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检索条件"主题词=Audio coding"

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Dual-mode Switching Used for Unified Speech and audio Codec

Dual-mode Switching Used for Unified Speech and Audio Codec

引用

2010 International Conference on audio,Language and Image Processing(2010年音频、语言与图像处理国际会议 ICALIP 2010)

作者： Min Lu Shuhua Zhang Weibei Dou Tsinghua National Laboratory for Information Science and Technology Department of Electronic Engineering Tsinghua University Beijing 100084 China

ISBN: (纸本)9781424458653

This paper presents a dual-mode switching method between time-domain codec and transform-domain codec of audio coding. It is a key technique of unified speech and audio (music) coding, since the replaying audio quality corresponds to the suitable codec selection and smooth switching between them. The proposed method consists of two steps, codec mode selection and switching. The binary decision trees (BDTs) algorithm is used to take a decision of mode selection, because of its advantages of high accuracy, low delay and low complexity. For smoothing transition between two codec, a pre-coding strategy is suggested in this paper. The classical speech codec. Algebraic Code Excited Linear Prediction (ACELP) and the Advanced audio coding (AAC) of MPEG are used for validating the proposed method. The scores of PESQ with 11 testing sequences show that the proposed switching method will not bring additional noise and can get higher objective evaluation of audio quality than single codec.

关键词： audio coding Dual mode codecs Additive noise PESQ Moving Pictures Experts Group smooth transition switching audio smoothing

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Quality enhancement of coded transient audio with a post-filter in frequency domain

Quality enhancement of coded transient audio with a post-fil...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： Jing Wang Hailong He Jingming Kuang Department of Electronic Engineering Beijing Institute of Technology Beijing China

This paper proposed a post-filter in frequency domain for enhancing the perceptual quality of the coded transient audio. The post-filter is used to filter the frequency coefficients of the coded transient and consists of a short-term post-filter which is derived from the Linear Predictive coding (LPC) predictor based on the Discrete Cosine Transform (DCT) coefficients of the transient frame. In order to compensate the overall spectral tilt, a spectral tilt compensation filter is used following the short-term post-filter. As a result, the post-filter in frequency domain shapes the temporal noise in time domain while introducing a little distortion in audio. This kind of post-filter doesn't need side information from the encoder and can be used in any audio decoder.

关键词： Transient analysis Discrete cosine transforms Frequency domain analysis Low pass filters Noise Time domain analysis audio coding

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HIGH FREQUENCY RECONSTRUCTION OF audio SIGNAL BASED ON CHAOTIC PREDICTION THEORY

HIGH FREQUENCY RECONSTRUCTION OF AUDIO SIGNAL BASED ON CHAOT...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Yong-tao SHA Chang-chun BAO Mao-shen JIA Xin Liu Speech and Audio Signal Processing Laboratory School of Electronic Information and Control Engineering Beijing University of Technology Beijing 100124 China

ISBN: (纸本)9781424442959;9781424442966

The quality of audio signals that have been encoded with low-bit rate audio coding standards is degraded because the high frequency information has been removed. The quality of such audio signals can, however, be improved by reconstructing the high frequency information which was lost. In this paper the principles of audio signal production and the characteristics of the human hearing system have been used to develop a blind high frequency reconstruction method based on chaotic prediction theory. Performance evaluation with objective and subjective tests has shown that this method is, in most cases, more efficient than other blind high frequency reconstruction methods.

关键词： audio coding Non-linear systems High Frequency Reconstruction Chaotic prediction audio coding Nonlinear systems High frequency audio signals audio Human Characteristics Subjective testing signals

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Template music transcription for different types of musical instruments

Template music transcription for different types of musical ...

