Masking threshold

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The masking threshold is the sound pressure level of a sound needed to make the sound perceptible in the presence of another noice, called a "masker". (Wikipedia.org)






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ICC 2021 - IEEE International Conference on Communications

IEEE ICC is one of the two flagship IEEE conferences in the field of communications; Montreal is to host this conference in 2021. Each annual IEEE ICC conference typically attracts approximately 1,500-2,000 attendees, and will present over 1,000 research works over its duration. As well as being an opportunity to share pioneering research ideas and developments, the conference is also an excellent networking and publicity event, giving the opportunity for businesses and clients to link together, and presenting the scope for companies to publicize themselves and their products among the leaders of communications industries from all over the world.


2020 IEEE International Conference on Image Processing (ICIP)

The International Conference on Image Processing (ICIP), sponsored by the IEEE SignalProcessing Society, is the premier forum for the presentation of technological advances andresearch results in the fields of theoretical, experimental, and applied image and videoprocessing. ICIP 2020, the 27th in the series that has been held annually since 1994, bringstogether leading engineers and scientists in image and video processing from around the world.


2020 IEEE International Conference on Systems, Man, and Cybernetics (SMC)

The 2020 IEEE International Conference on Systems, Man, and Cybernetics (SMC 2020) will be held in Metro Toronto Convention Centre (MTCC), Toronto, Ontario, Canada. SMC 2020 is the flagship conference of the IEEE Systems, Man, and Cybernetics Society. It provides an international forum for researchers and practitioners to report most recent innovations and developments, summarize state-of-the-art, and exchange ideas and advances in all aspects of systems science and engineering, human machine systems, and cybernetics. Advances in these fields have increasing importance in the creation of intelligent environments involving technologies interacting with humans to provide an enriching experience and thereby improve quality of life. Papers related to the conference theme are solicited, including theories, methodologies, and emerging applications. Contributions to theory and practice, including but not limited to the following technical areas, are invited.


2020 IEEE International Electron Devices Meeting (IEDM)

the IEEE/IEDM has been the world's main forum for reporting breakthroughs in technology, design, manufacturing, physics and the modeling of semiconductors and other electronic devices. Topics range from deep submicron CMOS transistors and memories to novel displays and imagers, from compound semiconductor materials to nanotechnology devices and architectures, from micromachined devices to smart -power technologies, etc.


2020 IEEE Nuclear Science Symposium and Medical Imaging Conference (NSS/MIC)

All areas of ionizing radiation detection - detectors, signal processing, analysis of results, PET development, PET results, medical imaging using ionizing radiation


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Advanced Packaging, IEEE Transactions on

The IEEE Transactions on Advanced Packaging has its focus on the modeling, design, and analysis of advanced electronic, photonic, sensors, and MEMS packaging.


Applied Superconductivity, IEEE Transactions on

Contains articles on the applications and other relevant technology. Electronic applications include analog and digital circuits employing thin films and active devices such as Josephson junctions. Power applications include magnet design as well asmotors, generators, and power transmission


Audio, Speech, and Language Processing, IEEE Transactions on

Speech analysis, synthesis, coding speech recognition, speaker recognition, language modeling, speech production and perception, speech enhancement. In audio, transducers, room acoustics, active sound control, human audition, analysis/synthesis/coding of music, and consumer audio. (8) (IEEE Guide for Authors) The scope for the proposed transactions includes SPEECH PROCESSING - Transmission and storage of Speech signals; speech coding; speech enhancement and noise reduction; ...


Biomedical Engineering, IEEE Transactions on

Broad coverage of concepts and methods of the physical and engineering sciences applied in biology and medicine, ranging from formalized mathematical theory through experimental science and technological development to practical clinical applications.


Circuits and Systems for Video Technology, IEEE Transactions on

Video A/D and D/A, display technology, image analysis and processing, video signal characterization and representation, video compression techniques and signal processing, multidimensional filters and transforms, analog video signal processing, neural networks for video applications, nonlinear video signal processing, video storage and retrieval, computer vision, packet video, high-speed real-time circuits, VLSI architecture and implementation for video technology, multiprocessor systems--hardware and software-- ...


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Xplore Articles related to Masking threshold

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Perceptual coding of speech using a fast wavelet packet transform algorithm

1996 8th European Signal Processing Conference (EUSIPCO 1996), 1996

This paper presents a new speech coding algorithm based on a fast wavelet packet transform algorithm and psychoacoustic modeling. The employed FFT-like overlapped block orthogonal transform allows us to approximate the auditory critical band decomposition in an efficient manner, which is a major advantage over previous approaches. Owing to such a decomposition of the original signal, we make use of ...


