Pub Date : 2013-12-01DOI: 10.1109/TASL.2013.2281570
Bruno Defraene, Naim Mansour, S. D. Hertogh, T. Waterschoot, M. Diehl, M. Moonen
The restoration of clipped audio signals, commonly known as declipping, is important to achieve an improved level of audio quality in many audio applications. In this paper, a novel declipping algorithm is presented, jointly based on the theory of compressed sensing (CS) and on well-established properties of human auditory perception. Declipping is formulated as a sparse signal recovery problem using the CS framework. By additionally exploiting knowledge of human auditory perception, a novel perceptual compressed sensing (PCS) framework is devised. A PCS-based declipping algorithm is proposed which uses $ell _{1}$-norm type reconstruction. Comparative objective and subjective evaluation experiments reveal a significant audio quality increase for the proposed PCS-based declipping algorithm compared to CS-based declipping algorithms.
在许多音频应用中，恢复被截断的音频信号(通常称为衰落)对于提高音频质量水平是很重要的。本文基于压缩感知(CS)理论和人类听觉感知特性，提出了一种新的衰落算法。使用CS框架将衰落表述为一个稀疏信号恢复问题。利用听觉感知知识，设计了一种新的感知压缩感知(PCS)框架。提出了一种基于pc的衰落算法，该算法采用$ well _{1}$范数类型重构。对比客观和主观评价实验表明，与基于cs的衰落算法相比，本文提出的基于pcs的衰落算法的音频质量有显著提高。
{"title":"Declipping of Audio Signals Using Perceptual Compressed Sensing","authors":"Bruno Defraene, Naim Mansour, S. D. Hertogh, T. Waterschoot, M. Diehl, M. Moonen","doi":"10.1109/TASL.2013.2281570","DOIUrl":"https://doi.org/10.1109/TASL.2013.2281570","url":null,"abstract":"The restoration of clipped audio signals, commonly known as declipping, is important to achieve an improved level of audio quality in many audio applications. In this paper, a novel declipping algorithm is presented, jointly based on the theory of compressed sensing (CS) and on well-established properties of human auditory perception. Declipping is formulated as a sparse signal recovery problem using the CS framework. By additionally exploiting knowledge of human auditory perception, a novel perceptual compressed sensing (PCS) framework is devised. A PCS-based declipping algorithm is proposed which uses $ell _{1}$-norm type reconstruction. Comparative objective and subjective evaluation experiments reveal a significant audio quality increase for the proposed PCS-based declipping algorithm compared to CS-based declipping algorithms.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2627-2637"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2281570","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892170","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 : 2013-12-01DOI: 10.1109/TASL.2013.2280218
Y. Ni, Matt McVicar, Raúl Santos-Rodríguez, T. D. Bie
To assess the performance of an automatic chord estimation system, reference annotations are indispensable. However, owing to the complexity of music and the sometimes ambiguous harmonic structure of polyphonic music, chord annotations are inherently subjective, and as a result any derived accuracy estimates will be subjective as well. In this paper, we investigate the extent of the confounding effect of subjectivity in reference annotations. Our results show that this effect is important, and they affect different types of automatic chord estimation systems in different ways. Our results have implications for research on automatic chord estimation, but also on other fields that evaluate performance by comparing against human provided annotations that are confounded by subjectivity.
