Source Coding and Compression

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Full List of Titles
1: Speech Processing
CELP Coding
Large Vocabulary Recognition
Speech Analysis and Enhancement
Acoustic Modeling I
ASR Systems and Applications
Topics in Speech Coding
Speech Analysis
Low Bit Rate Speech Coding I
Robust Speech Recognition in Noisy Environments
Speaker Recognition
Acoustic Modeling II
Speech Production and Synthesis
Feature Extraction
Robust Speech Recognition and Adaptation
Low Bit Rate Speech Coding II
Speech Understanding
Language Modeling I
2: Speech Processing, Audio and Electroacoustics, and Neural Networks
Acoustic Modeling III
Lexical Issues/Search
Speech Understanding and Systems
Speech Analysis and Quantization
Utterance Verification/Acoustic Modeling
Language Modeling II
Adaptation /Normalization
Speech Enhancement
Topics in Speaker and Language Recognition
Echo Cancellation and Noise Control
Coding
Auditory Modeling, Hearing Aids and Applications of Signal Processing to Audio and Acoustics
Spatial Audio
Music Applications
Application - Pattern Recognition & Speech Processing
Theory & Neural Architecture
Signal Separation
Application - Image & Nonlinear Signal Processing
3: Signal Processing Theory & Methods I
Filter Design and Structures
Detection
Wavelets
Adaptive Filtering: Applications and Implementation
Nonlinear Signals and Systems
Time/Frequency and Time/Scale Analysis
Signal Modeling and Representation
Filterbank and Wavelet Applications
Source and Signal Separation
Filterbanks
Emerging Applications and Fast Algorithms
Frequency and Phase Estimation
Spectral Analysis and Higher Order Statistics
Signal Reconstruction
Adaptive Filter Analysis
Transforms and Statistical Estimation
Markov and Bayesian Estimation and Classification
4: Signal Processing Theory & Methods II, Design and Implementation of Signal Processing Systems, Special Sessions, and Industry Technology Tracks
System Identification, Equalization, and Noise Suppression
Parameter Estimation
Adaptive Filters: Algorithms and Performance
DSP Development Tools
VLSI Building Blocks
DSP Architectures
DSP System Design
Education
Recent Advances in Sampling Theory and Applications
Steganography: Information Embedding, Digital Watermarking, and Data Hiding
Speech Under Stress
Physics-Based Signal Processing
DSP Chips, Architectures and Implementations
DSP Tools and Rapid Prototyping
Communication Technologies
Image and Video Technologies
Automotive Applications / Industrial Signal Processing
Speech and Audio Technologies
Defense and Security Applications
Biomedical Applications
Voice and Media Processing
Adaptive Interference Cancellation
5: Communications, Sensor Array and Multichannel
Source Coding and Compression
Compression and Modulation
Channel Estimation and Equalization
Blind Multiuser Communications
Signal Processing for Communications I
CDMA and Space-Time Processing
Time-Varying Channels and Self-Recovering Receivers
Signal Processing for Communications II
Blind CDMA and Multi-Channel Equalization
Multicarrier Communications
Detection, Classification, Localization, and Tracking
Radar and Sonar Signal Processing
Array Processing: Direction Finding
Array Processing Applications I
Blind Identification, Separation, and Equalization
Antenna Arrays for Communications
Array Processing Applications II
6: Multimedia Signal Processing, Image and Multidimensional Signal Processing, Digital Signal Processing Education
Multimedia Analysis and Retrieval
Audio and Video Processing for Multimedia Applications
Advanced Techniques in Multimedia
Video Compression and Processing
Image Coding
Transform Techniques
Restoration and Estimation
Image Analysis
Object Identification and Tracking
Motion Estimation
Medical Imaging
Image and Multidimensional Signal Processing Applications I
Segmentation
Image and Multidimensional Signal Processing Applications II
Facial Recognition and Analysis
Digital Signal Processing Education

Author Index
A B C D E F G H I
J K L M N O P Q R
S T U V W X Y Z

Optimal Bit Allocation with Side Information

Authors:

Paolo Prandoni, LCAV, Ecole Polytechnique Federale de Lausanne (USA)
Martin Vetterli, EECS Dept. UC Berkeley, USA (USA)

Page (NA) Paper number 2314

Abstract:

For a given set of quantizers and a data vector, the optimal bit allocation in a rate/distortion sense is the sequence of quantizers which minimizes the overall distortion for a given bit budget. In an operational framework, this sequence is dependent on the data realization rather than on its probabilistic model and the cost of describing the sequence itself becomes therefore part of the bit budget. We present an allocation algorithm based on dynamic programming which determines the optimal bit allocation taking into account the side information of describing the structure of the allocation itself; practical simplifications of the algorithm are also presented with respect to coding of continuous data sources.

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Performance of Multiple Description Coders on a Real Channel

Authors:

Amy R Reibman,
Hamid Jafarkhani,
Michael T Orchard,
Yao Wang,

Page (NA) Paper number 1744

Abstract:

In this paper we explore the ability of multiple description (MD) source coders to achieve good performance on channels other than ideal MD channels. We examine both the overall system design and compare the performance of a system with MD source coder to that of a more traditional system using a layered source coder. For the memoryless channels we consider, MD source coding cannot achieve acceptable performance for a Gaussian memoryless source without appropriate channel coding. Also, in memoryless channels, a system with MD source coding outperforms a layered source coding system only in very poor channels. The introduction of memory in the channel degrades the performance of both systems equally. Using interleaving to reduce the impact of memory in the channel has more influence on performance than the choice of source coder.

