Scalable Bit Research Articles

A Wideband Scalable Bit Rate Mixed Excitation Linear Prediction-Enhanced Speech Coder by Preserving Speaker-Specific Features

A Wideband Scalable Bit Rate Mixed Excitation Linear Prediction-Enhanced Speech Coder by Preserving Speaker-Specific Features

Bit Rate Transcoding for High Efficiency Video Coding

High efficiency video coding (HEVC) has demonstrated a notable increase in compression performance and is taken as a successor to H.264/AVC. Efficient bit rate adaptation algorithms are required to contain the HEVC standard between real life community facilities. A present issue of bit rate transcoding is its high computational complexity which is related with the encoder of a cascaded pixel domain transcoder. This paper gives Top to Bottom (T2B) approach to reduce complexity by using different complexity schemes. Proposed approach is effective in reducing complexity in Coding Unit (CU) optimization level. Coding Unit has been analyzed in T2B Approach. While examining the coding unit information of the input video is turned to account for decreasing the number of evaluation and early terminate the process. For the Prediction Unit (PU) level the units are powerfully chosen contingent upon likelihood of Prediction Unit sizes and co-found input prediction partitioning. By utilizing this approach, complexity scalable bit rate transcoding has achieved. Machine learning approach can be used to control computational complexity. Additionally, the T2B strategy is able to gain a spread on trade-offs in transrating complexity and coding performance. Using T2B approach 15% encoding time saving is accomplished. From this scheme, for the less resolution video 27% time saving has achieved.

Efficient Multiple Concatenated Codes With Turbo-Like Decoding for UEP Wireless Transmission of Scalable JPEG 2000 Images

Most multimedia source encoders produce scalable bit streams that are vulnerable to noisy channel effects. Unequal error protection (UEP) designs are attractive solutions for such scalable bit streams. In the other hand, the structure of multiple concatenated codes, which consist of the combination of two or more elementary encoders and interleavers, offers a flexible implementation of unequal data protection. To provide an efficient protection for embedded source bit streams, an adaptive concatenated coding/decoding design is proposed in this paper inspired by serially concatenated turbo codes. This scheme involves iterative decoding of concatenated codes for JPEG 2000 image transmission. In the case of time varying channels, the concatenated coding rates are adaptively chosen to maximize the visual quality. The simulation results show that the proposed scheme achieves better image quality improvements at the receiver side compared with a conventional multiple turbo codes-based design, with significant peak-signal-to-noise ratio gains.

Scalable Coding of Depth Images with Synthesis-Guided Edge Detection

This paper presents a scalable coding method for depth images by considering the quality of synthesized images in virtual views. First, we design a new edge detection algorithm that is based on calculating the depth difference between two neighboring pixels within the depth map. By choosing different thresholds, this algorithm generates a scalable bit stream that puts larger depth differences in front, followed by smaller depth differences. A scalable scheme is also designed for coding depth pixels through a layered sampling structure. At the receiver side, the full-resolution depth image is reconstructed from the received bits by solving a partial-differential-equation (PDE). Experimental results show that the proposed method improves the rate-distortion performance of synthesized images at virtual views and achieves better visual quality.

Scalable Bit Allocation Between Texture and Depth Views for 3-D Video Streaming Over Heterogeneous Networks

