Class / Patent application number | Description | Number of patent applications / Date published |
704204000 | Orthogonal functions | 15 |
20090287478 | Speech post-processing using MDCT coefficients - There is provided a speech post-processor for enhancing a speech signal divided into a plurality of sub-bands in frequency domain. The speech post-processor comprises an envelope modification factor generator configured to use frequency domain coefficients representative of an envelope derived from the plurality of sub-bands to generate an envelope modification factor for the envelope derived from the plurality of sub-bands, where the envelope modification factor is generated using FAC=αENV/Max+(1−α), where FAC is the envelope modification factor, ENV is the envelope, Max is the maximum envelope, and a is a value between 0 and 1, where α is a different constant value for each speech coding rate. The speech post-processor further comprises an envelope modifier configured to modify the envelope derived from the plurality of sub-bands by the envelope modification factor corresponding to each of the plurality of sub-bands. | 11-19-2009 |
20120296640 | ENCODING DEVICE AND ENCODING METHOD - Disclosed are an encoding device and encoding method capable of improving the quality of a decoded signal under very low bit rate conditions using a small amount of computation. A spectrum correction unit ( | 11-22-2012 |
20130006618 | SPEECH PROCESSING APPARATUS, SPEECH PROCESSING METHOD AND PROGRAM - The present invention relates to a speech processing apparatus, a speech processing method and a program which, when multichannel audio signals are downmixed and coded, prevent delay and an increase in the computation amount upon decoding of the audio signals. An inverse multiplexing unit ( | 01-03-2013 |
20130253920 | METHOD AND APPARATUS FOR ROBUST SPEAKER AND SPEECH RECOGNITION - A method of processing a speech signal comprises converting the speech signal to digital signals, converting the digital speech signal into short-time frames, applying a Fast Fourier Transform to each of the short-time frames to obtain an original spectrum, deriving a varied spectrum based on the original spectrum, applying discrete cosine transform to compute original cepstrum coefficients for the original spectrum and varied cepstrum coefficients for the varied spectrum and generating a set of frontend feature vectors for each of the short-time frames. | 09-26-2013 |
20130262097 | SYSTEMS AND METHODS FOR AUTOMATED SPEECH AND SPEAKER CHARACTERIZATION - Systems and methods utilize individually selected modulation spectral features for speech and speaker characterization. The method involves construction of a sparse feature space and a method of finding the approximately best feature subset for attributing a specific characteristic of speech or speaker. The current selection method is based on the Kolmogorov-Smirnov statistical test applied to individual features. The characterization task can be defined empirically and no a-priori theory is necessary to explain characteristic attribution processes. Experimental results indicate that employment of selected modulation spectral features works better than the current state-of-the-art at least in some instances of speech characterization task, e.g. prediction of speaker personality traits, as it is evident from the official results of Interspeech'2012 Speaker Personality Recognition Challenge. | 10-03-2013 |
20130297297 | SYSTEM AND METHOD FOR CLASSIFICATION OF EMOTION IN HUMAN SPEECH - A system performs local feature extraction. The system includes a processing device that performs a Short Time Fourier Transform to obtain a spectrogram for a discrete-time speech signal sample. The spectrogram is subdivided based on natural divisions of frequency to humans. Time-frequency-energy is then quantized using information obtained from the spectrogram. And, feature vectors are determined based on the quantized time-frequency-energy information. | 11-07-2013 |
20140337017 | Method for Converting Speech Using Sparsity Constraints - A method converts source speech to target speech by first mapping the source speech to sparse weights using compressive sensing technique, and the transforming, using transformation parameters, the sparse weights to the target speech. | 11-13-2014 |