{"title":"Understanding Effects of Subjectivity in Measuring Chord Estimation Accuracy","authors":"Y. Ni, Matt McVicar, Raúl Santos-Rodríguez, T. D. Bie","doi":"10.1109/TASL.2013.2280218","DOIUrl":"https://doi.org/10.1109/TASL.2013.2280218","url":null,"abstract":"To assess the performance of an automatic chord estimation system, reference annotations are indispensable. However, owing to the complexity of music and the sometimes ambiguous harmonic structure of polyphonic music, chord annotations are inherently subjective, and as a result any derived accuracy estimates will be subjective as well. In this paper, we investigate the extent of the confounding effect of subjectivity in reference annotations. Our results show that this effect is important, and they affect different types of automatic chord estimation systems in different ways. Our results have implications for research on automatic chord estimation, but also on other fields that evaluate performance by comparing against human provided annotations that are confounded by subjectivity.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"1 1","pages":"2607-2615"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2280218","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892616","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 : 2013-12-01DOI: 10.1109/TASL.2013.2279318
Katherine Ellis, E. Coviello, Antoni B. Chan, Gert R. G. Lanckriet
We present a content-based automatic tagging system for music that relies on a high-level, concise “Bag of Systems” (BoS) representation of the characteristics of a musical piece. The BoS representation leverages a rich dictionary of musical codewords, where each codeword is a generative model that captures timbral and temporal characteristics of music. Songs are represented as a BoS histogram over codewords, which allows for the use of traditional algorithms for text document retrieval to perform auto-tagging. Compared to estimating a single generative model to directly capture the musical characteristics of songs associated with a tag, the BoS approach offers the flexibility to combine different generative models at various time resolutions through the selection of the BoS codewords. Additionally, decoupling the modeling of audio characteristics from the modeling of tag-specific patterns makes BoS a more robust and rich representation of music. Experiments show that this leads to superior auto-tagging performance.
{"title":"A Bag of Systems Representation for Music Auto-Tagging","authors":"Katherine Ellis, E. Coviello, Antoni B. Chan, Gert R. G. Lanckriet","doi":"10.1109/TASL.2013.2279318","DOIUrl":"https://doi.org/10.1109/TASL.2013.2279318","url":null,"abstract":"We present a content-based automatic tagging system for music that relies on a high-level, concise “Bag of Systems” (BoS) representation of the characteristics of a musical piece. The BoS representation leverages a rich dictionary of musical codewords, where each codeword is a generative model that captures timbral and temporal characteristics of music. Songs are represented as a BoS histogram over codewords, which allows for the use of traditional algorithms for text document retrieval to perform auto-tagging. Compared to estimating a single generative model to directly capture the musical characteristics of songs associated with a tag, the BoS approach offers the flexibility to combine different generative models at various time resolutions through the selection of the BoS codewords. Additionally, decoupling the modeling of audio characteristics from the modeling of tag-specific patterns makes BoS a more robust and rich representation of music. Experiments show that this leads to superior auto-tagging performance.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2554-2569"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2279318","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892428","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 : 2013-12-01DOI: 10.1109/TASL.2013.2280210
O. Thiergart, G. D. Galdo, Maja Taseska, Emanuël Habets
Traditional spatial sound acquisition aims at capturing a sound field with multiple microphones such that at the reproduction side a listener can perceive the sound image as it was at the recording location. Standard techniques for spatial sound acquisition usually use spaced omnidirectional microphones or coincident directional microphones. Alternatively, microphone arrays and spatial filters can be used to capture the sound field. From a geometric point of view, the perspective of the sound field is fixed when using such techniques. In this paper, a geometry-based spatial sound acquisition technique is proposed to compute virtual microphone signals that manifest a different perspective of the sound field. The proposed technique uses a parametric sound field model that is formulated in the time-frequency domain. It is assumed that each time-frequency instant of a microphone signal can be decomposed into one direct and one diffuse sound component. It is further assumed that the direct component is the response of a single isotropic point-like source (IPLS) of which the position is estimated for each time-frequency instant using distributed microphone arrays. Given the sound components and the position of the IPLS, it is possible to synthesize a signal that corresponds to a virtual microphone at an arbitrary position and with an arbitrary pick-up pattern.
