This paper presents the implementation of an MPEG-2 advanced audio coding (AAC) coder on a fixed-point DSP. AAC is a powerful coding algorithm and offers high-quality multi-channel surround audio. We also discuss the ...
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This paper presents the implementation of an MPEG-2 advanced audio coding (AAC) coder on a fixed-point DSP. AAC is a powerful coding algorithm and offers high-quality multi-channel surround audio. We also discuss the implementation issue and effort in porting AAC algorithms to the DSP.
In this paper we propose a multiple frequency harmonics model for analysis and synthesis of audio signals. The novelty of this model is that a composite sound can be represented by a few number (two or three) of frequ...
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In this paper we propose a multiple frequency harmonics model for analysis and synthesis of audio signals. The novelty of this model is that a composite sound can be represented by a few number (two or three) of frequency harmonics, each with time-varying fundamental frequency. We present the model and some key issues in tuning it. Results demonstrate that the model is valid and promising. It is convenient for time- and frequency-scale modifications and is of interest for low bit rate audio coding.
We consider optimum Transform coding. The objective is to minimize the distortion variance by selecting an optimum transform matrix and bit allocation among the sub-band signals. Under an asymptotic expression for the...
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We consider optimum Transform coding. The objective is to minimize the distortion variance by selecting an optimum transform matrix and bit allocation among the sub-band signals. Under an asymptotic expression for the quantizer distortion function it is known that the optimum transform can be chosen to be orthonormal and regardless of the precise allocated bit values. In this paper we have given sufficient conditions on the quantizer distortion function that ensure this property.
This paper investigates tradeoff between complexity and memory size in the 3GPP enhanced aacPlus decoder based on 16-bit fixed-point DSP implementation. In order to investigate this tradeoff, the speed- and the-memory...
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This paper investigates tradeoff between complexity and memory size in the 3GPP enhanced aacPlus decoder based on 16-bit fixed-point DSP implementation. In order to investigate this tradeoff, the speed- and the-memory conscious decoders are implemented. The maximum number of operations for the implemented speed-conscious decoder is 29.3 million cycles per second (MCPS) for a 32 kb/s bitstream. The maximum number of operations for the memoryconscious decoder, where 70% of the data are allocated to an external memory area, increases by 5.7 MCPS (19%) for the bitstream. The investigation of this tradeoff provides an actual relationship between the computational complexity and the internal memory size of the 3GPP enhanced aacPlus decoder. The implemented decoders enable music download and streaming services on next-generation mobile terminals.
Speech and audio codecs are usually designed such that they encode all the frequency bands of the input signal spectrum. If the higher bands do not contain any perceptually meaningful content, these codecs often do no...
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ISBN:
(纸本)9781479975921
Speech and audio codecs are usually designed such that they encode all the frequency bands of the input signal spectrum. If the higher bands do not contain any perceptually meaningful content, these codecs often do not work optimally as they assign part of the available bit budget to encode these bands. In this paper we describe a bandwidth detection algorithm that determines the effective audio bandwidth of the input signal. This information is used to set the codec to its optimal configuration and consequently increase the coding efficiency for band-limited signals by allocating bits to encode only the useful bandwidth. The presented algorithm has been used in the new codec for Enhanced Voice Services (EVS), recently standardized by 3GPP, but it can be employed in other codecs as well.
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