The Infona portal uses cookies, i.e. strings of text saved by a browser on the user's device. The portal can access those files and use them to remember the user's data, such as their chosen settings (screen view, interface language, etc.), or their login data. By using the Infona portal the user accepts automatic saving and using this information for portal operation purposes. More information on the subject can be found in the Privacy Policy and Terms of Service. By closing this window the user confirms that they have read the information on cookie usage, and they accept the privacy policy and the way cookies are used by the portal. You can change the cookie settings in your browser.
For mobile communication systems computational complexity and memory requirements are serious problems in real-time digital signal processing of speech signal. In this article we proposed new structuralization algorithm intended to split vector quantizer codebook of LSF coefficients. Fast search procedure, based on structure of codebook and description tree, allows reduce the entire quantity of comparisons...
The analysis of three sets of feature vectors used in speaker identification (ID) systems for speech signals received in encoding-decoding process with AMR, SPEEX and MELP coders has been presented. We have analyzed feature sets for various speech coding bit rates using SVM-based speaker ID system. The results were compared with identification accuracy obtained with vectors where fundamental frequency...
Deep analysis of Polish phonology and phonetic can help in efficient automatic speech recognition of fluent speech.The goal of this paper is to present some techniques used in words boundary detection algorithm in stream of phonemes. This new idea of ASR of fluent speech will be presented. The results of the experiment on the word with two-phoneme occurrences will be presented in this paper.
The paper describes an algorithm for estimating the pitch and formantspsila center frequencies and bandwidths with the use of instantaneous complex frequency (ICF). ICF for each formant is computed from the analytic signal, obtained by filtering speech signal with a passband complex Hilbert filter, which additionally separates the analyzed formant from the speech signal.
In this paper we present our investigations on statistical classification of acoustic signals which is one special assignment in condition monitoring. We compare three pattern recognition methods and three selected feature extraction algorithms with regard to their capability to distinguish between structure-borne sound signatures emitted by intact and worn-out rollers in a drawframe. Our goal is...
An important feature of an iterative SVD algorithm is presented, which, under certain circumstances, makes real-time calculations of approximate independent components possible. Presented experiments show that for mixtures of signals with low SNR, the algorithm is an efficient ICA solution with low computational cost and acceptable quality.
This paper presents results of our research on automatic telephone call speaker recognition. A significant improvement of recognition quality can be observed if we use the knowledge on how the original speech was coded, e.g., using GSM or PCM telephone standard. On this basis we can improve the quality of the speaker models. In order to realize such a speech analyzer we propose to analyze the mean...
This paper presents a frame-based method for estimation of instantaneous amplitudes and frequencies of pitch harmonics. Proposed method assumes that the speech signal is a combination of voiced (harmonic) and unvoiced (noise) components present in whole speech band. Harmonic transform is used as a frequency analysis tool which kernel is synchronized with time-varying pitch frequency. The speech signal...
This paper describes an implementation of a long distance echo canceller, operating on full-duplex with hands-free and in real-time with a single Digital Signal Processor (DSP). The proposed solution is based on short length adaptive filters centered on the positions of the most significant echoes, which are tracked by time delay estimators, for which we use a new approach. To deal with double talking...
Fuzzy extreme analysis (FEA) is expedient for those cases when it is necessary to analyze the most important details of a shape of a signal or its spectrum as well as to find out its envelope. It is based on a capture of only main signal extremes and intervals between them on the basis of comparing of the signal increases with assigned limits. Apart from it, FEA allows to significantly compress of...
The use of Mel-frequency cepstral coefficients (MFCCs) is well established in the fields of speech processing, particularly for speaker modeling within a Gaussian mixture model (GMM) speaker recognition system. The use of GMMs for speech enhancement applications has only recently been proposed in the literature; the concept of direct inversion of the MFCCs, however, has not been studied. In this paper...
Set the date range to filter the displayed results. You can set a starting date, ending date or both. You can enter the dates manually or choose them from the calendar.