{"title":"Geometry-Based Spatial Sound Acquisition Using Distributed Microphone Arrays","authors":"O. Thiergart, G. D. Galdo, Maja Taseska, Emanuël Habets","doi":"10.1109/TASL.2013.2280210","DOIUrl":"https://doi.org/10.1109/TASL.2013.2280210","url":null,"abstract":"Traditional spatial sound acquisition aims at capturing a sound field with multiple microphones such that at the reproduction side a listener can perceive the sound image as it was at the recording location. Standard techniques for spatial sound acquisition usually use spaced omnidirectional microphones or coincident directional microphones. Alternatively, microphone arrays and spatial filters can be used to capture the sound field. From a geometric point of view, the perspective of the sound field is fixed when using such techniques. In this paper, a geometry-based spatial sound acquisition technique is proposed to compute virtual microphone signals that manifest a different perspective of the sound field. The proposed technique uses a parametric sound field model that is formulated in the time-frequency domain. It is assumed that each time-frequency instant of a microphone signal can be decomposed into one direct and one diffuse sound component. It is further assumed that the direct component is the response of a single isotropic point-like source (IPLS) of which the position is estimated for each time-frequency instant using distributed microphone arrays. Given the sound components and the position of the IPLS, it is possible to synthesize a signal that corresponds to a virtual microphone at an arbitrary position and with an arbitrary pick-up pattern.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2583-2594"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2280210","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892547","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 : 2013-12-01DOI: 10.1109/TASL.2013.2277941
Jean-Louis Durrieu, J. Thiran
Tracking vocal tract formant frequencies $(f_{p})$ and estimating the fundamental frequency $(f_{0})$ are two tracking problems that have been tackled in many speech processing works, often independently, with applications to articulatory parameters estimations, speech analysis/synthesis or linguistics. Many works assume an auto-regressive (AR) model to fit the spectral envelope, hence indirectly estimating the formant tracks from the AR parameters. However, directly estimating the formant frequencies, or equivalently the poles of the AR filter, allows to further model the smoothness of the desired tracks. In this paper, we propose a Factorial Hidden Markov Model combined with a vocal source/filter model, with parameters naturally encoding the $f_{0}$ and $f_{p}$ tracks. Two algorithms are proposed, with two different strategies: first, a simplification of the underlying model, with a parameter estimation based on variational methods, and second, a sparse decomposition of the signal, based on Non-negative Matrix Factorization methodology. The results are comparable to state-of-the-art formant tracking algorithms. With the use of a complete production model, the proposed systems provide robust formant tracks which can be used in various applications. The algorithms could also be extended to deal with multiple-speaker signals.
{"title":"Source/Filter Factorial Hidden Markov Model, With Application to Pitch and Formant Tracking","authors":"Jean-Louis Durrieu, J. Thiran","doi":"10.1109/TASL.2013.2277941","DOIUrl":"https://doi.org/10.1109/TASL.2013.2277941","url":null,"abstract":"Tracking vocal tract formant frequencies <formula formulatype=\"inline\"> <tex Notation=\"TeX\">$(f_{p})$</tex></formula> and estimating the fundamental frequency <formula formulatype=\"inline\"><tex Notation=\"TeX\">$(f_{0})$</tex> </formula> are two tracking problems that have been tackled in many speech processing works, often independently, with applications to articulatory parameters estimations, speech analysis/synthesis or linguistics. Many works assume an auto-regressive (AR) model to fit the spectral envelope, hence indirectly estimating the formant tracks from the AR parameters. However, directly estimating the formant frequencies, or equivalently the poles of the AR filter, allows to further model the smoothness of the desired tracks. In this paper, we propose a Factorial Hidden Markov Model combined with a vocal source/filter model, with parameters naturally encoding the <formula formulatype=\"inline\"><tex Notation=\"TeX\">$f_{0}$</tex></formula> and <formula formulatype=\"inline\"> <tex Notation=\"TeX\">$f_{p}$</tex></formula> tracks. Two algorithms are proposed, with two different strategies: first, a simplification of the underlying model, with a parameter estimation based on variational methods, and second, a sparse decomposition of the signal, based on Non-negative Matrix Factorization methodology. The results are comparable to state-of-the-art formant tracking algorithms. With the use of a complete production model, the proposed systems provide robust formant tracks which can be used in various applications. The algorithms could also be extended to deal with multiple-speaker signals.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2541-2553"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2277941","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892418","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 : 2013-12-01DOI: 10.1109/TASL.2013.2279338
A. D. Dileep, C. Sekhar
