• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.1
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• pp.18-25
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• 2018

In this paper, we present a low-power implementation of the multi-channel hearing aid system using a general-purpose DSP chip. The system includes an acoustic amplification algorithm based on Wide Dynamic Range Compression (WDRC), an adaptive howling canceller, and a single-channel noise reduction algorithm. To achieve a low-power implementation, each algorithm is re-constructed in forms of integer program, and the integer program is converted to the assembly program using BelaSigna(R) 250 instructions. Through experiments using the implementation system, the performance of each processing algorithm was confirmed in real-time. Also, the clock of the implementation system was measured, and it was confirmed that the entire signal processing blocks can be performed in real time at about 7.02MHz system clock.

• Proceedings of the KOSOMBE Conference
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• v.1997 no.05
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• pp.342-344
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• 1997

Owing to technical advances in very large-scale integrated circuits (VLSI), high-speed digital signal processing (DSP) chips become fast enough to allow for real-time implementation of hearing aid algorithms in units small enough to be wearable. In this paper, we present a digital hearing aid processor (DHAP) chip built around a general-purpose 16-bit DSP core. The designed DHAP performs a nonlinear loudness correction of 8 octave frequency bands based on audiometric measurements. By employing a programmable DSP, the DHAP provides all the flexibility needed to implement audiological algorithms. In addition, the has a low power feature and $5.410\times5.720mm^2$ dimensions that fit for wearable devices.

• The Journal of the Acoustical Society of Korea
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• v.8 no.5
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• pp.95-101
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• 1989

For developing new algorithms or dedicated hardware by using general purpose Digital Signal Processor chip, emulator H/W and simulator S/W are indispensible. But the most of DSP emulators have limitations on H/W flexibility according to their generalized architectures. In this paper, a DSP evaluation system for real time signal processing was developed using TMS 32020. The I/O buffers storing acquisition data of the system were designed to have variable length $(1\sim2048samp1es) &$ sampling frequency $l00\sim8KHz$.

• Proceedings of the KIEE Conference
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• 1990.11a
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• pp.439-442
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• 1990

A powerful general purpose image processing processor is developed using a high-speed DSP chip, TMS320C30. The image processing processor, compatible to the standard VME bus, is composed of VME bus interface unit, video rate image grabbing/coding unit, TMS320C30 interface unit and bank of high-speed SRAMs. The performance is evaluated experimentally with the general image processing algorithms and the results show that the developed processor is capable of high speed image processing.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.1
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• pp.96-102
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• 1995

This paper presents the VHSIC Hardware Description Language(VHDL) implementation of the Fixed Point Covariance Lattice(FLAT) algorithm for an Linear Predictive Coding(LPC) analysis and its related algorithms, such as the forth order high pass Infinite Impulse Response(IIR) filter, covariance matrix calculation, and Spectral Smoothing Technique(SST) in the Vector Sum Exited Linear Predictive(VSELP) speech coder that has been Selected as the standard speech coder for the North America and Japanese digital cellular. Existing Digital Signal Processor(DSP) chips used in digital cellular phones are derived from general purpose DSP chips, and thus, these DSP chips may not be optimal and effective architectures are to be designed for the above mentioned algorithms. Then we implemented the VHDL code based on the C code, Finally, we verified that VHDL results are the same as C code results for real speech data. The implemented VHDL code can be used for performing logic synthesis and for designing an LPC Application Specific Integrated Circuit(ASOC) chip and DsP chips. We first developed the C language code to investigate the correctness of algorithms and to compare C code results with VHDL code results block by block.

• Journal of Satellite, Information and Communications
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• v.5 no.1
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• pp.38-42
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• 2010

This paper shows that beamfoming module deign to adapt smart antenna system in CDMA2000 environments. The designed beamfroming module has been implemented on a general-purpose DSP as a test-bed to confirm the superior performances based on real-time processing. From the various simulation result, it is confirm that beamforming module is provide a superior beampattern in smart antenna system.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1995.06a
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• pp.165-170
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• 1995

In this paper, it is attempted to apply the adaptive digital signal processing to compensate for a linear distortion of a loudspeaker and implement a real time hardware for that purpose. The real time system is implemented by using the DSP56001, a general purpose signal processor, as a host processor and the DSP56200, a cascadable adaptive FIR filter peripheral chip, as an adaptive digital filter. The system has 1000 taps at a 44.1kHz. After inverse modeling of under_compensation_speaker, the system reduces loudspeaker's linear distortions by pre-processing an input audio signal to loudspeaker. The experiment shows satisfactory results; after adaption with white noise as input signal for 60sec, the flat amplitude and linear phase frequency characteristics is found to lie over a wide frequency range of 100Hz to 20kHz.

• The Journal of the Acoustical Society of Korea
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• v.19 no.1
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• pp.78-83
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• 2000

Vocoders used in digital mobile phone adopt new improved algorithm to achieve better communication quality. Therefore the communication problem occurs between mobile phones using different vocoder algorithms. In this paper, the efficient implementation of 8kbps and 13kbps QCELP into one DSP chip to solve this problem is presented. We also describe the optimization method at each level, that is, algorithm-level, equation-level, and coding-level, to reduce the complexity for the QCELP vocoder algorithm implementation. The complexity in the codebook search-loop that is the main part for the QCELP algorithm complexity can be reduced about 50% by using these optimizations. The QCELP implementation with our DSP requires only 25 MIPS of computation for the 8kbps and 33 MIPS for the 13kbps ones. The DSP for our real-time implementation is a 16-bit fixed-point one specifically designed for vocoder applications and has a simple architecture compared to general-purpose ones in order to reduce the power consumption.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.12-20
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• 1996

A real-time Dolby AC-3 decoder has been implemented using a 20-bit fixed point general purpose DSP chip. It is shown that AC-3 decoder of this paper has same performance as decoder of C-simulator on PC. And also, it is applied to DVD player. Especially, in this paper, we proved that it has shorter latency time than that of previous AC-3 decoder.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.38 no.12
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• pp.44-55
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• 2001

MAC(Multiply and ACcumulate) is the core operation of multimedia data processing. Because MAC units implemented on traditional DSP units or embedded processors have latency of three cycles and cannot operate on multiple data simultaneously, then, performances are seriously limited. Many high end general purpose microprocessors have SIMD MAC unit as a functional unit. But these high end MAC units must support pipeline structure for various operation modes and high clock frequency, which makes control logic complex and increases chip area. In this paper, a 64bit SIMD MAC unit for embedded processors is designed. It is implemented to have a latency of one clock cycle to remove pipeline control logics and a minimal area overhead for SIMD support is added to existing Booth multipliers.