Abstract: The exemplary embodiments of the invention provide at least a method and an apparatus to perform operations including dividing a sound signal into a series of successive frames, dividing each frame into a number of subframes, producing a residual signal by filtering the sound signal through a linear prediction analysis filter, locating a last pitch pulse of the sound signal of a previous frame from the residual signal, extracting a pitch pulse prototype of given length around a position of the last pitch pulse of the previous frame using the residual signal, and locating pitch pulses in a current frame using the pitch pulse prototype.

Abstract: In accordance with the exemplary embodiments of the invention there is disclosed at least a method and apparatus for determining a long-term-prediction delay parameter characterizing a long term prediction in a technique using signal modification for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, a feature of the sound signal is located in a previous frame, a corresponding feature of the sound signal is located in a current frame, and the long-term-prediction delay parameter is determined for the current frame while mapping, with the long term prediction, the signal feature of the previous frame with the corresponding signal feature of the current frame. Each divided frame of the sound signal is partitioned into a plurality of signal segments, and at least a part of the signal segments of the frame are warped while constraining the warped signal segments inside the frame.

Abstract: From a survey vehicle, two types of data are obtained: images from a camera and reflections from a laser. The laser data is filtered in order to detect assets. The detected assets are then processed to locate and measure the assets. Once this information has been obtained, it is combined with the images captured by the camera in order to recognize the assets.

Abstract: In a method and device for post-processing a decoded sound signal in view of enhancing a perceived quality of this decoded sound signal, the decoded sound signal is divided into a plurality of frequency sub-band signals, and post-processing is applied to at least one of the frequency sub-band signal. After post-processing of this at least one frequency sub-band signal, the frequency sub-band signals may be added to produce an output post-processed decoded sound signal. In this manner, the post-processing can be localized to a desired sub-band or sub-bands with leaving other sub-bands virtually unaltered.

Abstract: For determining a long-term-prediction delay parameter characterizing a long term prediction in a technique using signal modification for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, a feature of the sound signal is located in a previous frame, a corresponding feature of the sound signal is located in a current frame, and the long-term-prediction delay parameter is determined for the current frame while mapping, with the long term prediction, the signal feature of the previous frame with the corresponding signal feature of the current frame. In a signal modification method for implementation into a technique for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, each frame of the sound signal is partitioned into a plurality of signal segments, and at least a part of the signal segments of the frame are warped while constraining the warped signal segments inside the frame.

Abstract: A method for the time scaling of a sampled audio signal is presented. The method includes a first step of performing a pitch and voicing analysis of each frame of the signal in order to determine if a given frame is voiced or unvoiced and to evaluate a pitch profile for voiced frames. The results of this analysis are used to determine the length and position of analysis windows along each frame. Once an analysis window is determined, it is overlap-added to previously synthesized windows of the output signal.

Abstract: From a survey vehicle, two types of data are obtained: images from a camera and reflections from a laser. The laser data is filtered in order to detect assets. The detected assets are then processed to locate and measure the assets. Once this information has been obtained, it is combined with the images captured by the camera in order to recognize the assets.

Abstract: The invention concerns an electrical reactor for reforming, in the presence of an oxidant gas, a gas comprising at least one hydrocarbon, and/or at least one organic compound, including carbon and hydrogen atoms as well as at least one heteroatom. Said reactor comprises: an enclosure, a reaction chamber provided with at least two electrodes comprising at least one conductive lining material electrically isolated from the metal wall of the enclosure, at least one supply of gas to be reformed, at least one oxidant gas supply, at least one outlet for the gases from the reforming and one electrical source for powering the electrodes and resulting in generation of an electronic flux in the conductive lining between the electrodes and in heating said lining.

Abstract: In a method and device for post-processing a decoded sound signal in view of enhancing a perceived quality of this decoded sound signal, the decoded sound signal is divided into a plurality of frequency sub-band signals, and post-processing is applied to at least one of the frequency sub-band signal. After post-processing of this at least one frequency sub-band signal, the frequency sub-band signals may be added to produce an output post-processed decoded sound signal. In this manner, the post-processing can be localized to a desired sub-band or sub-bands with leaving other sub-bands virtually unaltered.

Abstract: For determining a long-term-prediction delay parameter characterizing a long term prediction in a technique using signal modification for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, a feature of the sound signal is located in a previous frame, a corresponding feature of the sound signal is located in a current frame, and the long-term-prediction delay parameter is determined for the current frame while mapping, with the long term prediction, the signal feature of the previous frame with the corresponding signal feature of the current frame. In a signal modification method for implementation into a technique for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, each frame of the sound signal is partitioned into a plurality of signal segments, and at least a part of the signal segments of the frame are warped while constraining the warped signal segments inside the frame.

