Pub Date : 2002-11-07DOI: 10.1109/ICDSP.2002.1028301
Su Cheol Kang, S. Hong
One of the most significant features for diagnostic echocardiographic images is to reduce speckle noise and improve image quality. We propose a simple and effective filter design for image denoising and contrast enhancement based on a multiscale wavelet method. Wavelet threshold algorithms replace small magnitude wavelet coefficients by zero and keep or shrink the other coefficients. This is basically a local procedure, since wavelet coefficients characterize the local regularity of a function. After we estimate the distribution of noise within an echocardiographic image, we apply it to a fitness wavelet threshold algorithm. A common way of estimating the speckle noise level in coherent imaging is to calculate the mean-to-standard-deviation ratio of the pixel intensity, often termed the equivalent number of looks (ENL), over a uniform image area. Unfortunately, this measure is not very robust, mainly due to the difficulty of identifying a uniform area in a real image. For this reason, we only use the S/MSE ratio, which corresponds to the standard SNR in case of additive noise. We have simulated some echocardiographic images by specialized hardware for a real-time application; processing of 512/spl times/512 images takes about 1 min. Our experiments show that the optimal threshold level depends on the spectral content of the image. With high spectral content, the noise standard deviation estimation performed at the finest level of the DWT tends to be over-estimated. Hence a lower threshold parameter is required to get the optimal S/MSE. The standard WCS theory predicts a threshold that depends only on the number of signal samples.
{"title":"A speckle reduction filter using wavelet-based methods for medical imaging application","authors":"Su Cheol Kang, S. Hong","doi":"10.1109/ICDSP.2002.1028301","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028301","url":null,"abstract":"One of the most significant features for diagnostic echocardiographic images is to reduce speckle noise and improve image quality. We propose a simple and effective filter design for image denoising and contrast enhancement based on a multiscale wavelet method. Wavelet threshold algorithms replace small magnitude wavelet coefficients by zero and keep or shrink the other coefficients. This is basically a local procedure, since wavelet coefficients characterize the local regularity of a function. After we estimate the distribution of noise within an echocardiographic image, we apply it to a fitness wavelet threshold algorithm. A common way of estimating the speckle noise level in coherent imaging is to calculate the mean-to-standard-deviation ratio of the pixel intensity, often termed the equivalent number of looks (ENL), over a uniform image area. Unfortunately, this measure is not very robust, mainly due to the difficulty of identifying a uniform area in a real image. For this reason, we only use the S/MSE ratio, which corresponds to the standard SNR in case of additive noise. We have simulated some echocardiographic images by specialized hardware for a real-time application; processing of 512/spl times/512 images takes about 1 min. Our experiments show that the optimal threshold level depends on the spectral content of the image. With high spectral content, the noise standard deviation estimation performed at the finest level of the DWT tends to be over-estimated. Hence a lower threshold parameter is required to get the optimal S/MSE. The standard WCS theory predicts a threshold that depends only on the number of signal samples.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"138 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131026042","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2002-11-07DOI: 10.1109/ICDSP.2002.1028276
C. B. D. Lima, Dirceu G. da Silva, A. Alcaim, J. A. Apolinário
This paper presents the performance of the AR-vector with cepstral mean subtraction (CMS) used to compensate the distortions caused by distinct telephone channels. The speaker recognition performance obtained with the use of CMS is compared with a system without compensation. With 60 s of speech signal used for training and 30 s used for testing, the error rate without channel normalization is around 2.82% against the 1.65% achieved with CMS. For 10 s testing time, the error rate dropped from 5.40% to 3.80% when using CMS. For the lowest testing time (3 s), the error rate of the AR-vector is close to 19% regardless of whether or not the normalization technique is used. Although there is a clear improvement in performance when using CMS, it is not of major significance. This leads to the conclusion that the AR-vector classification system is somewhat robust to channel distortion, especially as the testing time decreases.