20150066487 | VOICE PROCESSING APPARATUS AND VOICE PROCESSING METHOD - A voice processing apparatus includes: a dividing unit which divides a voice signal into frames in such a manner that any two successive frames overlap each other by a predetermined amount; a first windowing unit which multiplies each frame by a first windowing function that attenuates a signal at both ends of the frame; an orthogonal transform unit which computes a frequency spectrum for each frame multiplied by the first windowing function; a frequency signal processing unit which computes a corrected frequency spectrum; an inverse orthogonal transform unit which computes a corrected frame by applying an inverse orthogonal transform to the corrected frequency spectrum; a second windowing unit which multiplies each corrected frame by a second windowing function that attenuates a signal at both ends of the corrected frame; and an addition unit which adds up the each corrected frame multiplied by the second windowing function, sequentially in time order. | 03-05-2015 |
20150302864 | SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD - A signal processing apparatus generates a window signal, transforms the window signal into a frequency spectrum, and adjusts an amplitude component of the frequency spectrum. Then, the signal processing apparatus applies inverse transform to the amplitude component after adjustment and to a phase component of the frequency spectrum to generate a frame signal, and identifies an overlap segment such that the absolute value of the amplitude of the frame signal at at least one end of the overlap segment becomes smaller than the absolute value of the amplitude of the frame signal at a corresponding end of an overlapping section. Then, in the identified segment, the signal processing apparatus adds and compounds the frame signal corresponding to an immediately preceding frame and the frame signal corresponding to a processing-target frame. | 10-22-2015 |
20160049156 | METHOD FOR CODING PULSE VECTORS USING STATISTICAL PROPERTIES - Improved methods for coding an ensemble of pulse vectors utilize statistical models (i.e., probability models) for the ensemble of pulse vectors, to more efficiently code each pulse vector of the ensemble. At least one pulse parameter describing the non-zero pulses of a given pulse vector is coded using the statistical models and the number of non-zero pulse positions for the given pulse vector. In some embodiments, the number of non-zero pulse positions are coded using range coding. The total number of unit magnitude pulses may be coded using conditional (state driven) bitwise arithmetic coding. The non-zero pulse position locations may be coded using adaptive arithmetic coding. The non-zero pulse position magnitudes may be coded using probability-based combinatorial coding, and the corresponding sign information may be coded using bitwise arithmetic coding. Such methods are well suited to coding non-independent-identically-distributed signals, such as coding video information. | 02-18-2016 |
20160104488 | APPARATUS AND METHOD FOR IMPROVED SIGNAL FADE OUT FOR SWITCHED AUDIO CODING SYSTEMS DURING ERROR CONCEALMENT - An apparatus for decoding an audio signal includes a receiving interface, wherein the receiving interface is configured to receive a first frame and a second frame. Moreover, the apparatus includes a noise level tracing unit for determining noise level information being represented in a tracing domain. Furthermore, the apparatus includes a first reconstruction unit for reconstructing a third audio signal portion of the audio signal depending on the noise level information and a second reconstruction unit for reconstructing a fourth audio signal portion depending on noise level information being represented in the second reconstruction domain. | 04-14-2016 |
20160125893 | METHOD FOR AUDIO SOURCE SEPARATION AND CORRESPONDING APPARATUS - Separation of speech and background from an audio mixture by using a speech example, generated from a source associated with a speech component in the audio mixture, to guide the separation process. | 05-05-2016 |
20160155447 | Bitstream Syntax for Spatial Voice Coding | 06-02-2016 |
20160155448 | ENHANCED SOUND FIELD CODING USING PARAMETRIC COMPONENT GENERATION | 06-02-2016 |
20160163335 | METHOD AND DEVICE FOR PROCESSING A SOUND SIGNAL - A method of processing a sound signal is disclosed. The method of processing a sound signal includes receiving a sound signal from the outside of a device, converting the sound signal into a first frequency domain signal, determining whether or not the sound signal is a voice signal using the first frequency domain signal acquired through the conversion, converting the first frequency domain signal into a second frequency domain signal based on the determination, and recognizing the sound signal using the second frequency domain signal acquired through the conversion. | 06-09-2016 |