Current Issue : October - December Volume : 2011 Issue Number : 1 Articles : 5 Articles
Degrouping is the key component in MPEG Layer II audio decoding. It mainly contains the arithmetic operations of division and modulo. So far no dedicated degrouping algorithm and architecture is well realized. In the paper we propose a novel degrouping algorithm and its architecture design with low complexity design consideration. Our approach relies on only using the addition and subtraction instead of the division and modulo arithmetic operations. By use of this technique, it achieves the equivalent result without any loss of accuracy. The proposed design is without any multiplier, divider and ROM table and thus it can reduce the design complexity and chip area. In addition, it does not need any programming effort on numerical analysis. The result shows that it takes the advantages of simple and low cost design. Furthermore, it achieves high efficiency on fixed throughput with only one clock cycle per sample. The VLSI implementation result indicates the gate counts are only 527....
Organizing a database of user-contributed environmental sound recordings allows sound files to be linked not only by the semantic tags and labels applied to them, but also to other sounds with similar acoustic characteristics. Of paramount importance in navigating these databases are the problems of retrieving similar sounds using text- or sound-based queries, and automatically annotating unlabeled sounds. We propose an integrated system, which can be used for text-based retrieval of unlabeled audio, content-based query-by-example, and automatic annotation of unlabeled sound files. To this end, we introduce an ontological framework where sounds are connected to each other based on the similarity between acoustic features specifically adapted to environmental sounds, while semantic tags and sounds are connected through link weights that are optimized based on user-provided tags. Furthermore, tags are linked to each other through a measure of semantic similarity, which allows for efficient incorporation of out-of-vocabulary tags, that is, tags that do not yet exist in the database. Results on two freely available databases of environmental sounds contributed and labeled by nonexpert users demonstrate effective recall, precision, and average precision scores for both the text-based retrieval and annotation tasks....
In this paper we present a method to search for environmental sounds in large unstructured databases of user-submitted audio, using a general sound events taxonomy from ecological acoustics. We discuss the use of Support Vector Machines to classify sound recordings according to the taxonomy and describe two use cases for the obtained classification models: a content-based web search interface for a large audio database and a method for segmenting field recordings to assist sound design....
A method is described for quantifying the quality of wideband speech codecs. Two parameters are derived from signal-based speech quality model estimations: (i) a wideband equipment impairment factor Ie ,W B and (ii) a wideband packet-loss robustness factor Bp l,W B. The equipment impairment factor can be combined with impairment factors for other quality degradations to form an estimate of the overall conversational quality R of a wideband communication scenario, using a wideband extension of the E-model. The packet-loss robustness factor captures the robustness of the codec against packet-loss degradations. In contrast to past work, these parameters are no longer determined on the basis of auditory test results, but from signal-based speech quality models. We applied three intrusive models to several databases and compared the derived quality estimates and impairment factors to those obtained from auditory tests. The results show that when migrating from narrowband to wideband transmissionââ?¬â?a quality improvement of roughly 30% can be obtained, which is very similar to the one observed in auditory tests. The estimated impairment factors show a high correlation to those derived from auditory scores. Congruences and discrepancies to auditory test results are discussed, and an outline of work necessary to set up a wideband or even superwideband E-model is given....
Correlogram is an important representation for periodic signals. It is widely used in pitch estimation and source separation. For these applications, major problems of correlogram are its low resolution and redundant information. This paper proposes a voiced speech segregation system based on a newly introduced concept called dynamic harmonic function (DHF). In the proposed system, conventional correlograms are further processed by replacing the autocorrelation function (ACF) with DHF. The advantages of DHF are: 1) peak's width is adjustable by controlling the variance of the Gaussian function and 2) the invalid peaks of ACF, not at the pitch period, tend to be suppressed. Based on DHF, pitch detection and effective source segregation algorithms are proposed. Our system is systematically evaluated and compared with the correlogram-based system. Both the signal-to-noise ratio results and the perceptual evaluation of speech quality scores show that the proposed system yields substantially better performance....
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