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Advanced Audio Coding

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Advanced Audio Coding (AAC) is a digital audio compression format that provides high-quality sound at lower bit rates compared to its predecessors. It is designed to efficiently encode audio data for various applications, including streaming and broadcasting, while maintaining fidelity and minimizing file size.
lightbulbAbout this topic
Advanced Audio Coding (AAC) is a digital audio compression format that provides high-quality sound at lower bit rates compared to its predecessors. It is designed to efficiently encode audio data for various applications, including streaming and broadcasting, while maintaining fidelity and minimizing file size.
he name MPEG-4 High-Efficiency AAC (HE-AAC) refers to a family of recent audio coders that was developed by the International Organization for Standardization/ International Electrotechnical Commission (ISO/IEC) Moving Picture Experts... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper presents a double improvements to reduce the speech coding complexity of the pitch prediction in a Code-Excited Linear Prediction (CELP) coder. First, the pitch analysis structure is modified. A new fast Pitch Modelling by... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
Conventional music coders, based on a modified discrete cosine transform (MDCT) suffer greatly when lowering their bit-rate and delay. In particular, tonal music signals are penalized by short analysis windows and the variable length... more
As the latest extension of MPEG-4 Audio coding, MPEG-4 Lossless Audio Coding includes a scalable audio coding solution (SLS) that integrates the functionalities of lossless audio coding, perceptual audio coding, and fine granular scalable... more
As the latest extension of MPEG-4 Audio coding, MPEG-4 Lossless Audio Coding includes a scalable audio coding solution (SLS) that integrates the functionalities of lossless audio coding, perceptual audio coding, and fine granular scalable... more
As the latest extension of MPEG-4 Audio coding, MPEG-4 Lossless Audio Coding includes a scalable audio coding solution (SLS) that integrates the functionalities of lossless audio coding, perceptual audio coding, and fine granular scalable... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper presents ITU-T Embedded Variable Bit-Rate (EV-VBR) codec being standardized by Question 9 of Study Group 16 (Q9/16) as recommendation G.718. The codec provides a scalable solution for compression of 16 kHz sampled speech and... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
Audio coding based on Frequency Domain Linear Prediction (FDLP) uses autoregressive models to approximate Hilbert envelopes in frequency sub-bands. Although the basic technique achieves good coding efficiency, there is a need to improve... more
This paper proposed a novel fast algorithm and common structure design of analysis and synthesis quadrature mirror filterbanks (AQMF, SQMF) on the spectral band replication (SBR) in digital radio mondiale (DRM). Based on recent Lai et... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
Bandwidth extension (BWE) is an important parametric technique applied by modern audio coders in order to achieve efficient data rate compression at low bitrates. The perceptual quality of BWE enhanced signals is, however, often hampered... more
The Code-Excited Linear Prediction (CELP) model is very efficient in coding speech at low bit rates. However, if the bit rate of the coder is increased, the CELP model does not gain in quality as quickly as other approaches. Moreover, the... more
This paper presents ITU-T Embedded Variable Bit-Rate (EV-VBR) codec being standardized by Question 9 of Study Group 16 (Q9/16) as recommendation G.718. The codec provides a scalable solution for compression of 16 kHz sampled speech and... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper aims to produce a reference system that can simulate and calibrate degradations of conversational codecs which are currently used on telecommunications networks, for subjective assessment tests of voice quality. At first, 20... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
Presented here is MPEG-2 AAC low complexity profile decoder software for a low-power embedded RISC microprocessor, NEC V830 (300 mW @133 MHz). Fast processing methods for IMDCT reduce execution time by 41% and help achieve real-time... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
We present an integrated approach for flexible playout adaptation for high quality audio transmission over impaired network connections. The key concept of our framework is a continuous measurement of the transmission delay, the delay... more
The paper presents a novel perceptual based fuzzy approach towards classification and segmentation for MP3 and AAC audio in the compressed domain. The input audio is split into segments, which are classified as speech, music, fuzzy or... more
Bandwidth extension (BWE) is an important parametric technique applied by modern audio coders in order to achieve efficient data rate compression at low bitrates. The perceptual quality of BWE enhanced signals is, however, often hampered... more
Bandwidth extension (BWE) is an important parametric technique applied by modern audio coders in order to achieve efficient data rate compression at low bitrates. The perceptual quality of BWE enhanced signals is, however, often hampered... more
The present invention relates to audio signal processing, and in particular, to an apparatus and a method for generating a synthesis audio signal, an apparatus and a method for encoding an audio signal and an encoded audio signal. [0002]... more
This paper illustrates the suitability of ADSP 21160 for implementation of high performance DSP applications requiring 32 bit floating point precision. We present a real-time implementation of a multi-channel MPEG-2 AAC-LC encoder. The... more
We present an integrated approach for flexible playout adaptation for high quality audio transmission over impaired network connections. The key concept of our framework is a continuous measurement of the transmission delay, the delay... more
The reference model selected by MPEG for the forthcoming unified speech and audio codec (USAC) switches between a non-LPC based coding mode (based on AAC) operating in the transform domain and an LPC-based coding mode (derived from... more
In early 2012 the ISO/IEC JTC1/SC29/WG11 (MPEG) finalized the new MPEG-D Unified Speech and Audio Coding standard. The new codec brings together the previously separated worlds of general audio coding and speech coding. It does so by... more
The present invention relates to audio signal processing, and in particular, to an apparatus and a method for generating a synthesis audio signal, an apparatus and a method for encoding an audio signal and an encoded audio signal. [0002]... more
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