In the multiview video plus depth (MVD) coding format, both texture and depth views are jointly compressed to represent the 3-D video content. The MVD format enables synthesis of virtual views through depth-image-based rendering; hence, distortion in the texture and depth views affects the quality of the synthesized virtual views. Bit allocation between texture and depth views has been studied with some promising results. However, to the best of our knowledge, most of the existing bit-allocation methods attempt to allocate a fixed amount of total bit rate between texture and depth views; that is, to select appropriate pair of quantization parameters for texture and depth views to maximize the synthesized view quality subject to a fixed total bit rate. In this paper we propose a scalable bit-allocation scheme, where a single ordering of texture and depth packets is derived and used to obtain optimal bit allocation between texture and depth views for any total target rates. In the proposed scheme, both texture and depth views are encoded using the quality scalable coding method; that is, medium grain scalable (MGS) coding of the Scalable Video Coding (SVC) extension of the Advanced Video Coding (H.264/AVC) standard. For varying target total bit rates, optimal bit truncation points for both texture and depth views can be obtained using the proposed scheme. Moreover, we propose to order the enhancement layer packets of the H.264/SVC MGS encoded depth view according to their contribution to the reduction of the synthesized view distortion. On one hand, this improves the depth view packet ordering when considered the rate-distortion performance of synthesized views, which is demonstrated by the experimental results. On the other hand, the information obtained in this step is used to facilitate optimal bit allocation between texture and depth views. Experimental results demonstrate the effectiveness of the proposed scalable bit-allocation scheme for texture and depth views.

Efficient Prediction Structures for H.264 Multi View Coding Using Temporal Scalability

Prediction structures with disposable view components based hierarchical coding have been proven to be efficient for H.264 multi view coding. Though these prediction structures along with the QP cascading schemes provide superior compression efficiency when compared to the traditional IBBP coding scheme, the scalability requirements of the bit stream could not be met to the fullest. On the other hand, a fully scalable bit stream, obtained by temporal identifier based hierarchical coding, provides a number of advantages including bit rate adaptations and improved error resilience, but lacks in compression efficiency when compared to the former scheme. In this paper it is proposed to combine the two approaches such that a fully scalable bit stream could be realized with minimal reduction in compression efficiency when compared to state-of-the-art disposable view components based hierarchical coding. Simulation results shows that the proposed method enables full scalability with maximum BDPSNR reduction of only 0.34 dB. A novel method also has been proposed for the identification of identifier for the legacy H.264/AVC base layer packets. Simulation results also show that this enables the scenario where the enhancement views could be extracted at a lower frame rate (1/2nd or 1/4th of base view) with average extraction time for a view component of only 0.38 ms.

Joint Source/Channel Coding for Prioritized Wireless Transmission of Multiple 3-D Regions of Interest in 3-D Medical Imaging Data

This paper presents a 3-D medical image coding method featuring two major improvements to previous work on 3-D region of interest (RoI) coding for telemedicine applications. Namely, 1) a data prioritization scheme that allows coding of multiple 3-D-RoIs; and 2) a joint/source channel coding scheme that allows prioritized transmission of multiple 3-D-RoIs over wireless channels. The method, which is based on the 3-D integer wavelet transform and embedded block coding with optimized truncation with 3-D context modeling, generates scalable and error-resilient bit streams with 3-D-RoI decoding capabilities. Coding of multiple 3-D-RoIs is attained by prioritizing the wavelet-transformed data according to a Gaussian mixed distribution, whereas error resiliency is attained by employing the error correction capabilities of rate-compatible punctured turbo codes. The robustness of the proposed method is evaluated for transmission of real 3-D medical images over Rayleigh-fading channels with a priori knowledge of the channel condition. Evaluation results show that the proposed coding method provides a superior performance compared to equal error protection and unequal error protection techniques.

A Novel Error Resilient Scheme for Wavelet-based Image Coding Over Packet Networks

this paper presents a robust transmission strategy for wavelet based scalable bit stream over packet erasure channel. By taking the advantage of the bit plane coding and the multiple description coding, the proposed strategy adopts layered multiple description coding (LMDC) for the embedded wavelet coders to improve the error resistant capability of the important bit planes in the meaning of D(R) function. Then, the post-compression rate-distortion (PCRD) optimization process is used to improve the overall performance. At decoding, a simple error concealment technique is used to compensate the data loss. The proposed method is applied on Set Partitioning in Hierarchical Trees (SPIHT) coded image and the experimental results show graceful degradation for variable packet erasure rate.