Abstract: A device for marking a selected tissue location within a patient's body comprises at least two markers (16, 116, 216) pre-loaded into a hollow needle (18, 118, 218) and packaged in a sterile fashion with a stopper (28, 128, 218) to prevent a stylet (27, 127, 227) inserted into the needle (18, 118, 218) from releasing the markers (16, 116, 216) during shipping. An indexing mechanism (28/29, 128/129, 228/229) is also provided for releasably blocking the stylet (27, 127, 227) at successive predetermined levels of insertion into the needle with each level of insertion being associated with the discharge of a single marker (16, 116, 216).

Abstract: A method and device for analog simultaneous voice and data transmission over an analog channel encodes at a voice signal sampling rate a sampled voice signal including spectral and pitch information. During encoding, the spectral and pitch information is extracted from the sampled voice signal to produce a residual signal, and the residual signal is converted into a sequence of residual vectors at a Baud rate. Then, a bitstream including non voice data and the extracted spectral and pitch information is produced. A modem unit, operating at Baud rate and producing an output carrier wave supplied to the analog channel, produces a sequence of symbol vectors in response to the sequence of residual vectors and the bitstream, and modulates the output carrier wave in response to the sequence of symbol vectors.

Abstract: Instability inherent in analysis-by-synthesis speech/audio codecs and caused in particular by channel errors during transmission of highly periodic signals such as high-frequency sine waves is removed. Analysis-by-synthesis techniques involve production, in response to the speech/audio signal and at regular time intervals called frames, of (a) a set of spectral parameters for use in driving a synthesis filter in view of synthesizing the speech/audio signal, and (b) a pitch gain for constructing a past-excitation-signal component supplied to the synthesis filter.

Abstract: A method and device eradicate the occasional instability inherent in analysis-by-synthesis speech/audio codecs and caused in particular by channel errors during transmission of highly periodic signals such as high-frequency sine waves. Analysis-by-synthesis techniques involve production, in response to the speech/audio signal and at regular time intervals called frames, of (a) a set of spectral parameters for use in driving a synthesis filter in view of synthesizing the speech/audio signal, and (b) a pitch gain for constructing a past-excitation-signal component supplied to the synthesis filter.

Abstract: A codebook is searched in view of encoding a sound signal. This codebook consists of a set of pulse amplitude/position combinations each defining L different positions and comprising both zero-amplitude pulses and non-zero-amplitude pulses assigned to respective positions p=1, 2, . . . L of the combination, wherein each non-zero-amplitude pulse assumes at least one of q possible amplitudes. To reduce the search complexity, a subset of pulse amplitude/position combinations from the codebook is pre-selected in relation to the sound signal, and only this subset of combinations is searched. The pre-selection of the subset of combinations consists of pre-establishing, in relation to the sound signal, a function S.sub.p between the respective positions p=1, 2, . . . L and the q possible amplitudes, the search being limited to the combinations of the codebook having non-zero-amplitude pulses which respect the pre-established function.

Abstract: A codebook is searched in view of encoding a sound signal. This codebook consists of a set of codevectors each of 40 positions and comprising N non-zero-amplitude pulses assignable to predetermined valid positions. To reduce the search complexity, a depth-first search is used which involves a tree structure with levels ordered from 1 through M. A path-building operation takes place at each level whereby a candidate path from the previous level is extended by choosing a predetermined number of new pulses and selecting valid positions for said new pulses in accordance with a given pulse-order rule and a given selection criterion. A path originated at the first level and extended by the path-building operations of subsequent levels determines the respective positions of the N non-zero-amplitude pulse of a candidate codevector. Use of a signal-based pulse-position likelihood estimate during the first few levels enable initial pulse-screening to start the search on favorable conditions.

Abstract: A method of encoding a speech signal is provided. This method improves the excitation codebook and search procedure of the conventional Code-Excited Linear Prediction (CELP) speech encoders. This code is based on a sparse algebraic code consisting in particular, but not exclusively, of interleaving N single-pulse permutation codes. The search complexity in finding the best codeword is greatly reduced by bringing the search back to the algebraic code domain thereby allowing the sparsity of the algebraic code to speed up the necessary computations. More precisely, the sparsity of the code enable the use of a very fast procedure based on N-embedded computation loops.

Abstract: The present invention concerns efficient quantization of more than one LPC spectral models per frame in order to enhance the accuracy of the time-varying spectrum representation without compromising on the coding-rate. Such efficient representation of LPC spectral models is advantageous to a number of techniques used for digital encoding of speech and/or audio signals.

Abstract: A method of encoding a speech signal is presented. This method improves the excitation codebook and search procedure of the conventional Code Excited Linear Prediction (CELP) speech encoders. Use is made of a dynamic codebook (201, 202) based on the combination of two modules: a sparse algebraic code generator (201) associated to a filter (202) having a transfer function varying in time. The generator (102) is a structured codebook with codewords having very few non zero components. The filter (202) shapes the spectral characteristics whereby the resulting excitation codebook (201, 202) exhibits favorable perceptual properties. The search complexity in finding the best codeword is greatly reduced by bringing the search back to the algebraic code domain thereby allowing the sparsity of the algebraic code to speed up the necessary computations.