An Efficient Method to Improve the Audio Quality Using AAC Low Complexity Decoder

This paper presents a new approach to design a Digital Audio Broadcast (DAB) audio decoder is introduced to improve the superiority of audio. Countries all over the world use DAB broadcasting systems more prominently, in Europe. DAB+ is the upgraded version of digital audio broadcasting. DAB and DAB+ coexist in many countries, so receivers are essential to be compatible with both standards. DAB+ is approximately twice as efficient as DAB due to the adoption of the AAC+ audio codec, and DAB+ can provide high quality audio with bit rates as low as 64 kbit/s. Integrating an MPEG-1 Layer II (MP2) decoder and Advanced Audio Coding Low Complexity (AAC LC) decoder provides a fundamental audio decoding for DAB and DAB+. The generated audio frames data from the DAB channel decoders are stored in RAM. The bit stream demultiplexer parses the quantized spectrum data in the audio. The inverse quantization performs the inverse quantization computation and synthesis filter generates the time domain Pulse Code Modulation (PCM) samples, all the above operation results writes them back to the audio RAM. The existing system of this project uses HE AAC V2 decoder, that system consists has SBR and PS technologies. This two technologies are used to improve the sound quality in low bit rate program. The proposed scheme is uses AAC LC and MP2 decoder it improve the sound quality in high bit rate. The simulation of this project is carried out by using MATLAB R2011a and Xilinx ISE 9.2i. DOI: 10.17762/ijritcc2321-8169.15039

Perception-aware low-power audio processing techniques for portable devices

Ph.DDOCTOR OF PHILOSOPH

Coprojeto de um decodificador de áudio AAC-LC em FPGA

Dissertação (mestrado)—Universidade de Brasília, Instituto de Ciências Exatas, Departamento de Ciência da Computação, 2013.A Codificação de áudio está presente hoje nos mais diversos aparelhos eletrônicos desde o rádio, a televisão, o computador, os tocadores de música portáteis e nos celulares. Em 2007, o governo do Brasil definiu o padrão do Sistema Brasileiro de TV Digital (SBTVD) que adotou o AAC Advanced Audio Coding para codificação de áudio. Neste trabalho, utilizamos a abordagem de coprojeto combinando software e hardware para implementar uma solução de alto desempenho e baixo consumo de energia em um FPGA, capaz de decodificar até 6 canais de áudio em tempo real. Apresentamos os detalhes da solução bem como os testes de desempenho e qualidade. Por fim, apresentamos os resultados de utilização de hardware e performance juntamente com uma comparação com as demais soluções encontradas na literatura. _______________________________________________________________________________________ ABSTRACTAudio Coding is present today in many electronic devices. It can be found in radio, tv, computers, portable audio players and mobile phones. In 2007 the Brazilian Government defined the brazilian Digital TV System standard (SBTVD) and adopted the AAC - Advanced Audio Coding as the audio codec. In this work we use the co-design of hardware and software approach to implement a high performance and low energy solution on an FPGA, able to decode up to 6 channels of audio in real-time. The solution architecture and details are presented along with performance and quality tests. Finally, hardware usage and performance results are presented and compared to other solutions found in literature

Survey of error concealment schemes for real-time audio transmission systems

This thesis presents an overview of the main strategies employed for error detection and error concealment in different real-time transmission systems for digital audio. The “Adaptive Differential Pulse-Code Modulation (ADPCM)”, the “Audio Processing Technology Apt-x100”, the “Extended Adaptive Multi-Rate Wideband (AMR-WB+)”, the “Advanced Audio Coding (AAC)”, the “MPEG-1 Audio Layer II (MP2)”, the “MPEG-1 Audio Layer III (MP3)” and finally the “Adaptive Transform Coder 3 (AC3)” are considered. As an example of error management, a simulation of the AMR-WB+ codec is included. The simulation allows an evaluation of the mechanisms included in the codec definition and enables also an evaluation of the different bit error sensitivities of the encoded audio payload.Ingeniería Técnica en Telemátic