Bitrate Scalability for Multi-Pulse Based Code Excited Linear Prediction Speech Coder

Problem statement: The bitrate scalability is an important functionality that is defined in the Moving Picture Expert Group (MPEG-4) requirements. Since the traffic in a communication network varies with time, the capability of varying the bitrate of a speech coder is needed. Approach: This study proposes an approach to vary the bitrate by using a bitrate scalable tool which is attached with the core section of Multi-Pulse based Code Excited Linear Predictive (MP-CELP) coder. The bitrates scalable tool employs multi-stage excitation coding based on an embedded-coding approach. The multi-pulse excitation codebook at each stage is adaptively produced depending on the selected excitation signal at the previous stage. Results: The experimental results show that the speech quality of the proposed coder with the bitrate scalable tools is improved above the speech quality of the conventional coder. Conclusion: From the study, the proposed approach gives a tool to improve the speech compression and can also support a variety of coder bitrates.

Interlayer Bit Allocation for Scalable Video Coding

In this paper, we present a theoretical analysis of the distortion in multilayer coding structures. Specifically, we analyze the prediction structure used to achieve temporal, spatial, and quality scalability of scalable video coding (SVC) and show that the average peak signal-to-noise ratio (PSNR) of SVC is a weighted combination of the bit rates assigned to all the streams. Our analysis utilizes the end user's preference for certain resolutions. We also propose a rate-distortion (R-D) optimization algorithm and compare its performance with that of a state-of-the-art scalable bit allocation algorithm. The reported experiment results demonstrate that the R-D algorithm significantly outperforms the compared approach in terms of the average PSNR.

Analysis of Fundamental Frequency Contour of Coded Speech Based on Multi-Pulse Based Code Excited Linear Prediction Algorithm

Problem statement: In low-bit-rate speech communication, speech coding deteriorates the characteristics of the coded speech significantly. An important feature of the speech is the fundamental frequency contour which determines the pitch information of the speech. It has been known that pitch information is one of the core parameter of the multi-pulse based code excited linear prediction (MPCELP) speech coder. Therefore the study of the deteriorated fundamental frequency contour should be conducted properly. Approach: This study proposes an analysis of the fundamental frequency contour of the coded speech based on MP-CELP speech coder. The comparison of the fundamental frequency contour of the natural speech and that of the coded speech has been performed. The MP-CELP with three levels of bitrate scalability is selected as the core speech coder. The speech material includes a hundred of male speech utterances and a hundred of female speech utterances. Results: The experimental results show that the speech coder causes the deterioration of the fundamental frequency contour empirically. The Root Mean Square Error (RMSE) between the fundamental frequency contour of the natural speech and that of the coded speech for three different bitrates has been conducted. The lower bitrate causes the higher value of RMSE. Conclusion: From the study, it is a proved that the MP-CELP speech coder deteriorates the fundamental frequency contour of the transmitted speech.

Scalable Video Coding Bit Stream Extraction Based on Equivalent MSE Method

The bit stream extraction plays an important role in Scalable Video Coding (SVC) [1]. However, one downside of current video coding methods is to ignore the video contents which is in fact an important factor for video coding efficiency. Therefore, an equivalent MSE method is proposed in this paper to extract substreams in the temporal and spatial enhancement layers. When the Motion Vectors (MVs) are large in one video, a larger frame rate is necessary to maintain the continuity of the object movement which makes no jump in the visual sense. In this sense, substreams extraction in temporal enhancement layer has to be satisfied. On the other hand, if there are some larger high-frequency components in a single frame of the video, that is to say, there are some higher spatial details in the video stream. As a result, it should try to meet the extraction requirement in spatial enhancement layer. This method has the advantage of considering the contents of the video, which can effectively improve the coding performance and quality. The experimental results have demonstrated the improved quality of reconstructed video for the equivalent MSE method when extracting bit stream arbitrarily at the same bandwidth.

