Class / Patent application number | Description | Number of patent applications / Date published |
704224000 | Normalizing | 18 |
20080235010 | Reproducing Apparatus - Disclosed is a reproducing apparatus comprising: a reproduction section to reproduce reproduction data comprising sound data and/or image data; a selection section to calculate evaluation values between a link source set for the reproduction data and each of a plurality of link destinations corresponding to the link source by a predetermined arithmetic expression based on link information of the plurality of link destinations, and to select a link destination having a highest evaluation among the evaluation values out of the plurality of link destinations; and a reproduction control section to move a reproduction point of the reproduction data reproduced by the reproduction section to a position corresponding to the link destination by linking the link source with the link destination when the reproduction point reaches a given point with respect to a position corresponding to the link source, and to instruct the reproduction section to reproduce the reproduction data. | 09-25-2008 |
20090281800 | SPECTRAL SHAPING FOR SPEECH INTELLIGIBILITY ENHANCEMENT - A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal. | 11-12-2009 |
20100030555 | CLIPPING DETECTION DEVICE AND METHOD - A clipping detection device calculates an amplitude distribution of an input signal for each predetermined period, calculates a deflection degree of the distribution on the basis of the calculated amplitude distribution, and then detects clipping of a communication signal on the basis of the calculated deflection degree of the distribution. | 02-04-2010 |
20100094622 | FEATURE NORMALIZATION FOR SPEECH AND AUDIO PROCESSING - Systems, method, and apparatus for processing a speech utterance or audio record that includes receiving one or more feature vectors characterizing the speech utterance or audio record, each feature vector having a plurality of feature elements, each feature element being associated with a spectral representation of a characteristic of one of a plurality of sequential segments of the speech utterance or audio record; and processing the one or more feature vectors in a rank order filter to obtain one or more normalized feature vectors, each normalized feature vector having a plurality of normalized feature elements corresponding to the plurality of feature elements. | 04-15-2010 |
20100106494 | Signal Processing Apparatus and Method, and Program - A coded code string from an input terminal | 04-29-2010 |
20100121634 | Speech Enhancement in Entertainment Audio - The invention relates to audio signal processing. More specifically, the invention relates to enhancing entertainment audio, such as television audio, to improve the clarity and intelligibility of speech, such as dialog and narrative audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods. | 05-13-2010 |
20100145692 | METHODS AND ARRANGEMENTS IN A TELECOMMUNICATIONS NETWORK - The present invention relates to a postfilter and a postfilter control to be associated with a postfilter for improving perceived quality of speech reconstructed at a speech decoder. The postfilter control comprises means for measuring stationarity of a speech signal reconstructed at a decoder, means for determining a coefficient to a postfilter control parameter based on the measured stationarity, and means for transmitting the determined coefficient to a postfilter, such that the postfilter can process the reconstructed speech signal by applying the determined coefficient to the postfilter control parameter to obtain an enhanced speech signal. | 06-10-2010 |
20100174540 | Time-Varying Audio-Signal Level Using a Time-Varying Estimated Probability Density of the Level - Methods, media and apparatus for smoothing a time-varying level of a signal. A method includes estimating a time-varying probability density of a short-term level of the signal and smoothing a level of the signal by using the probability density. The signal may be an audio signal. The short-term level and the smoothed level may be time series, each having current and previous time indices. Here, before the smoothing, computing a probability of the smoothed level at the previous time index may occur. Before the smoothing, calculating smoothing parameters using the probability density may occur. Calculating the smoothing parameters may include calculating the smoothing parameters using the smoothed level at the previous time index, the short-term level at the current time index and the probability of the smoothed level at the previous time index. Calculating the smoothing parameters may include calculating the smoothing parameters using breadth of the estimated probability density. | 07-08-2010 |
20120232889 | METHOD AND APPARATUS FOR PERFORMING PACKET LOSS OR FRAME ERASURE CONCEALMENT - A method for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder receives encoded frames of compressed speech information transmitted from an encoder. The method determines whether an encoded frame has been lost, corrupted in transmission, or erased, synthesizes properly received frames, and decides on an overlap-add window to use in combining a portion of the synthesized speech signal with a subsequent speech signal resulting from a received and decoded packet, where the size of the overlap-add window is based on the unavailability of packets. If it is determined that an encoded frame has been lost, corrupted in transmission, or erased, the method performed an overlap-add operation on the portion of the synthesized speech signal and the subsequent speech signal, using the decided-on overlap-add window. | 09-13-2012 |
20120323570 | RECONSTRUCTION OF A SMOOTH SPEECH SIGNAL FROM A STUTTERED SPEECH SIGNAL - Described herein are methods, systems, apparatuses and products for reconstruction of a smooth speech signal from a stuttered speech signal. One aspect provides for accessing a stored speech signal having stuttering; identifying at least one stuttered region in the stored speech signal; modifying the at least one stuttered region in the stored speech signal; and responsive to modifying the at least one stuttered region, reconstructing a smooth speech signal corresponding to the stored speech signal. Other embodiments are disclosed. | 12-20-2012 |