引用

International Conference on Computer and Automation Engineering, ICCAE

作者： Md. Omar Faruqe Shamim Ahmad Md. Al-Mehedi Hasan Farazul Haque Bhuiyan Department of CSE BRAC University Dhaka Bangladesh Department of CSE RUET Rajshahi Bangladesh Department of CSE Rajshahi University Rajshahi Bangladesh

Automatically recovering the musical notation for a given signal allows reproducing and modifying the original performance. Music transcription can be considered as extracting the sequence of the notes that best models our audio signal. So it is understood as the process of analyzing a music signal in order to write down the parameters of the sounds occurring in it. Another principal application is structured audio coding: a MIDI-like representation which is highly compact but holds on to the identification and characteristics of a musical note to a significant extent. The applied notation can be the traditional musical notation which gives sufficient information for performing the piece using the available musical instruments. Automatic Music Transcription (extracting musical notes from an audio stream) is a complicated task due to the complexity of musical sounds that requires an effective solution. The scope of this article is in the automatic transcription of the harmonic and melodic parts of the real-world music signals. Algorithms are proposed that address two distinct sub-problems of music transcription. The main part of the article emphasizes the estimation of the several concurrent musical notes. The other sub-problem discusses musical meter estimation. This has to do with rhythmic aspects of music and refers to the estimation of the regular pattern of strong and weak beats in a piece of music. Signal processing methods for the automatic transcription of music are developed in this paper.

关键词： Instruments Music Multiple signal classification Frequency Humans Signal analysis Signal processing audio coding Data mining Streaming media

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A SOT based digital audio coder using reference frame ordering method

A SOT based digital audio coder using reference frame orderi...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Yu-Lin Wang Wei-Hsiang Liao Alvin Wen-Yu Su SCREAM Laboratory CSIE. National Cheng Kung University Tainan Taiwan

In SOT based coding methods, ordering of the coefficients could affect coding efficiency. A small coefficient placed close to tree root usually reduces the coding efficiency significantly. Hence, a good reordering scheme for a SOT is critical. In digital audio coding, information of neighboring frames is usually highly correlated. Based on this property, a SOT based audio coding scheme with a coefficient reordering method is proposed. The proposed coder may have the coefficients of the current frame reordered depending on the decoded results of some previous frames. The coding qualities of the proposed coder and MPEG-4 AAC coder are compared using ODG (Objective Differential Grade) on a variety of sound sources.

关键词： Decoding audio coding MPEG 4 Standard Encoding Distortion measurement Scalability Image coding Video coding Search methods Testing

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Lossy audio Compression via Compressed Sensing

Lossy Audio Compression via Compressed Sensing

引用

Data Compression Conference (DCC)

作者： Rubem J. V. de Medeiros Edmar C. Gurjão Joâo M. de Carvalho Electrical Engineer Department Federal University of Campina Grande Campina Grande Brazil

We propose a Compressed Sensing application to audio signals and analyze its audio perceptual quality with PEAQ. We analyzed the CS sensing framework for audio signals and evaluate its perceptual audio quality. The pr... 详细信息

ISBN: (纸本)9781424464258;9780769539942

关键词： audio compression Compressed sensing Signal analysis Sampling methods Discrete cosine transforms Data compression Data engineering audio coding Decoding System performance

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TOP-DOWN STRATEGIES IN PARAMETER SELECTION OF SINUSOIDAL MODELING OF audio

TOP-DOWN STRATEGIES IN PARAMETER SELECTION OF SINUSOIDAL MOD...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Toni Hirvonen Athanasios Mouchtaris Department of Computer Science University of Crete and Institute of Computer Science Foundation for Research and Technology - Hellas (FORTH-ICS) Heraklion Crete Greece

ISBN: (纸本)9781424442959;9781424442966

Sinusoidal modeling of audio requires the model parameters to be selected by analyzing the original signal spectrum. This paper proposes two improvements in sinusoidal selection by considering how psychoacoustic masking curves can be calculated using a top-down strategy in certain situations. First, a non-iterative component selection method to be used in combination with an added residual signal is presented. Tests indicate computational gain and quality increase when the method is used with a noise-synthesized residual. Secondly, the estimation of the masking curve in binaural listening when signals are panned is considered. Tests show that knowledge of the degree of panning is beneficial when heavy panning is applied to simultaneously rendered audio object signals.