Tonal Language Speech Compression Based on a Bitrate Scalable Multi-Pulse Based Code Excited Linear Prediction Coder

Problem statement: Speech compression is an important issue in the modern digital speech communication. The functionality of bitrates scalability also plays significant role, since the capacity of communication system varies all the time. When considering tonal speech, such as Thai, tone plays important role on the naturalness and the intelligibility of the speech, it must be treated appropriately. Therefore these issues are taken into account in this study. Approach: This study proposes a modification of flexible Multi-Pulse based Code Excited Linear Predictive (MP-CELP) coder with bitrates scalabilities for tonal language speech in the multimedia applications. The coder consists of a core coder and bitrates scalable tools. The high pitch delay resolutions are applied to the adaptive codebook of core coder for tonal language speech quality improvement. The bitrates scalable tool employs multi-stage excitation coding based on an embedded-coding approach. The multi-pulse excitation codebook at each stage is adaptively produced depending on the selected excitation signal at the previous stage. Results: The experimental results show that the speech quality of the proposed coder is improved above the speech quality of the conventional coder without pitch-resolution adaptation. Conclusion: From the study, the proposed approach is able to improve the speech compression quality for tonal language and the functionality of bitrates scalability is also developed.

High Pitch Delay Resolution Technique for Tonal Language Speech Coding Based on Multi-Pulse Based Code Excited Linear Prediction Algorithm

Problem statement: In spontaneous speech communication, speech coding is an important process that should be taken into account, since the quality of coded speech depends on the efficiency of the speech coding algorithm. As for tonal language which tone plays important role not only on the naturalness and also the intelligibility of the speech, tone must be treated appropriately. Approach: This study proposes a modification of flexible Multi-Pulse based Code Excited Linear Predictive (MP-CELP) coder with multiple bitrates and bitrate scalabilities for tonal language speech in the multimedia applications. The coder consists of a core coder and bitrate scalable tools. The High Pitch Delay Resolutions (HPDR) are applied to the adaptive codebook of core coder for tonal language speech quality improvement. The bitrate scalable tool employs multi-stage excitation coding based on an embedded-coding approach. The multi-pulse excitation codebook at each stage is adaptively produced depending on the selected excitation signal at the previous stage. Results: The experimental results show that the speech quality of the proposed coder is improved above the speech quality of the conventional coder without pitch-resolution adaptation. Conclusion: From the study, it is a strong evidence to further apply the proposed technique in the speech coding systems or other speech processing technologies.

Performance Evaluation of Multi-Pulse Based Code Excited Linear Predictive Speech Coder with Bitrate Scalable Tool over Additive White Gaussian Noise and Rayleigh Fading Channels

Problem statement: In the modern speech communication technology, the speech coding with bitrate scalability was needed. However, various types of noises in the communication channels cause damages in the transmitted information especially speech data. Tonal-language speech was also affected by this situation. Approach: Based on the high pitch delay resolution technique, the MPCELP speech coding was proposed over the environment with CDMA AWGN and Rayleigh fading channels for tonal language. The proposed coder supports multiple bitrates and also has the functionality of bitrate scalability. Results: Through performance analysis and computer simulation, the quality of the proposed coding was presented with an improvement from conventional scalable MP-CELP in the specific-noise environments. The HPDR technique was applied to the MP-CELP to use for tonal language, meanwhile it can support the core coding rate of 5.6, 8.2, 12.2 kbps and additional scaled bitrates. Conclusion: By applying the high pitch delay resolution technique with the MP-CELP speech coding, we can improve the quality of tonal encoded speech. Moreover, the coding quality of the proposed coder was better than the conventional coder for Thai language over both AWGN channel and Rayleigh fading channel.

All-optical Label Swapping Techniques for Optical Packets at Bit-rate Beyond 160 Gb/s

In this paper t wo different paradigms to realize a scalable all-optical packet switch with label swapping will be presented. All the functions required for switching the packets are based on all-optical signal processing without any electronic control. This allows very low latency and potential photonic integration of the systems. We report for both techniques experimental results showing the routing operation of the 160 Gb/s packets and beyond. W e will discuss and compare both techniques in term of devices and bit-rate scalability, latency, power consumption, power penalty performance and cascadability as key parameters for the realization of an all-optical packet switch.

