the encoding step would cause a sensible degradation on the synthesized view and on the 3D perception. As a consequence recent approaches include contour-based coding of depths. However, the impact of contour-preserving depth-coding on the perceived quality of synthesized images has not been conveniently studied. Therefore in this paper we make an investigation by means of a subjective study to better understand the limits and the potentialities of the different techniques. Our results show that the contour information is indeed relevant in the synthesis step: preserving the contours and coding coarsely the rest typically leads to images that users cannot tell apart from the reference ones, even at low bit rate. Moreover, our results show that objective metrics that are commonly used to evaluate synthesized images may have a low correlation coefficient with MOS rates and are in general not consistent across several techniques and contents.

The authors try to represent a image with triangular patches. The triangular patch is defined as a triangle with linearly varying gray-level within it. To represent an image with triangular patches effectively for the human visual system, and to describe these triangular patches with small amount of data, the authors have tried to get a new style image coding method at very low bit rate. At first, a modified Delaunay triangulation algorithm is applied to the feature points extracted on the boundary lines in the image. Then linear interpolation is done within every triangular region using the gray-value at these feature points. Thus the approximately represented image is described with triangular patches. Using this image description method, an image coding method with progressive build-up is proposed. By the simulation results and subjective tests, it is confirmed that this coding scheme using triangular patch structure is effective at super-low bit rate. >

With the rapid growth of Internet video image information, there is a large amount of redundancy in image data. Use less data stream information to transfer the image or the amount of information contained in the image. Its purpose is to reduce the redundancy of images, so as to store them at low bit rate and reduce the data storage space. In the general image compression method, the hybrid coding framework is adopted. Each algorithm adopts a fixed algorithm mode through a specific design algorithm, without global optimization. Image compression is mainly divided into prediction, transformation, quantization, digital entropy coding and other steps. At present, there are many researches on super-resolution network based on deep learning technology. The main function is to reconstruct high-resolution image replace image magnification low-resolution images such as linear interpolation, which has a great performance improvement image resolution, noise reduction, deblurring and so on, but there is no effective way to use super-resolution network applications to improve quality of compression reconstructed image quality. This paper involves a new method that using image super-resolution residual learning network to improve quality of compression image, our method, the reduced image is encoded into a content stream and a transmission corresponding parameter is encoded into a model stream. Firstly, the original image is scaled down 1/2 size of source image, then encode the small image into content stream with the existing codec. Secondly, the residual learning super-resolution (SR) network is used for image filtering to scale up reconstructed image with decode image resizing method and boost the quality of edge feature extraction of image. Our results show that there is significant performance improvement of h265 in low resolution reconstructed image (bits-per-pixel less than 0.1).

In IPTV streaming, issues of entire frame losses limit the use of state-of-the-art video quality analyses techniques. We present an experimental evaluation of degradation in video quality when streaming different types of video in an IPTV scenario. The PacketStorm network emulator is used to emulate real IPTV networks as detailed in the ITU-T G.1050 recommendation. Our results show that in the face of equal network impairments, videos with different characteristics showed different levels of resilience in terms of net frame losses. Particularly, low bit rate videos showed more resilience to frame drops as compared to high bit rate videos. Our analysis considers different types of content (with variable spatial and temporal complexity) and encoding schemes. We found that MPEG videos perform better than QuickTime videos in all the IPTV network severity levels recommended by ITU-T.