In this paper, we address the issues in the design of an intermediate matching kernel (IMK) for classification of sequential patterns using support vector machine (SVM) based classifier for tasks such as speech recognition. Specifically, we address the issues in constructing a kernel for matching sequences of feature vectors extracted from the speech signal data of utterances. The codebook based IMK and Gaussian mixture model (GMM) based IMK have been proposed earlier for matching the varying length patterns represented as sets of features vectors for tasks such as image classification and speaker recognition. These methods consider the centers of clusters and the components of GMM as the virtual feature vectors used in the design of IMK. As these methods do not use sequence information in matching the patterns, these methods are not suitable for matching sequential patterns. We propose the hidden Markov model (HMM) based IMK for matching sequential patterns of varying length. We consider two approaches to design the HMM-based IMK. In the first approach, each of the two sequences to be matched is segmented into subsequences with each subsequence aligned to a state of the HMM. Then the HMM-based IMK is constructed as a combination of state-specific GMM-based IMKs that match the subsequences aligned with the particular states of the HMM. In the second approach, the HMM-based IMK is constructed without segmenting sequences, and by matching the local feature vectors selected using the responsibility terms that account for being in a state and generating the feature vectors by a component of the GMM of that state. We study the performance of the SVM based classifiers using the proposed HMM-based IMK for recognition of isolated utterances of E-set in English alphabet and recognition of consonent–vowel segments in Hindi language.
{"title":"HMM Based Intermediate Matching Kernel for Classification of Sequential Patterns of Speech Using Support Vector Machines","authors":"A. D. Dileep, C. Sekhar","doi":"10.1109/TASL.2013.2279338","DOIUrl":"https://doi.org/10.1109/TASL.2013.2279338","url":null,"abstract":"In this paper, we address the issues in the design of an intermediate matching kernel (IMK) for classification of sequential patterns using support vector machine (SVM) based classifier for tasks such as speech recognition. Specifically, we address the issues in constructing a kernel for matching sequences of feature vectors extracted from the speech signal data of utterances. The codebook based IMK and Gaussian mixture model (GMM) based IMK have been proposed earlier for matching the varying length patterns represented as sets of features vectors for tasks such as image classification and speaker recognition. These methods consider the centers of clusters and the components of GMM as the virtual feature vectors used in the design of IMK. As these methods do not use sequence information in matching the patterns, these methods are not suitable for matching sequential patterns. We propose the hidden Markov model (HMM) based IMK for matching sequential patterns of varying length. We consider two approaches to design the HMM-based IMK. In the first approach, each of the two sequences to be matched is segmented into subsequences with each subsequence aligned to a state of the HMM. Then the HMM-based IMK is constructed as a combination of state-specific GMM-based IMKs that match the subsequences aligned with the particular states of the HMM. In the second approach, the HMM-based IMK is constructed without segmenting sequences, and by matching the local feature vectors selected using the responsibility terms that account for being in a state and generating the feature vectors by a component of the GMM of that state. We study the performance of the SVM based classifiers using the proposed HMM-based IMK for recognition of isolated utterances of E-set in English alphabet and recognition of consonent–vowel segments in Hindi language.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2570-2582"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2279338","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892483","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 : 2013-11-01DOI: 10.1109/TASL.2013.2273715
Min Zhang, Wenliang Chen, Xiangyu Duan, Rong Zhang
For graph-based dependency parsing, how to enrich high-order features without increasing decoding complexity is a very challenging problem. To solve this problem, this paper presents an approach to representing high-order features for graph-based dependency parsing models using a dependency language model and beam search. Firstly, we use a baseline parser to parse a large-amount of unannotated data. Then we build the dependency language model (DLM) on the auto-parsed data. A set of new features is represented based on the DLM. Finally, we integrate the DLM-based features into the parsing model during decoding by beam search. We also utilize the features in bilingual text (bitext) parsing models. The main advantages of our approach are: 1) we utilize rich high-order features defined over a view of large scope and additional large raw corpus; 2) our approach does not increase the decoding complexity. We evaluate the proposed approach on the monotext and bitext parsing tasks. In the monotext parsing task, we conduct the experiments on Chinese and English data. The experimental results show that our new parser achieves the best accuracy on the Chinese data and comparable accuracy with the best known systems on the English data. In the bitext parsing task, we conduct the experiments on a Chinese-English bilingual data and our score is the best reported so far.
