• Title/Summary/Keyword: Audio DSP

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Implementation of the High-Quality Audio System with the Separately Processed Musical Instrument Channels (악기별 분리처리를 통한 고음질 오디오 시스템 구현)

  • Kim, Tae-Hoon;Lee, Sang-Hak;Kim, Dae-Kyung;Lee, Sang-Chan
    • The Journal of the Acoustical Society of Korea
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    • v.32 no.4
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    • pp.346-353
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    • 2013
  • This paper deals with the implementation of a high-quality audio system for karaoke. For improving the key/tempo changes performance, we separated the audio into many musical instrument channels. By separating musical instrument channels, high-quality key/tempo changes can be achieved and we confirmed this using the cross-correlation distribution and the MOS evaluation. The improved audio system was implemented using the TMS320C6747 DSP with fixed/floating-point operations. The implemented audio system can perform the multi-channel WMA decoding, the MP3 encoding/decoding, the wav playing, the EQ, and the key/tempo changes in real time. The WMA channels used for processing the separated instrument channels. The audio system includs the MP3 encoding/decoding function for playing and recording and the wav channel for the effect sound.

The Design of Chorus DSP Chip Using Psychoacoustic Model and SOLA Algorithm (심리음향모델과 SOLA 알고리즘을 이용한 코러스 칩 설계)

  • 김태훈;박주성
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.3
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    • pp.11-19
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    • 2000
  • This research deals with the implementation procedures of a chorus processing DSP for karaoke system. It is necessary to compress the chorus data to store as many choruses as we can. We apply MPEG-1 audio algorithm to compress the chorus data. And the chorus system must be accompanied with the karaoke that can change the key and the tempo. So the chorus DSP must be able to change the key and tempo of the chorus data. We apply SOLA (Synchronized Overlap and Add) to do it. We designed the chorus DSP that can compress the chorus, change the key and tempo. And we verified the chorus DSP logic using FPGA. The used FPGA are two FLEX10K100s made by ALTERA. Finally we make the ASIC chip of chorus DSP and verify its operation.

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Implementation of On-site Audio Center based on AoIP

  • Lee, Jaeho;Kwon, Soonchul;Lee, Seunghyun
    • International journal of advanced smart convergence
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    • v.6 no.2
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    • pp.51-58
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    • 2017
  • Recently, rapid advances of Ethernet and IP technology have brought many changes in the sound industry. In addition, due to AoIP-based audio transmission technology, various problems of the acoustic system (sound quality deterioration due to long distance transmission, complicated wiring) have improved dramatically. However, when many distributed audio systems are connected with AoIP equipment, if there is a problem in the equipment, it is impossible to operate the connected system. AoIP equipment only can transmit audio signals but cannot adjust the system for acoustic environment. In this paper, AoIP equipment is to be installed with sound equipment on a one-to-one basis, so that various existing problems can be solved and adjustment of sound quality (reverberation, echo, delay and EQ) can be possible by AoIP-based OAC (On-site Audio Center) with built-in DSP function. As a result, uncompressed real-time transmission by distributed transmission/receipt module in OAC (On-site Audio Center) and high quality sound by adjustment of sound quality with built-in DSP can be acquired. It is expected that OAC based sound system will be the industry standard in ubiquitous environment.

Development of Digital/Analog Hybrid Redundancy System for Audio Mixer (오디오믹서용 디지털-아날로그 하이브리드 이중화 시스템 개발)

  • KIM, Kwan-Woong;CHO, JUPHIL
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.16 no.5
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    • pp.63-68
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    • 2016
  • Audio mixer is an electronic device which performs a mixing of multiple audio signals. Digital mixer having various functions and scalability is spreaded thanks to advanced DSP and IT technology. However, digital mixer is more vulnerable to stability comparing to conventional analog mixer in the digital error or software error sense because its control is executed by SW. To solve this problem, in this paper, we propose a multi-channel digital analog hybrid mixer scheme, digital mixer error detection mechanism and malfunctioning switching technique. Also we develop the audio mixer having digital-analog hybrid structure. By simulation, we can sense the error of digital mixer except power loss in a 120ms, change into analog mixer mode automatically and provide continuous broadcasting function without mixer function loss.

New Non-linear Inverse Quantization Algorithm and Hardware Architecture for Digital Audio Codecs (디지털 오디오 코덱을 위한 새로운 비선형 역 양자화 알고리즘과 하드웨어 구조)

  • Moon, Jong-Ha;Baek, Jae-Hyun;SunWoo, Myung-Hoon
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.1C
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    • pp.12-18
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    • 2008
  • This paper This paper proposes a new inverse-quantization(IQ) table interpolation algorithm, specialized Digital Signal Processor(DSP) instructions and hardware architecture for digital audio codecs. Non-linear inverse quantization algorithm is representatively used in both MPEG-1 Layer-3 and MPEG-2/4 Advanced Audio Coding(AAC). The proposed instructions are optimized for the non-linear inverse quantization. The proposed algorithm can minimize operational complexity which reduces total computational load. Performance comparisons show a significant improvement of average error. The proposed instructions and hardware architecture can reduce 20% of the instruction counts and minimize computational loads of IQ algorithms effectively compared with existing IQ table interpolation algorithms. Proposed algorithm can implement commercial DSPs.