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Multiple Description Coding via Scaling Rotation Transform

Authors:

Wenqing Jiang,
Antonio Ortega,

Page (NA) Paper number 2475

Abstract:

In this paper, we propose a two-stage transform design technique for Multiple Description Transform Coding. The first stage is the structure design in which we enforce a Scaling-Rotation factorization of the transform and we further constrain the transform for specific channel conditions using the knowledge of the input correlation matrix and the desired output correlation matrix. In the second stage, magnitude design, we find the optimal transform from all admissible transforms given by the structure design using the numerical algorithm proposed by Goyal et al. (citation deletec). Such a design enables a structured transform framework which reduces both the design and implementation complexities compared to an exhaustive search through the whole space of nonorthogonal transforms. We give two examples to illustrate the design idea, the Scaling-Hadamard transform for equal rate channels and the Scaling-DST transform for sequential protection channels.

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Unequal Error Protection Methods for Perceptual Audio Coders

Authors:

Deepen Sinha,
Carl-Erik W Sundberg,

Page (NA) Paper number 2364

Abstract:

In most source coded bit streams certain bits can be much more sensitive to transmission errors than others. Unequal error protection (UEP) offers a mechanism for matching error protection capability to sensitivity to transmission errors. A UEP system typically has the same average transmission rate as a corresponding equal error protection (EEP) system but offers an improved perceived signal quality at equal channel signal to noise ratio. In this work we introduce methods of UEP to the perceptual audio coder (PAC). An error sensitivity classifier divides the bits in classes of different sensitivity. Different channel codes are then applied to each class. We show how punctured convolutional codes can be used for UEP of the PAC bitstream. Experimental results for channels with uniform as well as non-uniform noise/interference level indicate that the systems with UEP exhibit graceful degradation and extended range for applications auch as digital audio broadcasting (DAB).

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Embedded Joint Source-Channel Coding Of Speech Using Symbol Puncturing Of Trellis Codes

Authors:

Alexis P Bernard,
Xueting Liu,
Richard Wesel,
Abeer Alwan,

Page (NA) Paper number 2040

Abstract:

This paper presents an embedded joint source-channel coding scheme of speech. The source coder is an embedded variable bit rate perceptually based sub-band coder producing bits with different error sensitivities. The channel encoder is a Rate Compatible Punctured Trellis code (RCPT) which permits rate variability and unequal error protection by puncturing symbols. Furthermore, RCPT code design naturally incorporates large constellation sizes, allowing high information rate per symbol. The embedded speech coder and the rate compatible puncturing of symbols provide the embeddibility of the joint coding scheme. The coder is robust to acoustic noise and produces good quality speech for a wide range of channel conditions (AWGN or fading), allowing digital transmission of speech with analog-like graceful degradation.

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Adaptive Rate-Distortion-Based Thresholding: Application In JPEG Compression Of Mixed Images For Printing

Authors:

Marcia G Ramos,
Ricardo L De Queiroz,

Page (NA) Paper number 2008

Abstract:

In this paper, we propose a new technique for transform coding based on rate-distortion (RD) optimized thresholding (i.e. discarding) of wasteful coefficients. The novelty in this proposed algorithm is that the distortion measure is made adaptive. We apply the method to the compression of mixed documents (containing text, natural images, and graphics) using JPEG for printing. Although human visual system's response to compression artifacts varies depending on the region, JPEG applies the same coding algorithm throughout the mixed document. This paper takes advantage of perceptual classification to improve the performance of the standard JPEG implementation via adaptive thresholding, while being compatible with the baseline standard. A computationally efficient classification algorithm is presented, and the improved performance of the classified JPEG coder is verified. Tests demonstrate the method's efficiency compared to regular JPEG and to JPEG using non-adaptive thresholding. The non-stationary nature of distortion perception is true for most signal classes and the same concept can be used elsewhere.

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Separable Karhunen Loeve Transforms for the Weighted Universal Transform Coding Algorithm

Authors:

Hanying Feng,
Michelle Effros,

Page (NA) Paper number 2124

Abstract:

The weighted universal transform code (WUTC) is a two-stage transform code that replaces JPEG's single, non-optimal transform code with a jointly designed collection of transform codes to achieve good performance across a broader class of possible sources. Unfortunately, the performance gains of WUTC are achieved at the expense of significant increases in computational complexity and larger codes. We here present a faster, more space-efficient WUTC algorithm. The new algorithm uses separable coding instead of direct KLT. While separable coding gives performance comparable to that of WUTC, it uses only 1/8 of the floating-point multiplications and 1/32 of storage of direct KLT. Experimental results included in this work compare the performance of new separable WUTC with both the WUTC and other fast variations of that algorithm.

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A Theoretical Model for Time Code Modulation

Authors:

Dimitris Kalogiros,
Vassilis Stylianakis,

Page (NA) Paper number 1239

Abstract:

The traditional waveform coding techniques for digital communication systems convert the original analog input signal into a digital bit stream using uniform sampling in the time domain, i.e., PCM, DM, ADPCM. In this paper we propose the Time Code Modulation (TCM) technique as an alternative coding scheme, where information is extracted from the signal, only at the time instants when necessary. This results in a variable sampling rate, where its mean value is significantly less than the Nyquist rate. In addition we suggest a general theoretical model for TCM and we present simulation results for various implementations of TCM coders and decoders. A theoretical estimation of SNR vs. sampling rate performance is also presented.

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