Implementation of a MPEG 1 layer I audio decoder with variable bit lengths

One of the most popular forms of audio compression is MPEG (Moving Picture Experts Group). By using a VHDL (Very high-speed integrated circuit Hardware Description Language) implementation of a MPEG audio decoder and varying the word length of the constants and the multiplications used in the decoding process, and comparing the error, the minimum word length required can be determined. In general, the smaller the word length, the smaller the hardware resources required. This thesis is an investigation to find the minimum bit lengths required for each of the four multiplication sections used in a MPEG Audio decoder, that will still meet the quality levels specified in the MPEG standard. The use of the minimum bit lengths allows the minimum area resources of a FPGA (Field Programmable Gate Array) to be used. A FPGA model was designed that allowed the number of bits used to represent four constants and the results of the multiplications using these constants to vary. In order to limit the amount of data generated, testing was restricted to a single channel of audio data sampled at a frequency of 32kHz. This was then compared to the supplied C model distributed with the MPEG Audio Standard. It was found that for the MPEG audio coder to be fully compliant with the standard the bit lengths of the constants and the multiplications could be reduced by 75% and to be partial compliant with the standard, the bit lengths of the constants and the multiplications could be reduced by up to 82%. An implementation of a MPEG audio decoder in VHDL has the advantage of specific hardware, optimised, for all the different complex mathematical operations thereby reducing the repetitive operations and therefore power consumption and the time required performing these complex operations

Improvements in the Perceived Quality of Streaming and Binaural Rendering of Ambisonics

With the increasing popularity of spatial audio content streaming and interactive binaural audio rendering, it is pertinent to study the quality of the critical components of such systems. This includes low-bitrate compression of Ambisonic scenes and binaural rendering schemes. This thesis presents a group of perceptual experiments focusing on these two elements of the Ambisonic delivery chain. The first group of experiments focused on the quality of low-bitrate compression of Ambisonics. The first study evaluated the perceived timbral quality degradation introduced by the Opus audio codec at different bitrate settings and Ambisonic orders. This experiment was conducted using multi-loudspeaker reproduction as well as binaural rendering. The second study has been dedicated to auditory localisation performance in bitrate-compressed Ambisonic scenes reproduced over loudspeakers and binaurally using generic and individually measured HRTF sets. Finally, the third study extended the evaluated set of codec parameters by testing different channel mappings and various audio stimuli contexts. This study was conducted in VR thanks to a purposely developed listening test framework. The comprehensive evaluation of the Opus codec led to a set of recommendations regarding optimal codec parameters. The second group of experiments focused on the evaluation of different methods for binaural rendering of Ambisonics. The first study in this group focused on the implementation of the established methods for designing Ambisonic-to-binaural filters and subsequent objective and subjective evaluations of these. The second study explored the concept of hybrid binaural rendering combining anechoic filters with reverberant ones. Finally, addressing the problem of non-individual HRTFs used for spatial audio rendering, an XR-based method for acquiring individual HRTFs using a single loudspeaker has been proposed. The conducted perceptual evaluations identified key areas where the Ambisonic delivery chain could be improved to provide a more satisfactory user experience

Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression

Proceedings of the 8th Workshop on Detection and Classification of Acoustic Scenes and Events (DCASE 2023)

This volume gathers the papers presented at the Detection and Classification of Acoustic Scenes and Events 2023 Workshop (DCASE2023), Tampere, Finland, during 21–22 September 2023

A multi-level perspective analysis of the change in music consumption 1989-2014

This thesis seeks to examine the historical socio-technical transitions in the music industry through the 1990s and 2000s which fundamentally altered the way in which music is consumed along with the environmental resource impact of such transitions. Specifically, the investigation seeks to establish a historical narrative of events that are significant to the story of this transition through the use of the multi-level perspective on socio-technical transitions as a framework. This thesis adopts a multi-level perspective for socio-technical transitions approach to analyse this historical narrative seeking to identify key events and actors that influenced the transition as well as enhance the methodological implementation of the multi-level perspective. Additionally, this thesis utilised the Material Intensity Per Service unit methodology to derive several illustrative scenarios of music consumption and their associated resource usage to establish whether the socio-technical transitions experienced by the music industry can be said to be dematerialising socio-technical transitions. This thesis provides a number of original empirical and theoretical contributions to knowledge. This is achieved by presenting a multi-level perspective analysis of a historical narrative established using over 1000 primary sources. The research identifies, examines and discusses key events, actors and transition pathways denote the complex nature of dematerialising socio-technical systems as well as highlights specifically the influence different actors and actor groups can have on the pathways that transitions take. The thesis also provides a broader contribution to the understanding of dematerialisation and technology convergence