O Sistema Brasileiro de Televisão Digital estabeleceu como padrão de codificação de áudio o algoritmo MPEG-4 Advanced Audio Coding, mais precisamente nos perfis Low Complexity, High Efficiency versão 1 e High Efficiency versão 2. O trabalho apresenta um estudo detalhado sobre o padrão, contendo desde alguns conceitos da psicoacústica como o mascaramento até a metodologia de decodificação do stream codificado, sempre voltado para o mercado do SBTVD. É proposta uma arquitetura em hardware para um decodificador compatível com o padrão MPEG-4 AAC LC. O decodificador é separado em dois grandes blocos mantendo em um deles o banco de filtros, considerado a parte mais custosa em termos de processamento. No bloco restante é realizada a decodificação do espectro, onde ocorre a decodificação dos códigos de Huffman, o segundo ponto crítico do algoritmo em termos de demandas computacionais. Por fim é descrita a implementação da arquitetura proposta em VHDL para prototipação em um FPGA da família Cyclone II da Altera. / MPEG-4 Advanced Audio Coding is the chosen algorithm for the Brazilian Digital Television System (SBTVD), supporting the Low Complexity, High Efficiency version 1 and High Efficiency version 2 profiles. A detailed study of the algorithm is presented, ranging from psychoacoustics concepts like masking to a review of the AAC bitstream decoding process, always keeping in mind the SBTVD. A digital hardware architecture is proposed, in which the algorithm is split in two separate blocks, one of them containing the Filter Bank, considered the most demanding task. The other block is responsible for decoding the coded spectrum, which contains the second most demanding task of the system: the Huffman decoding. In the final part of this work the conversion of the proposed architecture into VHDL modules meant to be prototyped with an Altera Cyclone II FPGA is described.

O Sistema Brasileiro de Televisão Digital estabeleceu como padrão de codificação de áudio o algoritmo MPEG-4 Advanced Audio Coding, mais precisamente nos perfis Low Complexity, High Efficiency versão 1 e High Efficiency versão 2. O trabalho apresenta um estudo detalhado sobre o padrão, contendo desde alguns conceitos da psicoacústica como o mascaramento até a metodologia de decodificação do stream codificado, sempre voltado para o mercado do SBTVD. É proposta uma arquitetura em hardware para um decodificador compatível com o padrão MPEG-4 AAC LC. O decodificador é separado em dois grandes blocos mantendo em um deles o banco de filtros, considerado a parte mais custosa em termos de processamento. No bloco restante é realizada a decodificação do espectro, onde ocorre a decodificação dos códigos de Huffman, o segundo ponto crítico do algoritmo em termos de demandas computacionais. Por fim é descrita a implementação da arquitetura proposta em VHDL para prototipação em um FPGA da família Cyclone II da Altera. / MPEG-4 Advanced Audio Coding is the chosen algorithm for the Brazilian Digital Television System (SBTVD), supporting the Low Complexity, High Efficiency version 1 and High Efficiency version 2 profiles. A detailed study of the algorithm is presented, ranging from psychoacoustics concepts like masking to a review of the AAC bitstream decoding process, always keeping in mind the SBTVD. A digital hardware architecture is proposed, in which the algorithm is split in two separate blocks, one of them containing the Filter Bank, considered the most demanding task. The other block is responsible for decoding the coded spectrum, which contains the second most demanding task of the system: the Huffman decoding. In the final part of this work the conversion of the proposed architecture into VHDL modules meant to be prototyped with an Altera Cyclone II FPGA is described.

O Sistema Brasileiro de Televisão Digital estabeleceu como padrão de codificação de áudio o algoritmo MPEG-4 Advanced Audio Coding, mais precisamente nos perfis Low Complexity, High Efficiency versão 1 e High Efficiency versão 2. O trabalho apresenta um estudo detalhado sobre o padrão, contendo desde alguns conceitos da psicoacústica como o mascaramento até a metodologia de decodificação do stream codificado, sempre voltado para o mercado do SBTVD. É proposta uma arquitetura em hardware para um decodificador compatível com o padrão MPEG-4 AAC LC. O decodificador é separado em dois grandes blocos mantendo em um deles o banco de filtros, considerado a parte mais custosa em termos de processamento. No bloco restante é realizada a decodificação do espectro, onde ocorre a decodificação dos códigos de Huffman, o segundo ponto crítico do algoritmo em termos de demandas computacionais. Por fim é descrita a implementação da arquitetura proposta em VHDL para prototipação em um FPGA da família Cyclone II da Altera. / MPEG-4 Advanced Audio Coding is the chosen algorithm for the Brazilian Digital Television System (SBTVD), supporting the Low Complexity, High Efficiency version 1 and High Efficiency version 2 profiles. A detailed study of the algorithm is presented, ranging from psychoacoustics concepts like masking to a review of the AAC bitstream decoding process, always keeping in mind the SBTVD. A digital hardware architecture is proposed, in which the algorithm is split in two separate blocks, one of them containing the Filter Bank, considered the most demanding task. The other block is responsible for decoding the coded spectrum, which contains the second most demanding task of the system: the Huffman decoding. In the final part of this work the conversion of the proposed architecture into VHDL modules meant to be prototyped with an Altera Cyclone II FPGA is described.