关键词： audio coding sinusoidal modeling psychoacoustic masking audio audio coding Panning Modeling model parameters Masking signals

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Fixed quality layered truncation for scalable lossless audio

Fixed quality layered truncation for scalable lossless audio

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IEEE Conference on Industrial Electronics and Applications (ICIEA)

作者： Te Li Rongshan Yu Haiyan Shu Susanto Rahardja Institute for Infocomm Research (I2R) Agency for Science Technology and Research Singapore

The paper proposes a fixed quality layered bitstream truncator based on the MPEG-4 scalable lossless (SLS) audio coding system where, in contrast to original SLS truncator, the bitrate for each audio frame after truncation is not constant but instead an attempt is made to create constant quality audio. With the bitrate constraint for a period, the proposed system truncates each audio frame according to a pre-trained bitrate table. Simulation results show that with the same bandwidth, a constant and enhanced quality can be achieved by using the proposed method comparing with the one using the traditional fixed bitrate truncation method.

关键词： Laser sintering Bit rate Encoding MPEG 4 Standard audio coding Bandwidth Electronic switching systems ISO standards Paper technology Poles and towers

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ROBUST CRITICAL DATA RECOVERY FOR MPEG-4 AAC ENCODED BITSTREAMS

ROBUST CRITICAL DATA RECOVERY FOR MPEG-4 AAC ENCODED BITSTRE...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Ruijing Hu Xucen Huang Michel Kieffer Olivier Derrien Pierre Duhamel L2S CNRS-SUPELEC-Univ Paris-Sud 3 rue Joliot-Curie 91192 Gif-sur-Yvette France LMA CNRS Marseille France

ISBN: (纸本)9781424442959;9781424442966

This paper presents a bandwidth-efficient method for improved decoding of critical data generated by the MPEG-4 AAC audio coder when encoded bitstreams are transmitted over noisy channels. The critical data of each encoded frame is estimated using the redundancy due to the correlation between successive frame headers and using an optional CRC as an error-correcting code. Simulation results for an AWGN channel show a substantial link budget improvement (more than 5 dB) compared to a classical hard-decoding method. Once critical data are efficiently estimated, the work of previously proposed joint source-channel decoding techniques to recover the remaining parts of the MPEG4 AAC frames is significantly facilitated.

关键词： audio coding Cyclic codes Decoding MAP estimation Mobile communication Robustness audio coding cyclic codes DEcoding mobile communication MPEG 4 Standard automatic amplitude control AWGN channels Robustness CRITICAL

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Measurement and Analysis of Just Noticeable Difference of Interaural Level Difference Cue

Measurement and Analysis of Just Noticeable Difference of In...

引用

International Conference on Multimedia Technology

作者： Weiping Tu Ruimin Hu Heng Wang Wenqin Chen National Engineering Research Center of Multimedia Software Wuhan University of China Wuhan China Computer Science and Electronics department Mondragon Unibertsitatea Mondragon Spain National Engineering Research Center for Multimedia Software Wuhan University Wuhan Hubei CN

Just Noticeable Difference (JND) of interaural level difference (ILD) are measured in 30 subjects with nomal hearing. Single-frequency sine tones with the frequency of 200, 500, 1000, 2000, 5000, 8000 or 12000Hz are used in the subjective listening test. The reference ILDs are 0, 3, 6, 9, 12 and 22dB, indicating 6 directions of sound sources from midline to the direction of the right ear. The results show that direction of the sound source has notable effects on JNDs of ILD. At high frequency, JNDs increase dramatically with source frequency.

关键词： System-on-a-chip audio coding Quantization Transform coding Auditory system Humans Ear

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