All-Optical Packet Switching and Label Rewriting for Data Packets Beyond 160 Gb/s

In this paper, two different paradigms to realize a scalable all-optical packet switch with label swapping will be reviewed. The two paradigms are based on wavelength routing switch and space routing switch. All the functions required for switching the packets, namely, the label processor, the label rewriter, and the optical switch, are based on all-optical signal processing with no electronic control. This allows for very fast processing time and potential photonic integration of the systems. We report, for both techniques, experimental results showing the routing operation of the 160-Gb/s packets and beyond. We will discuss and compare both techniques in terms of devices and bit-rate scalability, power consumption, power penalty performance, and cascadability as key parameters for the realization of an all-optical packet switch.

Adaptive Signal Modeling Based on Sparse Approximations for Scalable Parametric Audio Coding

<para xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> This paper deals with the application of adaptive signal models for parametric audio coding. A fully parametric audio coder, which decomposes the audio signal into sinusoids, transients and noise, is here proposed. Adaptive signal models for sinusoidal, transient, and noise modeling are therefore included in the parametric scheme in order to achieve high-quality and low bit-rate audio coding. In this paper, a new sinusoidal modeling method based on a perceptual distortion measure is proposed. For transient modeling, a fast and effective method based on matching pursuit with a mixed dictionary is chosen. The residue of the previous models is analyzed as a noise-like signal. The proposed parametric audio coder allows high quality audio coding for one-channel audio signals at 16 kbits/s (average bit rate). A bit-rate scalable version of the parametric audio coder is also proposed in this work. Bit-rate scalability is intended for audio streaming applications, which are highly demanded nowadays. The performance of the proposed parametric audio coders (nonscalable and scalable coders) is assessed in comparison to widely used audio coders operating at similar bit rates. </para>

Perceptual Optimization for Scalable Video Compression Based on Visual Masking Principles

This paper describes a visual optimization strategy for scalable video compression. The challenge scalable coding presents is that truncation of an embedded codestream may induce variable and highly visible distortion. To overcome the deficiencies of visually lossless coding schemes, we propose using an adaptive masking slope to model the perceptual impact of suprathreshold distortion arising from resolution and bit-rate scaling. This allows important scene structures to be better preserved. Following visual masking principles, local sensitivity to distortion is assessed within each frame. To keep the perceptual response uniform against spatiotemporal errors, we mitigate errors compounded by the motion field during temporal synthesis. Visual sensitivity weights are projected into the subband domain along motion trajectories via a process called perceptual mapping. This uses error propagation paths to capture some of the noise-shaping effects attributed to the motion-compensated transform. A key observation is that low contrast regions in the video are generally more susceptible to unmasking of quantization errors. The proposed approach raises the distortion-length slope associated with these critical regions, altering the bitstream embedding order so that visually sensitive sites may be encoded with higher fidelity. Subjective evaluation demonstrates perceptual improvement with respect to bit-rate, spatial and temporal scalability.

Scalable Audio Compression at Low Bitrates

A perceptually scalable audio coder generates a bit-stream that contains layers of audio fidelity and is encoded in such a way that adding one of these layers enhances the reconstructed audio by an amount that is just noticeable by the listener. Such algorithms have applications like music on demand at variable levels of fidelity, for instance using 3G and 4G cellular radio systems operating at different bit rates. While the MPEG-4 natural audio coder can create finely scalable bit streams using bit sliced arithmetic coding (BSAC), its perceptual quality at low bit rates is poor. On the other hand, the nonscalable transform-domain weighted interleaved vector quantization (TWIN-VQ) performs well at low bit rates. In this paper, we present a modified version of TWIN-VQ algorithm that generates a perceptually scalable bit-stream with many fine layers of audio fidelity. Using TWIN-VQ as our base ensures the best possible perceptual quality at low bit rates. Specifically, the proposed scalable algorithm performs as well as TWIN-VQ at rates of 8 to 16 kb/s and outperforms scalable BSAC by between 64% and 172% at rates of less than 24 kb/s.