20130226571 | METHOD AND APPARATUS FOR PERFORMING PACKET LOSS OR FRAME ERASURE CONCEALMENT - A method for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder receives encoded frames of compressed speech information transmitted from an encoder. The method determines whether an encoded frame has been lost, corrupted in transmission, or erased, synthesizes properly received frames, and decides on an overlap-add window to use in combining a portion of the synthesized speech signal with a subsequent speech signal resulting from a received and decoded packet, where the size of the overlap-add window is based on the unavailability of packets. If it is determined that an encoded frame has been lost, corrupted in transmission, or erased, the method performed an overlap-add operation on the portion of the synthesized speech signal and the subsequent speech signal, using the decided-on overlap-add window. | 08-29-2013 |
20140278383 | APPARATUSES AND METHODS FOR MULTI-CHANNEL SIGNAL COMPRESSION DURING DESIRED VOICE ACTIVITY DETECTION - Systems and methods are described to create a desired voice activity detection signal. A main acoustic signal and a plurality of reference acoustic signals are compressed. The compressed main acoustic signal is normalized by the plurality of compressed reference acoustic signals to create a plurality of normalized compressed main acoustic signals. The plurality of normalized compressed main acoustic signals is processed with a plurality of single channel normalized voice threshold comparators to form a plurality of normalized desired voice activity detection signals. One of the plurality of normalized desired voice activity detection signals is selected from the plurality of normalized desired voice activity detection signals to output as the desired voice activity detection signal. | 09-18-2014 |
20140278384 | APPARATUSES AND METHODS FOR ACOUSTIC CHANNEL AUTO-BALANCING DURING MULTI-CHANNEL SIGNAL EXTRACTION - Systems and methods are described to automatically balance acoustic channel sensitivity. A long-term power level of a main acoustic signal is calculated to obtain an averaged main acoustic signal. Segments of the main acoustic signal are excluded from the averaged main acoustic signal using a desired voice activity detection signal. A long-term power level of a reference acoustic signal is calculated to obtain an averaged reference acoustic signal. Segments of the reference acoustic signal are excluded from the averaged reference acoustic signal using a desired voice activity detection signal. An amplitude correction signal is created using the averaged main acoustic signal and the averaged reference acoustic signal. | 09-18-2014 |
20150066494 | SMART CIRCULAR AUDIO BUFFER - An audio buffer is used to capture audio in anticipation of a user command to do so. Sensors and processor activity may be monitored, looking for indicia suggesting that the user command may be forthcoming. Upon detecting such indicia, a circular buffer is activated. Audio correction may be applied to the audio stored in the circular buffer. After receiving the user command instructing the device to process or record audio, at least a portion of the audio that was stored in the buffer before the command is combined with audio received after the command. The combined audio may then be processed, transmitted or stored. | 03-05-2015 |
20150302862 | ADAPTIVE EQUALIZATION SYSTEM - An adaptive equalization system that adjusts the spectral shape of a speech signal based on an intelligibility measurement of the speech signal may improve the intelligibility of the output speech signal. Such an adaptive equalization system may include a speech intelligibility measurement module, a spectral shape adjustment module, and an adaptive equalization module. The speech intelligibility measurement module is configured to calculate a speech intelligibility measurement of a speech signal. The spectral shape adjustment module is configured to generate a weighted long-term speech curve based on a first predetermined long-term average speech curve, a second predetermined long-term average speech curve, and the speech intelligibility measurement. The adaptive equalization module is configured to adapt equalization coefficients for the speech signal based on the weighted long-term speech curve. | 10-22-2015 |
20160019912 | VOICE SIGNAL MODULATION SERVICE FOR GEOGRAPHIC AREAS - Modulating a voice signal is provided. The voice signal corresponding to a voice communication is received from a sending voice communication device via a network. Voice signal features corresponding to the voice communication are extracted. A set of voice signal filters are selected to modulate the extracted voice signal features corresponding to the voice communication to an average voice signal associated with a geographic area where the voice communication is destined for. The voice signal features corresponding to the voice communication are modulated by applying the selected set of voice signal filters to generate the average voice signal associated with the geographic area where the voice communication is destined for. | 01-21-2016 |
20160189729 | APPARATUSES AND METHODS FOR MULTI-CHANNEL SIGNAL COMPRESSION DURING DESIRED VOICE ACTIVITY DETECTION - Apparatuses and methods are described to identify desired audio. A first input of an apparatus is configured to receive a main signal. A second input of the apparatus is configured to receive a reference signal. A normalizer is configured to normalize a compressed main signal by a compressed reference signal to create a normalized main signal. A single channel normalized voice threshold comparator is configured to receive as an input the normalized main signal and to output a desired voice activity detection signal. | 06-30-2016 |
20160203123 | COGNITIVE CONTEXTUALIZATION OF EMERGENCY MANAGEMENT SYSTEM COMMUNICATIONS | 07-14-2016 |