A study on the extended fixed-point arithmetic computation for MPEG audio data processing (MPEG Audio 데이터 처리를 위한 확장된 고정소수점 연산처리에 관한 연구)

  • 한상원;공진흥
    • Proceedings of the IEEK Conference
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    • 2000.06b
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    • pp.250-253
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    • 2000
  • In this paper, we Implement a new arithmetic computation for MPEG audio data to overcome the limitations of real number processing in the fixed-point arithmetics, such as: overheads in processing time and power consumption. We aims at efficiently dealing with real numbers by extending the fixed-point arithmetic manipulation for floating-point numbers in MPEG audio data, and implementing the DSP libraries to support the manipulation and computation of real numbers with the fixed-point resources.

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AVS Video Decoder Implementation for Multimedia DSP (멀티미디어 DSP를 위한 AVS 비디오 복호화기 구현)

  • Kang, Dae-Beom;Sim, Dong-Gyu
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.46 no.5
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    • pp.151-161
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    • 2009
  • Audio Video Standard (AVS) is the audio and video compression standard that was developed for domestic video applications in China. AVS employs low complexity tools to minimize degradation of RD performance of the state-the-art video codec, H.264/AVC. The AVS video codec consists of $8{\times}8$ block prediction and the same size transform to improve compression efficiency for VGA and higher resolution sequences. Currently, the AVS has been adopted more and more for IPTV services and mobile applications in China. So, many consumer electronics companies and multimedia-related laboratories have been developing applications and chips for the AVS. In this paper, we implemented the AVS video decoder and optimize it on TI's Davinci EVM DSP board. For improving the decoding speed and clocks, we removed unnecessary memory operations and we also used high-speed VLD algorithm, linear assembly, intrinsic functions and so forth. Test results show that decoding speed of the optimized decoder is $5{\sim}7$ times faster than that of the reference software (RM 5.2J).

High Quality Multi-Channel Audio System for Karaoke Using DSP (DSP를 이용한 가라오케용 고음질 멀티채널 오디오 시스템)

  • Kim, Tae-Hoon;Park, Yang-Su;Shin, Kyung-Chul;Park, Jong-In;Moon, Tae-Jung
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.1
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    • pp.1-9
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    • 2009
  • This paper deals with the realization of multi-channel live karaoke. In this study, 6-channel MP3 decoding and tempo/key scaling was operated in real time by using the TMS320C6713 DSP, which is 32 bit floating-point DSP made by TI Co. The 6 channel consists of front L/R instrument, rear L/R instrument, melody, and woofer. In case of the 4 channel, rear L/R instrument can be replaced with drum L/R channel. And the final output data is generated as adjusted to a 5.1 channel speaker. The SOLA algorithm was applied for tempo scaling, and key scaling was done with interpolation and decimation in the time domain. Drum channel was excluded in key scaling by separating instruments into drums and non-drums, and in processing SOLA, high-quality tempo scaling was made possible by differentiating SOLA frame size, which was optimized for real-time process. The use of 6 channels allows the composition of various channels, and the multi-channel audio system of this study can be effectively applied at any place where live music is needed.

Optimization of Multichannel HE-AAC decoder for DVB-T (DVB-T를 워한 멀티채널 HE-AAC 디코더의 최적화)

  • Woo, Won-Hee
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2008.11a
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    • pp.251-253
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    • 2008
  • 최근 유럽에서 DVB-T HDTV 방송 표준이 정하지면서 오디오 포맷으로 HE-AAC가 채택되었다. HE-AAC는 압축효율은 높지만 연산량이 높아 낮은 성능의 DSP에서 수행하기에는 어려움이 있다. DVB-T에서는 5.1채널을 사용하고 있어 더욱더 많은 연산을 필요로 한다. 본 논문은 ISO/DEC 14496-3 MPEG4 HE(High Efficiency)-AAC의 Level4에 해당하는 Multichannel Decoder를 최적화하여 구현하고. 가장 많은 연산을 필요로 하는 Synthesis Filter Bank에 제안된 알고리즘을 적용하여 연산량을 줄였고 대부분의 연산부를 어셈블리로 코드 최적화를 하여 작은 성능의 DSP를 사용하여 실시간 Multichannel HE-AAC Audio Decoder의 구현이 가능하게 하였다. DVB-T 오디오 시스템에 필수로 필요한 Audio Description, Dynamic Range Control, Downmix 등을 함께 구현하여 실제 수신기에 사용이 가능하도록 하였다. DSP는 Samsung의 CalmRISC16 + MAC24 core 를 사용하였다.

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Real-Time Implementation of MPEG-1 Audio decoder on ARM RISC (ARM RISC 상에서의 MPEG-1 Audio decoder의 실시간 구현)

  • 김선태
    • Proceedings of the IEEK Conference
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    • 2000.11d
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    • pp.119-122
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    • 2000
  • Recently, many complex DSP (Digital Signal Processing) algorithms have being realized on RISC CPU due to good compilation, low power consumption and large memory space. But, real-time implementation of multiple DSP algorithms on RISC requires the minimum and efficient memory usage and the lower occupancy of CPU. In this thesis, the original floating-point code of MPEG-1 audio decoder is converted to the fixed-point code and then optimized to the efficient assembly code in time-consuming function in accord with RISC feature. Finally, compared with floating-point and fixed-point, about 30 and 3 times speed enhancements are achieved respectively. And 3~4 times memory spaces are spared.

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