Real-time MPEG-1 audio coding and decoding on a DSP chip

Murphy, C.D.; Anandakumar, K.

doi:10.1109/30.580383

Cited by 17 publications

(3 citation statements)

References 3 publications

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“…Pipelining and ping-pong buffers allow calculations to be performed in parallel with the input of the samples and the output of the bitstream. Past work has been done with implementing a DSP based MPEG-1 encoder as in [8]. A single floating point TI DSP was used to implement an encoder and decoder.…”

Section: Dsp Software Implementationmentioning

confidence: 99%

Design and implementation of a DSP based MPEG-1 audio encoder

Hoekstra

Shaaban

Czernikowski

et al. 1999

IEEE Trans. Consumer Electron.

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Unit of frequency measure based on critical bandwidth Coder/decoderprovides 16 bit samples of analog signal Discrete Cosine Transform Difference between two frequencies in barks. Finite Impulse Responsefilter based on the weighted sum of input values. Fast Fourier Transformtime to frequency transform Subbands that are less than the bound are coded in stereo, while 1 the sum of the two channels is coded for the higher subbands. Interrupt Service Routinecode executed to service an interrupt Kilo bits per second Least Significant Bytelower 8 bits of a 16 bit value The science of how the ear and brain perceive sounds. The effect of one tone or noise to make a test tone inaudible or less loud. The SPL of the surrounding frequencies that is masked out by the masking frequency Million Instructions Per Second Mask to Noise Ratio Moving Pictures Experts Group Most Significant Byteupper 8 bits of a 16 bit value Encoding of stereo signals as their sum and difference Noise like frequency components Personal Computer Pulse Code Modulation Continuous represented as discrete values.

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Section: Dsp Software Implementationmentioning

confidence: 99%

Design and implementation of a DSP based MPEG-1 audio encoder

Hoekstra

Shaaban

Czernikowski

et al. 1999

IEEE Trans. Consumer Electron.

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“…We mainly focus on realizing more efficient VLSI designs by reducing the number of multiplications; compared to adder components, multipliers are relatively slow and often result in considerably more demanding design restrictions [3]. The proposed algorithm is based upon the optimization of redundant computations carried out in subsequent synthesis filtering operations and also extends and complements techniques discussed in [4]- [6] (encoding only), [7]- [9] (decoding only), and [10]- [12] (both).…”

Section: Introductionmentioning

confidence: 98%

Optimized recursive subband synthesis windowing for implementing efficient MPEG audio decoders

Smet

Bruyland

2003

IEEE Signal Process. Lett.

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In this letter, we discuss a recursive optimization approach for the subband synthesis windowing component used in all three layers of the MPEG-1 and -2 audio decoding frameworks. We explain the design of the proposed algorithm and briefly discuss how it can be combined with other novel and existing optimization strategies. The impact of these optimizations on the number of multiplications and additions is shown, and the influence of the algorithm on the amount of memory required by the adapted decoder is also discussed. We show that the proposed algorithm offers very interesting possibilities for implementing efficient VLSI designs.Index Terms-Efficient decoder, MPEG audio, recursive windowing.

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“…Real time signal processing based on both general purpose microprocessors and fast digital signal processors (DSPs) is a technique that emerged over 20 years ago, and is now widely considered one of the fastest growing application areas in the field of digital technology. Applications include biomedical signal analysis, image analysis, image coding and decoding and audio signal enhancement (Ifeachor et al 1986, Murphy and Anandakumar 1997, Baccigalupi et al 1997. Typically for filtering, the analogue waveform is first digitized by an analogue to digital converter (ADC), and the binary values are transmitted to a DSP device that performs a real time convolution operation in discrete space using either a finite impulse response (FIR) or infinite impulse response (IIR) algorithm.…”

Section: Introductionmentioning

confidence: 99%

A real time programmable digital filter for biomedical signal enhancement incorporating a high-level design interface

Gaydecki¹

2000

Physiol. Meas.

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A simple but highly integrated digital signal processing system is described for real time filtering of biomedical signals. It includes the necessary processing and communications hardware, the processing code itself and a high-level software interface that enables the user to design and download arbitrary finite impulse response filters to the run-time system. The filter coefficients are calculated using the frequency sampling method. Since the filters are realized using a finite impulse response, no phase distortion is introduced into the processed signals. A unique feature of the design is the manner in which the software and hardware components have been organized as an intelligent system, obviating on the part of the user a detailed knowledge of filter design theory or any abilities in processor architecture and assembly code programming.

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Real-time MPEG-1 audio coding and decoding on a DSP chip

Cited by 17 publications

References 3 publications

Design and implementation of a DSP based MPEG-1 audio encoder

Design and implementation of a DSP based MPEG-1 audio encoder

Optimized recursive subband synthesis windowing for implementing efficient MPEG audio decoders

A real time programmable digital filter for biomedical signal enhancement incorporating a high-level design interface

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