A backward-adaptive perceptual audio coder

1996 8th European Signal Processing Conference (EUSIPCO 1996), 1996

This paper presents a new audio compression algorithm that includes a nonuniform filter bank, gain-adaptive logarithmic quantizers, arithmetic entropy coding and an explicit psychoacoustic model to adapt the quantization according to perceptual considerations. Unlike existing perceptual coders, the new system is backward-adaptive, i.e., adaptation depends exclusively on already quantized samples, not on the original signal. We discuss the advantages of ...


Sound quality and transmission characteristic improvement for acoustic OFDM

2010 10th International Symposium on Communications and Information Technologies, 2010

In this paper, we propose a novel method to embed information data on DCT(Discrete Cosine Transform) domain and control transmission power of each sub-carrier using frequency masking for acoustic OFDM(Orthogonal Frequency Division Multiplex) in order to improve the sound quality and the transmission characteristic. In the acoustic OFDM, information data is placed on the phase of each sub-carrier, so that ...


A new Wiener filtering based detection scheme for time domain perceptual audio watermarking

2004 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2004

The paper presents a new detection method for a spread spectrum and perceptual watermarking system, viewed as a hidden data transmission system, where the generic detection operation is achieved by a Wiener deconvolution filter. We point out the insufficiencies of the generic reception scheme concerning truncation errors and an ill-conditioned deconvolution operation, and we propose a cascade realization of the ...


Speech enhancement based on masking properties of the auditory system

1995 International Conference on Acoustics, Speech, and Signal Processing, 1995

This paper addresses the problem of the intelligibility enhancement of speech corrupted by additive background noise in a single channel system. The proposed algorithm uses a criterion based on the human perception. It is a variation of the well-known spectral subtraction method which is attractive because of its simplicity, but introduces an unnatural and unpleasant residual noise. The proposed approach ...


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IEEE-USA E-Books

  • Perceptual coding of speech using a fast wavelet packet transform algorithm

    This paper presents a new speech coding algorithm based on a fast wavelet packet transform algorithm and psychoacoustic modeling. The employed FFT-like overlapped block orthogonal transform allows us to approximate the auditory critical band decomposition in an efficient manner, which is a major advantage over previous approaches. Owing to such a decomposition of the original signal, we make use of the human ear masking properties to decrease the mean bit rate of the encoder.

  • A backward-adaptive perceptual audio coder

    This paper presents a new audio compression algorithm that includes a nonuniform filter bank, gain-adaptive logarithmic quantizers, arithmetic entropy coding and an explicit psychoacoustic model to adapt the quantization according to perceptual considerations. Unlike existing perceptual coders, the new system is backward-adaptive, i.e., adaptation depends exclusively on already quantized samples, not on the original signal. We discuss the advantages of backward adaptiveness and show that it can be successfully applied to perceptual coding.

  • Sound quality and transmission characteristic improvement for acoustic OFDM

    In this paper, we propose a novel method to embed information data on DCT(Discrete Cosine Transform) domain and control transmission power of each sub-carrier using frequency masking for acoustic OFDM(Orthogonal Frequency Division Multiplex) in order to improve the sound quality and the transmission characteristic. In the acoustic OFDM, information data is placed on the phase of each sub-carrier, so that the sound quality deteriorates as the transmission rate increases. Moreover, sub-carriers with low electric power deteriorate the whole transmission characteristic due to additive noise because the frequency spectrum of audio signal is not flat. We therefore propose an embedding method using DCT for the deterioration of the sound quality and a power control method using frequency masking for the deterioration of the transmission characteristic. Moreover, we demonstrate that the combination of the proposed methods can simultaneously realize an increased transmission rate and an improved sound quality through some simulation results.

  • A new Wiener filtering based detection scheme for time domain perceptual audio watermarking

    The paper presents a new detection method for a spread spectrum and perceptual watermarking system, viewed as a hidden data transmission system, where the generic detection operation is achieved by a Wiener deconvolution filter. We point out the insufficiencies of the generic reception scheme concerning truncation errors and an ill-conditioned deconvolution operation, and we propose a cascade realization of the reception filter, which significantly improves the detection performances (to a multiplicative factor of 20) at higher bit rates, even in the presence of MPEG compression.