{"title":"AR-vector using CMS for robust text independent speaker verification","authors":"C. B. D. Lima, Dirceu G. da Silva, A. Alcaim, J. A. Apolinário","doi":"10.1109/ICDSP.2002.1028276","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028276","url":null,"abstract":"This paper presents the performance of the AR-vector with cepstral mean subtraction (CMS) used to compensate the distortions caused by distinct telephone channels. The speaker recognition performance obtained with the use of CMS is compared with a system without compensation. With 60 s of speech signal used for training and 30 s used for testing, the error rate without channel normalization is around 2.82% against the 1.65% achieved with CMS. For 10 s testing time, the error rate dropped from 5.40% to 3.80% when using CMS. For the lowest testing time (3 s), the error rate of the AR-vector is close to 19% regardless of whether or not the normalization technique is used. Although there is a clear improvement in performance when using CMS, it is not of major significance. This leads to the conclusion that the AR-vector classification system is somewhat robust to channel distortion, especially as the testing time decreases.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131002667","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2002-11-07DOI: 10.1109/ICDSP.2002.1028271
N. Wong, T. Ng
This paper presents a generic, scalable approach to obtain closed-form state-trajectory expressions for high-order (order > 2) lowpass sigma-delta (/spl Sigma//spl Delta/) modulators with distinct noise transfer function (NTF) zeros. Constant modulator input is assumed. The techniques of state-space diagonalization, continuous-time embedding, and Poincare map analysis are combined and extended. It is shown that an even-order modulator can be decomposed into individual second-order subsystems with circular trajectories about two half-plane centers, while an odd-order modulator will result in an additional first-order subsystem represented by an oscillating quantity. The trajectory and half-plane transition expressions thus obtained provide effective tools for stability analysis of /spl Sigma//spl Delta/ modulators.
{"title":"State-trajectory behavior in high-order, lowpass sigma-delta modulators with distinct NTF zeros","authors":"N. Wong, T. Ng","doi":"10.1109/ICDSP.2002.1028271","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028271","url":null,"abstract":"This paper presents a generic, scalable approach to obtain closed-form state-trajectory expressions for high-order (order > 2) lowpass sigma-delta (/spl Sigma//spl Delta/) modulators with distinct noise transfer function (NTF) zeros. Constant modulator input is assumed. The techniques of state-space diagonalization, continuous-time embedding, and Poincare map analysis are combined and extended. It is shown that an even-order modulator can be decomposed into individual second-order subsystems with circular trajectories about two half-plane centers, while an odd-order modulator will result in an additional first-order subsystem represented by an oscillating quantity. The trajectory and half-plane transition expressions thus obtained provide effective tools for stability analysis of /spl Sigma//spl Delta/ modulators.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"148 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132086907","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2002-11-07DOI: 10.1109/ICDSP.2002.1028254
A. Kounoudes, P. Naylor, M. Brookes
This paper presents an automatic method to determine the instants of glottal closure (GCIs), or epochs, from the speech signal alone without the need of a laryngograph signal. The proposed algorithm incorporate a new technique for estimating GCI candidates and dynamic programming to select the best candidates according to predefined cost functions. Results show accuracy in estimation to within /spl plusmn/0.25ms on 87% of the test database and less that 1% false alarms and misses. Preliminary experiments using the telephone -degraded NTIMIT database have shown that the algorithm continues to perform well even in the presence of noise.