{"title":"Improving Graph-Based Dependency Parsing Models With Dependency Language Models","authors":"Min Zhang, Wenliang Chen, Xiangyu Duan, Rong Zhang","doi":"10.1109/TASL.2013.2273715","DOIUrl":"https://doi.org/10.1109/TASL.2013.2273715","url":null,"abstract":"For graph-based dependency parsing, how to enrich high-order features without increasing decoding complexity is a very challenging problem. To solve this problem, this paper presents an approach to representing high-order features for graph-based dependency parsing models using a dependency language model and beam search. Firstly, we use a baseline parser to parse a large-amount of unannotated data. Then we build the dependency language model (DLM) on the auto-parsed data. A set of new features is represented based on the DLM. Finally, we integrate the DLM-based features into the parsing model during decoding by beam search. We also utilize the features in bilingual text (bitext) parsing models. The main advantages of our approach are: 1) we utilize rich high-order features defined over a view of large scope and additional large raw corpus; 2) our approach does not increase the decoding complexity. We evaluate the proposed approach on the monotext and bitext parsing tasks. In the monotext parsing task, we conduct the experiments on Chinese and English data. The experimental results show that our new parser achieves the best accuracy on the Chinese data and comparable accuracy with the best known systems on the English data. In the bitext parsing task, we conduct the experiments on a Chinese-English bilingual data and our score is the best reported so far.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"14 1","pages":"2313-2323"},"PeriodicalIF":0.0,"publicationDate":"2013-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2273715","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62891502","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 : 2013-11-01DOI: 10.1109/TASL.2013.2280209
Fabian Triefenbach, A. Jalalvand, Kris Demuynck, J. Martens
Accurate acoustic modeling is an essential requirement of a state-of-the-art continuous speech recognizer. The Acoustic Model (AM) describes the relation between the observed speech signal and the non-observable sequence of phonetic units uttered by the speaker. Nowadays, most recognizers use Hidden Markov Models (HMMs) in combination with Gaussian Mixture Models (GMMs) to model the acoustics, but neural-based architectures are on the rise again. In this work, the recently introduced Reservoir Computing (RC) paradigm is used for acoustic modeling. A reservoir is a fixed - and thus non-trained - Recurrent Neural Network (RNN) that is combined with a trained linear model. This approach combines the ability of an RNN to model the recent past of the input sequence with a simple and reliable training procedure. It is shown here that simple reservoir-based AMs achieve reasonable phone recognition and that deep hierarchical and bi-directional reservoir architectures lead to a very competitive Phone Error Rate (PER) of 23.1% on the well-known TIMIT task.
{"title":"Acoustic Modeling With Hierarchical Reservoirs","authors":"Fabian Triefenbach, A. Jalalvand, Kris Demuynck, J. Martens","doi":"10.1109/TASL.2013.2280209","DOIUrl":"https://doi.org/10.1109/TASL.2013.2280209","url":null,"abstract":"Accurate acoustic modeling is an essential requirement of a state-of-the-art continuous speech recognizer. The Acoustic Model (AM) describes the relation between the observed speech signal and the non-observable sequence of phonetic units uttered by the speaker. Nowadays, most recognizers use Hidden Markov Models (HMMs) in combination with Gaussian Mixture Models (GMMs) to model the acoustics, but neural-based architectures are on the rise again. In this work, the recently introduced Reservoir Computing (RC) paradigm is used for acoustic modeling. A reservoir is a fixed - and thus non-trained - Recurrent Neural Network (RNN) that is combined with a trained linear model. This approach combines the ability of an RNN to model the recent past of the input sequence with a simple and reliable training procedure. It is shown here that simple reservoir-based AMs achieve reasonable phone recognition and that deep hierarchical and bi-directional reservoir architectures lead to a very competitive Phone Error Rate (PER) of 23.1% on the well-known TIMIT task.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"156 1","pages":"2439-2450"},"PeriodicalIF":0.0,"publicationDate":"2013-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2280209","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892491","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 : 2013-11-01DOI: 10.1109/TASL.2013.2273716
Florian Pflug, T. Fingscheidt
