• Title/Summary/Keyword: Audio Enhancement

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High Frequency Enhancement of Sound Using Wavelet Transform

  • Yoon Won-Jung;Lee Kang-Kyu;Park Kyu-Sik
    • Proceedings of the IEEK Conference
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    • summer
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    • pp.233-236
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    • 2004
  • This paper proposes new method for the enhancement of nonexistent high frequency spectral contents from low sample rate audio signal. For example, Due to the protocol constraint, the audio bandwidth of MP3 is restricted to 16Khz. Although band-restricted MP3 audio provide savings of storage space and network bandwidth, it suffers a major problem of a loss in high frequency fidelity such as localization, ambient information, and bright nature of audio. This paper provides a new mathematical analysis for the adaptive estimation of the high frequency contents based on the nature of the input low sample rate audio. Proposed method can be worked globally to any kind of audio such as speech and music that are restricted by sampling rate and bandwidth.

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Sound Enhancement of low Sample rate Audio Using LMS in DWT Domain (DWT영역에서 LMS를 이용한 저 샘플링 비율 오디오 신호의 음질 향상)

  • 백수진;윤원중;박규식
    • The Journal of the Acoustical Society of Korea
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    • v.23 no.1
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    • pp.54-60
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    • 2004
  • In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio, current digital audio is always restricted by sampling rate and bandwidth. This restriction normally results in low sample rate audio or calls for the data compression scheme such as MP3. However, they can only reproduce a lower frequency range than a regular CD quality because of the Nyquist sampling theory. Consequently they lose rich spatial information embedded in high frequency. The propose of this paper is to propose efficient high frequency enhancement of low sample rate audio using n adaptive filtering and DWT analysis and synthesis. The proposed algorithm uses the LMS adaptive algorithm to estimate the missing high frequency contents in DWT domain and it then reconstructs the spectrally enhanced audio by using the DWT synthesis procedure. Several experiments with real speech and audio are performed and compared with other algorithm. From the experimental results of spectrogram and sonic test, we confirm that the proposed algorithm outperforms the other algorithm and reasonably works well for the most of audio cases.

An Implementation of an ARM Platform based MP3 Sound Enhancement System (ARM 플랫폼 기반의 MP3 오디오 음질 향상 시스템 구현)

  • Oh, Sang-Hun;Park, Kyu-Sik
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.44 no.1
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    • pp.70-75
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    • 2007
  • In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio with 44.1 kHz sampling rate, current existing digital audio is always restricted by sampling rate and bandwidth. This kind of restriction normally can be resolved by using low bit rate audio codec such as MP3, OGG, and AAC. However it suffers a major problem such as a loss of high frequency fidelity. This high frequency loss will reproduce only the band-limited low-frequency part of audio in the standard CD-quality audio. In general, the high frequency contents of audio have lots of information such as localization and ambient information, and bright nature of audio. The purpose of this paper is to implement on ARM platform system that can effectively estimate and compensate the missing high frequency contents of MP3 audio. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed algorithms for MP3 audio quality enhancement.

Audio Enhancement Algorithm Using Adaptive Perceptual Filter (적응 지각 필터를 이용한 오디오 음질 개선 알고리즘)

  • 엄혜영;한헌수;홍민철;차형태
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.8
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    • pp.687-693
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    • 2003
  • In this paper, a new adaptive audio signal enhancement algorithm is proposed. In order to remove a broadband noise from a noisy signal, a filter is designed and applied adaptively to noisy audio signal. The noisy signal is first transformed to frequency domain and divided into bark domain to calculate excitation energy. A filter will be calculated to eliminate the noise by using the excitation energy and noisy energy which is obtained from a silent area. The filter is adaptively adjusted and continuously applied until the threshold point is met. The algorithm also works well even though the noise's energy change all of a sudden. SNR, NMR comparison and MOS Test are performed to show the effectiveness of the proposed algorithm.

A Robust Audio Fingerprinting System with Predominant Pitch Extraction in Real-Noise Environment

  • Son, Woo-Ram;Yoon, Kyoung-Ro
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2009.01a
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    • pp.390-395
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    • 2009
  • The robustness of audio fingerprinting system in a noisy environment is a principal challenge in the area of content-based audio retrieval. The selected feature for the audio fingerprints must be robust in a noisy environment and the computational complexity of the searching algorithm must be low enough to be executed in real-time. The audio fingerprint proposed by Philips uses expanded hash table lookup to compensate errors introduced by noise. The expanded hash table lookup increases the searching complexity by a factor of 33 times the degree of expansion defined by the hamming distance. We propose a new method to improve noise robustness of audio fingerprinting in noise environment using predominant pitch which reduces the bit error of created hash values. The sub-fingerprint of our approach method is computed in each time frames of audio. The time frame is transformed into the frequency domain using FFT. The obtained audio spectrum is divided into 33 critical bands. Finally, the 32-bit hash value is computed by difference of each bands of energy. And only store bits near predominant pitch. Predominant pitches are extracted in each time frames of audio. The extraction process consists of harmonic enhancement, harmonic summation and selecting a band among critical bands.