Thesis (M. S.)--Electrical and Computer Engineering, Georgia Institute of Technology, 2008. / Committee Chair: Bing Benny; Committee Co-Chair: Fred B-H. Juang; Committee Member: Gee-Kung Chang. Part of the SMARTech Electronic Thesis and Dissertation Collection.

Audio coding is used to compress digital audio signals, thereby reducing the amount of bits needed to transmit or to store an audio signal. This is useful when network bandwidth or storage capacity is very limited. Audio compression algorithms are based on an encoding and decoding process. In the encoding step, the uncompressed audio signal is transformed into a coded representation, thereby compressing the audio signal. Thereafter, the coded audio signal eventually needs to be restored (e.g. for playing back) through decoding of the coded audio signal. The decoder receives the bitstream and reconverts it into an uncompressed signal. ISO-MPEG is a standard for high-quality, low bit-rate video and audio coding. The audio part of the standard is composed by algorithms for high-quality low-bit-rate audio coding, i.e. algorithms that reduce the original bit-rate, while guaranteeing high quality of the audio signal. The audio coding algorithms consists of MPEG-1 (with three different layers), MPEG-2, MPEG-2 AAC, and MPEG-4. This work presents a study of the MPEG-4 AAC audio coding algorithm. Besides, it presents the implementation of the AAC algorithm on different platforms, and comparisons among implementations. The implementations are in C language, in Assembly of Intel Pentium, in C-language using DSP processor, and in HDL. Since each implementation has its own application niche, each one is valid as a final solution. Moreover, another purpose of this work is the comparison among these implementations, considering estimated costs, execution time, and advantages and disadvantages of each one.

Audio coding is used to compress digital audio signals, thereby reducing the amount of bits needed to transmit or to store an audio signal. This is useful when network bandwidth or storage capacity is very limited. Audio compression algorithms are based on an encoding and decoding process. In the encoding step, the uncompressed audio signal is transformed into a coded representation, thereby compressing the audio signal. Thereafter, the coded audio signal eventually needs to be restored (e.g. for playing back) through decoding of the coded audio signal. The decoder receives the bitstream and reconverts it into an uncompressed signal. ISO-MPEG is a standard for high-quality, low bit-rate video and audio coding. The audio part of the standard is composed by algorithms for high-quality low-bit-rate audio coding, i.e. algorithms that reduce the original bit-rate, while guaranteeing high quality of the audio signal. The audio coding algorithms consists of MPEG-1 (with three different layers), MPEG-2, MPEG-2 AAC, and MPEG-4. This work presents a study of the MPEG-4 AAC audio coding algorithm. Besides, it presents the implementation of the AAC algorithm on different platforms, and comparisons among implementations. The implementations are in C language, in Assembly of Intel Pentium, in C-language using DSP processor, and in HDL. Since each implementation has its own application niche, each one is valid as a final solution. Moreover, another purpose of this work is the comparison among these implementations, considering estimated costs, execution time, and advantages and disadvantages of each one.

Audio coding is used to compress digital audio signals, thereby reducing the amount of bits needed to transmit or to store an audio signal. This is useful when network bandwidth or storage capacity is very limited. Audio compression algorithms are based on an encoding and decoding process. In the encoding step, the uncompressed audio signal is transformed into a coded representation, thereby compressing the audio signal. Thereafter, the coded audio signal eventually needs to be restored (e.g. for playing back) through decoding of the coded audio signal. The decoder receives the bitstream and reconverts it into an uncompressed signal. ISO-MPEG is a standard for high-quality, low bit-rate video and audio coding. The audio part of the standard is composed by algorithms for high-quality low-bit-rate audio coding, i.e. algorithms that reduce the original bit-rate, while guaranteeing high quality of the audio signal. The audio coding algorithms consists of MPEG-1 (with three different layers), MPEG-2, MPEG-2 AAC, and MPEG-4. This work presents a study of the MPEG-4 AAC audio coding algorithm. Besides, it presents the implementation of the AAC algorithm on different platforms, and comparisons among implementations. The implementations are in C language, in Assembly of Intel Pentium, in C-language using DSP processor, and in HDL. Since each implementation has its own application niche, each one is valid as a final solution. Moreover, another purpose of this work is the comparison among these implementations, considering estimated costs, execution time, and advantages and disadvantages of each one.