Applications such as professional wireless digital microphones require a transmission of practically uncoded high-quality audio with ultra-low latency on the one hand and robustness to error-prone channels on the other hand. The delay restrictions, however, prohibit the utilization of efficient block or convolutional channel codes for error protection. The contribution of this work is fourfold: We revise and summarize concisely a Bayesian framework for soft-decision audio decoding and present three novel approaches to (almost) latency-free robust decoding of uncompressed audio. Bit reliability information from the transmission channel is exploited, as well as short-term and long-term residual redundancy within the audio signal, and optionally some explicit redundancy in terms of a sample-individual block code. In all cases we utilize variants of higher-order linear prediction to compute prediction probabilities in three novel ways: Firstly by employing a serial cascade of multiple predictors, secondly by exploiting explicit redundancy in form of parity bits, and thirdly by utilizing an interpolative forward/backward prediction algorithm. The first two presented approaches work fully delayless, while the third one introduces an ultra-low algorithmic delay of just a few samples. The effectiveness of the proposed algorithms is proven in simulations with BPSK and typical digital microphone FSK modulation schemes on AWGN and bursty fading channels.
{"title":"Robust Ultra-Low Latency Soft-Decision Decoding of Linear PCM Audio","authors":"Florian Pflug, T. Fingscheidt","doi":"10.1109/TASL.2013.2273716","DOIUrl":"https://doi.org/10.1109/TASL.2013.2273716","url":null,"abstract":"Applications such as professional wireless digital microphones require a transmission of practically uncoded high-quality audio with ultra-low latency on the one hand and robustness to error-prone channels on the other hand. The delay restrictions, however, prohibit the utilization of efficient block or convolutional channel codes for error protection. The contribution of this work is fourfold: We revise and summarize concisely a Bayesian framework for soft-decision audio decoding and present three novel approaches to (almost) latency-free robust decoding of uncompressed audio. Bit reliability information from the transmission channel is exploited, as well as short-term and long-term residual redundancy within the audio signal, and optionally some explicit redundancy in terms of a sample-individual block code. In all cases we utilize variants of higher-order linear prediction to compute prediction probabilities in three novel ways: Firstly by employing a serial cascade of multiple predictors, secondly by exploiting explicit redundancy in form of parity bits, and thirdly by utilizing an interpolative forward/backward prediction algorithm. The first two presented approaches work fully delayless, while the third one introduces an ultra-low algorithmic delay of just a few samples. The effectiveness of the proposed algorithms is proven in simulations with BPSK and typical digital microphone FSK modulation schemes on AWGN and bursty fading channels.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2324-2336"},"PeriodicalIF":0.0,"publicationDate":"2013-11-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2273716","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62891352","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 : 2013-11-01DOI: 10.1109/TASL.2013.2274694
Koji Seto, T. Ogunfunmi
High quality speech at low bit rates makes code excited linear prediction (CELP) the dominant choice for a narrowband coding technique despite the susceptibility to packet loss. One of the few techniques which received attention after the introduction of CELP coding technique is the internet low bitrate codec (iLBC) because of inherent high robustness to packet loss. Addition of rate flexibility and scalability makes the iLBC an attractive choice for voice communication over IP networks. In this paper, performance improvement schemes of multi-rate iLBC and its scalable structure are proposed, and the proposed codec enhanced from the previous work is re-designed based on the subjective listening quality instead of the objective quality. In particular, perceptual weighting and the modified discrete cosine transform (MDCT) with short overlap in weighted signal domain are employed along with the improved packet loss concealment (PLC) algorithm. The subjective evaluation results show that the speech quality of the proposed codec is equivalent to that of state-of-the-art codec, G.718, under both a clean channel condition and lossy channel conditions. This result is significant considering that development of the proposed codec is still in early stage.
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