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Audio Listening Enhancement in Adverse Environment based on Loudness Restoration (라우드니스 복원에 기반한 잡음 환경에서의 오디오 청취 향상)

  • Pak, Junhyeong;Shin, Jong Won
    • Journal of the Institute of Electronics and Information Engineers
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    • v.50 no.12
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    • pp.210-216
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    • 2013
  • It is hard to listen to the music clearly in the presence of background noise. In this paper, a method that modifies the audio signal automatically to enhance the audio listening experience in adverse environment is proposed. Specifically, the method that amplifies the audio signal so that the perceived loudness of audio signal in each band becomes similar to that of the noiseless signal. The loudness perception model proposed by Moore et. al is utilized. Extending the previous work that is applied to speech reinforcement, the full band signal sampled at 48kHz is manipulated based on the loudness restoration principle. Moreover, based on the observation that the audio clarity is compromised even with loudness restored signal, a modification that intentionally boosts high frequency loudness more than lower band is also proposed. Experimental results showed that the proposed algorithm can enhance the audio listening experience in adverse environment.

An Enhancement of the MPEG-2 Audio Encoder Using General DSPs (범용 DSP를 이용한 MPEG-2 오디오 부호화기의 성능 개선)

  • 오현오;김성윤;윤대희;차일환;이준용
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 1997.11a
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    • pp.63-67
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    • 1997
  • The ISO(International Standard Organization) has standardized MPEG-2 audio. The MPEG-2 audio compression algorithm is based upon subband analysis and exploits the human auditory characteristics to achieve a low bit rate with minimum perceptual loss of audio signal quality. This thesis presents an enhanced MPEG-2 audio encoder using multiple TMS320C30 general purpose DSP's. The developed system is made up of five slave boards and one master board. Each slave board performs susband analysis psychoacoustic parameter calculation for one channel, and the master board manages bit allocation, quantization, and bit-stream formatting for all channels. Parallel processing and pipelining techniques are used in hardware structure and fast algorithms are applied in each subroutine to implement a real-time process. The implemented system supports multichannel up to 5.1 and various bitrates.

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A System-on-a-Chip Design for Digital TV

  • Rhee, Seung-Hyeon;Lee, Hun-Cheol;Kim, Sang-Hoon;Choi, Byung-Tae;Lee, Seok-Soo;Choi, Seung-Jong
    • JSTS:Journal of Semiconductor Technology and Science
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    • v.5 no.4
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    • pp.249-254
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    • 2005
  • This paper presents a system-on-a-chip (SOC) design for digital TV. The single LSI incorporates almost all essential parts such as CPU, ISO/IEC 11172/13818 system/audio/video decoders, a video post-processor, a graphics/OSD processor and a display processor. It has analog IP's inside such as video DACs, an audio PLL, and a system PLL to reduce the system-level implementation cost. Descramblers and Smart Card interface are included to support widely used conditional access systems. The video decoder can decode two video streams simultaneously. The DSP-based audio decoder can process various audio coding specifications. The functional blocks for video quality enhancement also form outstanding features of this SoC. The SoC supports world-wide major DTV services including ATSC, ARIB, DVB, and DIRECTV.

An Implementation of Sound Enhanced MPEG-1 Audio Decoder on Embedded OS Platform (음질향상 알고리즘을 내장한 MPEG-1 오디오 디코더의 Embedded OS 플랫폼에의 구현)

  • Hong, Sung-Min;Park, Kyu-Sik
    • Journal of Korea Multimedia Society
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    • v.10 no.8
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    • pp.958-966
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    • 2007
  • In this paper, we implement a sound-enhanced MPEG-1 audio decoder on embedded OS Platform. Low bit rate lossy audio codecs such as MP3, OGG, and AAC for mitigating the problems in storage space and network bandwidth suffer a major common problem such as a loss of high frequency fidelity of audio signal. This high frequency loss will reproduce only a band-limited low-frequency part of audio in the standard CD-quality audio. In order to overcome this problem, we embedded a sound enhancement algorithm into the MPEG-1 audio decoder and then the algorithms optimized according to the characteristic of the MPEG-1 audio layer I, II, III were implemented on an embedded OS platform. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed system compared to the standard MPEG-1 audio decoder.

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Intelligibility Enhancement of Multimedia Contents Using Spectral Shaping (스펙트럼 성형기법을 이용한 멀티미디어 콘텐츠의 명료도 향상)

  • Ji, Youna;Park, Young-cheol;Hwang, Young-su
    • Journal of the Institute of Electronics and Information Engineers
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    • v.53 no.11
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    • pp.82-88
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    • 2016
  • In this paper, we propose an intelligibility enhancement algorithm for multimedia contents using spectral shaping. The dialogue signals is essential to understand the plot of audio-visual media contents such as movie and TV. However, the non-dialogue components as like sound effects and background music often degrade the dialogue clarity. To overcome this problem, this paper tries to improves the dialogue clarity of audio soundtracks which contain important cues for the visual scenes. In the proposed method, the dialogue components are first detected by soft masker based on speech presence probability (SPP) which is widely used in speech enhancement field. Then, extracted dialogue signals are applied to the spectral shaping method. It reallocate the spectral-temporal energy of speech to enhanced the intelligibility. The total energy is maintained as unchanged via a loudness normalization process to prevent saturation. The algorithm was evaluated using the modeled and real movie soundtracks and it was shown that the proposed algorithm enhances the dialogue clarity while preserving the total audio power.