In this thesis, we analyze the bandwidth requirements of MPEG-4 AVC video traffic and then propose and evaluate the accuracy of new MPEG-4 AVC video traffic models. First, we analyze the bandwidth requirements of the videos by comparing the statistical characteristics of the different frame types. We analyze their coefficient of variability, autocorrelation, and crosscorrelation in both short and long term. The Hurst parameter is also used to investigate the long range dependence of the video traces. We then provide an insight into B-frame dropping and its impact on the statistical characteristics of the video trace. This leads us to design two algorithms that predict the size of the B-frame and the size of the group of pictures (GOP) in the short-term. To evaluate the accuracy of the prediction, a model for the error is proposed. In a broadband cable network, B-frame size prediction can be employed by a cable headend to provision video bandwidth efficiently or more importantly, reduce bit rate variability and bandwidth requirements via selective B-frame dropping, thereby minimizing buffering requirements and packet losses at the set top box. It will be shown that the model provides highly accurate prediction, in particular for movies encoded in high quality resolution. The GOP size prediction can be used to provision bandwidth. We then enhance the B-frame and GOP size prediction models using a new scene change detector metric. Finally, we design an algorithm that predicts the size of different frame types in the long-term. Clearly, a long-term prediction algorithm may suffer degraded prediction accuracy and the higher complexity may result in higher latency. However, this is offset by the additional time available for long-term prediction and the need to forecast bandwidth usage well ahead of time in order to minimize packet losses during periods of peak bandwidth demands. We also analyze the impact of the video quality and the video standard on the accuracy of the model.

Submitted by William Justo Figueiro (williamjf) on 2015-07-08T20:56:12Z No. of bitstreams: 1 03b.pdf: 3126214 bytes, checksum: 0f98dbf86ae74816af91944aa7dec80f (MD5) / Made available in DSpace on 2015-07-08T20:56:12Z (GMT). No. of bitstreams: 1 03b.pdf: 3126214 bytes, checksum: 0f98dbf86ae74816af91944aa7dec80f (MD5) Previous issue date: 2013 / Nenhuma / A qualidade perceptual alcançada pelos codificadores de áudio depende diretamente da escolha de seus parâmetros. O codificador MPEG-4 AAC (Advanced Audio Coding), utilizado no Sistema Brasileiro de Televisão Digital (SBTVD), possui em sua estrutura uma etapa composta por um laço de iteração para escolher os parâmetros do codificador, de maneira dinâmica durante o processo de codificação. Este processo de escolha pode ser definido como um problema de Pesquisa Operacional, sendo um problema de Seleção de Partes, denominado como o Problema de Codificação AAC. A estrutura existente no codificador de referência, não resolve este problema de maneira ótima. Desta forma, este trabalho propõe o desenvolvimento e implementação de um modelo de uma estrutura de simulação, para encontrar os parâmetros do codificador de áudio MPEG-4 AAC, de maneira a otimizar a qualidade perceptual do áudio, para uma determinada taxa de bits (bit rate). A implementação da estrutura de otimização foi desenvolvida em linguagem C, utilizando as metaheurísticas Busca Tabu e Algoritmo Genético em uma estrutura híbrida. Através da minimização da métrica ANMR (Average Noise-to-Mask Ratio), o algoritmo procura identificar a melhor configuração dos parâmetros internos do codificador MPEG-4 AAC, de maneira que possa garantir uma qualidade perceptual para o sinal áudio. Os resultados obtidos utilizando a estrutura híbrida de otimização apresentaram valores menores para a métrica ANMR, ou seja, uma melhor qualidade perceptual de áudio, quando comparados com os resultados obtidos com o codificador de referência MPEG-4 AAC. / The perceptual quality achieved by audio encoders depends directly on the choice of its parameters. The MPEG-4 AAC (Advanced Audio Coding), used in the Brazilian Digital Television System (BDTS), has a step in its structure that consists in iteration loop to choose the parameters of the encoder dynamically during the encoding process. This selection process can be defined as a problem of Operational Research, being a Part Selection Problem, termed as AAC Encoding Problem. The structure in the reference encoder not solves this problem optimally. Thus, this paper proposes the development and implementation of a model simulation of a structure, to find the internal parameters of the MPEG-4 AAC audio encoder, so as to optimize the perceptual audio quality for a given bit rate. The implementation of the optimization framework was developed in ANSI C programming language, using the Tabu Search and Genetic Algorithm metaheuristics in a hybrid structure. Through the minimization of the ANMR (Average Noise-to-Mask Ratio) metric, the algorithm tries to identify the best configuration of internal parameters of the MPEG-4 AAC. The results obtained using the optimization hybrid structure achieve lower values for the ANMR metric, i.e., an better perceptual audio quality, compared with the obtained with the reference encoder MPEG-4 AAC.