{"title":"Automatic epoch extraction for closed-phase analysis of speech","authors":"A. Kounoudes, P. Naylor, M. Brookes","doi":"10.1109/ICDSP.2002.1028254","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028254","url":null,"abstract":"This paper presents an automatic method to determine the instants of glottal closure (GCIs), or epochs, from the speech signal alone without the need of a laryngograph signal. The proposed algorithm incorporate a new technique for estimating GCI candidates and dynamic programming to select the best candidates according to predefined cost functions. Results show accuracy in estimation to within /spl plusmn/0.25ms on 87% of the test database and less that 1% false alarms and misses. Preliminary experiments using the telephone -degraded NTIMIT database have shown that the algorithm continues to perform well even in the presence of noise.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"14 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133137882","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2002-11-07DOI: 10.1109/ICDSP.2002.1028248
M. Sabry-Rizk, W. Zgallai
In this paper, we specifically turn our attention to long-term prediction of dynamic multi-fractal chaotic systems. Here, the linear, quadratic, cubic, and nth-order non-linearities are each multiplied by a weighting function. The weighting functions can take a time-varying form, if necessary, to cater for the non-stationary dynamics of the signal. During the training phase, the characteristic parameters of the weighting functions adapt to the varying nature and emphasis of non-linearity. Once the training of the new adaptive structure is completed; the generalization performance is evaluated by performing recursive prediction in an autonomous fashion. Specifically, the long-term predictive capability of the structure is tested by using a closed-loop adaptation scheme without any external input signal applied to the structure. The dynamic invariants computed from the reconstructed time series must now closely match the corresponding ones computed from the original time series. We will provide evidence of long-term prediction in excess of several thousand samples of highly complex (nine dimension) multi-fractal labour contraction signals using only a small fraction of this sample (only 300 samples for the training phase). Also presented are interesting results obtained using Lorenz attractor, and performing two recursive long-term predictions; (i) the regularized Gaussian radial basis function networks, and (ii) our novel embedded Volterra-like structure with weighted linear, quadratic and cubic nonlinearities, which demonstrate the superior performance of the latter with reduced SNRs.
{"title":"A new class of non-linear, multi-dimensional structures for long-term dynamic modelling of chaotic systems","authors":"M. Sabry-Rizk, W. Zgallai","doi":"10.1109/ICDSP.2002.1028248","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028248","url":null,"abstract":"In this paper, we specifically turn our attention to long-term prediction of dynamic multi-fractal chaotic systems. Here, the linear, quadratic, cubic, and nth-order non-linearities are each multiplied by a weighting function. The weighting functions can take a time-varying form, if necessary, to cater for the non-stationary dynamics of the signal. During the training phase, the characteristic parameters of the weighting functions adapt to the varying nature and emphasis of non-linearity. Once the training of the new adaptive structure is completed; the generalization performance is evaluated by performing recursive prediction in an autonomous fashion. Specifically, the long-term predictive capability of the structure is tested by using a closed-loop adaptation scheme without any external input signal applied to the structure. The dynamic invariants computed from the reconstructed time series must now closely match the corresponding ones computed from the original time series. We will provide evidence of long-term prediction in excess of several thousand samples of highly complex (nine dimension) multi-fractal labour contraction signals using only a small fraction of this sample (only 300 samples for the training phase). Also presented are interesting results obtained using Lorenz attractor, and performing two recursive long-term predictions; (i) the regularized Gaussian radial basis function networks, and (ii) our novel embedded Volterra-like structure with weighted linear, quadratic and cubic nonlinearities, which demonstrate the superior performance of the latter with reduced SNRs.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115998590","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2002-11-07DOI: 10.1109/ICDSP.2002.1028291
R. Verdú, J. Morales-Sánchez, R. Berenguer-Vidal, L. Weruaga
This paper shows a novel method related to motion compensated image reconstruction in computed tomography. The proposed method is based on deformable models. Computing the movement directly from the original non-compensated reconstruction avoiding artifacts selectively in a semiautomatic way is the first step in proposed motion compensation. The motion estimation provided by active contour models fitting and interpolation are used to reduce the image artifacts when it is applied during reconstruction. Preliminary results shows that the method produces images with a low-level artifacts for phantoms with synthetic motion.