  • Speech enhancement based on masking properties of the auditory system

    This paper addresses the problem of the intelligibility enhancement of speech corrupted by additive background noise in a single channel system. The proposed algorithm uses a criterion based on the human perception. It is a variation of the well-known spectral subtraction method which is attractive because of its simplicity, but introduces an unnatural and unpleasant residual noise. The proposed approach incorporates in this method considerations about noise masking of the auditory system. It succeeds in finding the best trade- off between noise reduction and speech distortion in a perceptual sense. Simulations show perceptually very satisfactory results and objective measures indicate a quality improvement. The speech processed with this new algorithm sounds more pleasant to a human listener than those obtained by the classical methods. This shows the relevance to incorporate perceptual aspects in the enhancement process.

  • Performance of the modified Bark spectral distortion as an objective speech quality measure

    The modified Bark spectral distortion (MBSD), used for an objective speech quality measure, was presented previously (see IEEE Speech Coding Workshop, p.55-6, 1997). The MBSD measure takes into account the noise masking threshold in order to use only audible distortions in the calculation of the distortion measure. Preliminary simulation results have shown improvement of the MBSD over the conventional BSD. In this paper, performance of the MBSD is reported in terms of frame sizes, speech classes, and spectral regions. The performance of the MBSD is not very sensitive to the frame size. The performance of the MBSD for voiced speech is almost the same as for non-silent speech. The high frequency region appears to play an important role in human perception of speech quality.

  • Audio watermarking using m-sequences and temporal masking

    A novel algorithm for embedding a spread-spectrum based watermark into uncompressed, raw audio sequences is presented. The scheme takes advantage of temporal masking phenomenon in the human auditory system (HAS) in order to embed watermark data below the masking threshold of the audio signal. Detection of the watermark is done by blind detection, without using the original audio. As watermarking embedding and extraction are performed in the temporal domain, transformations to and from the frequency domain are not performed, neither on the embedding nor extraction part of the proposed scheme, resulting in the calculation simplicity of the embedding and detection process. The algorithm proved to be robust against mp3 compression (resulting in a BER of about 1.7%) and other common signal processing attacks, while the embedded watermark introduces no audible distortion of the host audio signal.

  • Noise reduction in audio signals based on the perceptual coding approach

    A new concept for the reduction of noise affecting audio signals transmitted in telecommunication channels is proposed. This concept is exploiting some features of the human auditory system. A strong subjective effect of noise suppression in noisy audio can be obtained by uplifting the masking thresholds above the estimated level of the noisy components or by reducing this level in such a way that the components be maintained just below the masking thresholds. The foundations of the engineered method together with the appropriate algorithms are described. A discussion on the results of experiments carried out and some conclusions are also included. The main focus is put on the perceptual foundations of the noise reduction method.

  • Auditory masking based acoustic front-end for robust speech recognition

    This paper presents an acoustic front-end which uses the properties of auditory masking for extracting acoustic features from the speech signal. Using the properties of simultaneous masking found in the human auditory system, we compute a masking threshold as a function of frequency for a given speech frame from its power spectrum. All those portions of the power spectrum which are below the auditory threshold are not heard by the human auditory system due to masking effects and hence can be discarded. These portions are replaced by the corresponding portions in the masking threshold spectrum. This modified power spectrum is processed by the linear prediction analysis or homomorphic analysis procedure to derive cepstral features for each speech frame. We study the performance of this front-end for speech recognition under noisy environments. This front-end performs significantly better than the conventional linear prediction or homomorphic analysis based front-ends for noisy speech. In terms of signal-to-noise ratio, simultaneous masking offers an advantage of more than 5 dB over the LPCC front-end in isolated word recognition experiments and 3 dB in continuous speech recognition experiments.

  • An improved stereophonic coding scheme compatible to the ISO/MPEG audio coding algorithm

    Stereophonic coding has been applied to the ISO/MPEG audio coding algorithm to improve the quality of stereo signals at very low bit rates. While reasonable quality has been achieved at bit rates of 2*64 kbit/s or lower, it was found that the proposed scheme could not achieve consistent good quality sounds for all types of audio signals. An improved stereophonic coding scheme is presented. This improved scheme employs the use of error minimisation and power equalization to obtain the improvements. Simulations of the scheme at a bit rate of 128 kbit per stereophonic channel produced a significant improvement in the quality of the audio over the original scheme. A quality that is close to transparency can be obtained at a bit rate of 128 kbit/s.<<ETX>>



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