{"title":"Active contours for heart motion-compensated reconstruction in computed tomography","authors":"R. Verdú, J. Morales-Sánchez, R. Berenguer-Vidal, L. Weruaga","doi":"10.1109/ICDSP.2002.1028291","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028291","url":null,"abstract":"This paper shows a novel method related to motion compensated image reconstruction in computed tomography. The proposed method is based on deformable models. Computing the movement directly from the original non-compensated reconstruction avoiding artifacts selectively in a semiautomatic way is the first step in proposed motion compensation. The motion estimation provided by active contour models fitting and interpolation are used to reduce the image artifacts when it is applied during reconstruction. Preliminary results shows that the method produces images with a low-level artifacts for phantoms with synthetic motion.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129770597","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2002-11-07DOI: 10.1109/ICDSP.2002.1028223
Anwar S. Dawood, John A. Williams, S. J. Visser
The High Performance Computing (HPC-I) payload is an innovative computing device designed for deployment on the Australian scientific mission satellite FedSat. HPC-I will validate and evaluate the practicality of using reconfigurable field programmable gate array (FPGA) technology in the space environment. The deployment of reconfigurable FPGA technology on-board satellites is a very promising solution for digital signal processing in the challenging space environment, offering tremendous flexibility to adapt to changing operation requirements, while achieving very high performance. Such combined flexibility and performance is not found in conventional signal processing architectures. This paper presents the design and implementation on HPC-I of two common digital signal filtering algorithms, a 4-tap low pass FIR filter and a 32-tap moving average filter. The flexibility and adaptability of the system is discussed in the context of more complex functionality and changing operation requirements.
{"title":"Flexible real time signal filtering in space using reconfigurable logic","authors":"Anwar S. Dawood, John A. Williams, S. J. Visser","doi":"10.1109/ICDSP.2002.1028223","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028223","url":null,"abstract":"The High Performance Computing (HPC-I) payload is an innovative computing device designed for deployment on the Australian scientific mission satellite FedSat. HPC-I will validate and evaluate the practicality of using reconfigurable field programmable gate array (FPGA) technology in the space environment. The deployment of reconfigurable FPGA technology on-board satellites is a very promising solution for digital signal processing in the challenging space environment, offering tremendous flexibility to adapt to changing operation requirements, while achieving very high performance. Such combined flexibility and performance is not found in conventional signal processing architectures. This paper presents the design and implementation on HPC-I of two common digital signal filtering algorithms, a 4-tap low pass FIR filter and a 32-tap moving average filter. The flexibility and adaptability of the system is discussed in the context of more complex functionality and changing operation requirements.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"55 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128263164","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2002-11-07DOI: 10.1109/ICDSP.2002.1028213
P. Pérez-Alcázar, Andrés Santos
Few works have been done about the dependency of the quantization noise with the sampling rate for uniform quantizers. Some of these works have considered the problem from a deterministic point of view while others study it from a stochastic one, having explained the noise behavior in some specific cases. By using computer simulations with a sinusoidal input signal, we show that the quantization noise spectrum can show a discrete or complex structure depending on the sampling rate used. The results confirm that an integer ratio between the sampling rate (f/sub r/) and the frequency of the input signal (f/sub s/) produces quantization noise with the components in odd harmonics of the signal frequency. If there is not an integer ratio between f/sub r/ and f/sub s/, then the quantization noise can present a stochastic structure for some rational ratios. An additional result is that the phase of the input signal can also modify the magnitude of the spectral components of the quantization noise. These results show that the quantization noise is clearly dependent on the input signal and the sampling rate.
{"title":"Relationship between sampling rate and quantization noise","authors":"P. Pérez-Alcázar, Andrés Santos","doi":"10.1109/ICDSP.2002.1028213","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028213","url":null,"abstract":"Few works have been done about the dependency of the quantization noise with the sampling rate for uniform quantizers. Some of these works have considered the problem from a deterministic point of view while others study it from a stochastic one, having explained the noise behavior in some specific cases. By using computer simulations with a sinusoidal input signal, we show that the quantization noise spectrum can show a discrete or complex structure depending on the sampling rate used. The results confirm that an integer ratio between the sampling rate (f/sub r/) and the frequency of the input signal (f/sub s/) produces quantization noise with the components in odd harmonics of the signal frequency. If there is not an integer ratio between f/sub r/ and f/sub s/, then the quantization noise can present a stochastic structure for some rational ratios. An additional result is that the phase of the input signal can also modify the magnitude of the spectral components of the quantization noise. These results show that the quantization noise is clearly dependent on the input signal and the sampling rate.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"51 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129816317","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2002-11-07DOI: 10.1109/ICDSP.2002.1028314
A. Torokhti, P. Howlett, C. Pearce
We provide a non-linear optimal physically realizable filter which guarantees a smaller associated error than those of the known linear optimal filters proposed by H.W. Bode and C.E. Shannon (see Proc. IRE, vol.38, p.417-25, 1950) and M.V. Ruzhansky and V.N. Fomin (see Bulletin of St. Petersburg University, Mathematics, vol.28, p.50-5, 1995). The technique presented has potential applications to numerous areas in signal processing including, for example, filtering, blind channel equalization, feature selection and classification in pattern recognition, target detection, etc. The technique is based on the best approximation of a stochastic signal by a specific non-linear operator acting on the noisy observed data.
{"title":"A causal optimal filter of the second order","authors":"A. Torokhti, P. Howlett, C. Pearce","doi":"10.1109/ICDSP.2002.1028314","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028314","url":null,"abstract":"We provide a non-linear optimal physically realizable filter which guarantees a smaller associated error than those of the known linear optimal filters proposed by H.W. Bode and C.E. Shannon (see Proc. IRE, vol.38, p.417-25, 1950) and M.V. Ruzhansky and V.N. Fomin (see Bulletin of St. Petersburg University, Mathematics, vol.28, p.50-5, 1995). The technique presented has potential applications to numerous areas in signal processing including, for example, filtering, blind channel equalization, feature selection and classification in pattern recognition, target detection, etc. The technique is based on the best approximation of a stochastic signal by a specific non-linear operator acting on the noisy observed data.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"os-3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127848160","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2002-11-07DOI: 10.1109/ICDSP.2002.1028220
Dae-young Jang, Jeongil Seo, Taejin Lee, Jin-Woo Hong, Kyeongok Kang
In this paper, we describe the implementation of the multi-channel AAC encoder system for digital audio broadcasting. The encoder system is based on MPEG-2/4 advanced audio coding (AAC) and is capable of real-time encoding up to 5.1 channel audio. To give a flexible functionality, it consists of multiple DSPs, IEC61937 and TCP/IP interface and 6 channel audio input facilities. The reference AAC decoder was implemented for verification test of the encoder. The encoder system is also integrated with the AAC streaming system for interoperation test. Through these tests, the encoder system is verified to be a good solution for high quality audio broadcasting.
{"title":"Implementation of multi-channel AAC encoder for high quality audio broadcasting","authors":"Dae-young Jang, Jeongil Seo, Taejin Lee, Jin-Woo Hong, Kyeongok Kang","doi":"10.1109/ICDSP.2002.1028220","DOIUrl":"https://doi.org/10.1109/ICDSP.2002.1028220","url":null,"abstract":"In this paper, we describe the implementation of the multi-channel AAC encoder system for digital audio broadcasting. The encoder system is based on MPEG-2/4 advanced audio coding (AAC) and is capable of real-time encoding up to 5.1 channel audio. To give a flexible functionality, it consists of multiple DSPs, IEC61937 and TCP/IP interface and 6 channel audio input facilities. The reference AAC decoder was implemented for verification test of the encoder. The encoder system is also integrated with the AAC streaming system for interoperation test. Through these tests, the encoder system is verified to be a good solution for high quality audio broadcasting.","PeriodicalId":351073,"journal":{"name":"2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628)","volume":"2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2002-11-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129240692","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
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