Patent application number | Description | Published |
20080203065 | System and Method For Controlling And Coordinating Welding-Type Processes and Gouging-Type Processes - A system and method for an integrated structural welding system includes a welding-type power source configured to deliver welding-type power for a variety of welding-type processes. The system also includes a gouging torch connected to the welding-type power source to receive welding-type power during a gouging-type process and a wire feeder connected to the welding-type power source to receive welding-type power during a welding-process. Furthermore, the system includes a controller configured to coordinate operation of the wire feeder, the gouging torch, and the power supply to perform only one of the gouging-type process and the welding-type process at a given time. | 08-28-2008 |
20080203066 | MULTI-CABLE UMBILICAL CORD SYSTEM - A system and method for an integrated structural welding system is designed to improve work flow efficiency. Accordingly, a multi-cable umbilical cord system is provided that combines and couples a number of power, control, and gas deliver cables. | 08-28-2008 |
20080203067 | SYSTEM AND METHOD FOR COORDINATING WELDING AND GOUGING OPERATIONS - A system and method for an integrated structural welding system includes a user interface and controller configured to coordinate both welding and gouging operations. | 08-28-2008 |
20080203073 | WELDING-TYPE SYSTEM HAVING A WIRE FEEDER SYSTEM HAVING INTEGRATED POWER SOURCE CONTROLS AND A WELDING-TYPE POWER SOURCE THAT IS FREE POWER PARAMETER SELECTION INTERFACES - A system and method for an integrated structural welding system is designed to protect improve work flow efficiency. Specifically, a welding-type power source is provided that is free of user interface devices designed to select power parameters. Instead, a wire feeder is provided that includes a user interface configured to receive operational and power parameters and a controller configured to control operation of the welding-type power source based on the feedback received through the user interface. | 08-28-2008 |
20080203074 | SYSTEM AND METHOD FOR PROTECTING A WELDING-TYPE SYSTEM FROM STRAIN - A system and method for an integrated structural welding system is designed to protect the components of the system against accidental damage and undue stresses. Specifically, portable structure and a strain protection system are provided. The strain protection system is supported by a portable structure and engages one or both of a welding-type cable and a power cable to transfer forces applied to the welding-type cable or power cable when adjusting a position of the portable structure to the support structure. | 08-28-2008 |
20080203075 | PORTABLE STRUCTURAL WELDING SYSTEM HAVING INTEGRATED RESOURCES - A system and method for an integrated structural welding system is designed to improve work flow efficiency. Specifically, a portable support structure is provided that includes a number of integrated components, such as transmission power receptacles and gas supply connections, that provide ready access to resources to perform a number of welding-type processes. | 08-28-2008 |
20110204013 | Welding Power Supply External Protective Support Structure - An external support structure for a welding power supply includes a cage-like assembly that is configured to receive and at least partially surround the welding power supply. The cage-like assembly includes fastening points allowing for securement of the welding power supply within the assembly. The assembly further includes stacking members to permit stacking of the assembly with a self-similar structure. | 08-25-2011 |
20130168374 | METHOD AND DEVICE FOR CONTROLLING CURRENT FLOW THROUGH A WELDING CLAMP - A method and device for controlling current flow through a welding clamp are provided. One welding clamp includes a first contact piece configured to contact a workpiece and to provide a first current path that limits current flow. The welding clamp also includes a second contact piece configured to contact the workpiece and to provide a second current path. When the welding clamp is being clamped to the workpiece, the first contact piece is configured to contact the workpiece prior to the second contact piece. In addition, the welding clamp is configured so that current flows through the first current path prior to flowing through the second current path when the welding clamp is being clamped to the workpiece. | 07-04-2013 |
20150034602 | System and Method for Controlling and Coordinating Welding-Type Processes and Gouging-Type Processes - A multi-operational welding-type system including a user interface supported by a support structure and including a first user interface device moveable between a welding position, a gouging position, and an off position, and a second user interface device configured to alter operation parameters of the multi-operational welding-type system. A controller is supported by the support structure and monitors the position of the first user interface device and utilizes the operational parameters supplied via the second user interface device. The multi-operational welding-type system may perform only one of a gouging-type process and a welding-type process at a given time depending on the position of the first user interface device. | 02-05-2015 |
Patent application number | Description | Published |
20100223054 | SINGLE-MICROPHONE WIND NOISE SUPPRESSION - A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame. | 09-02-2010 |
20110029317 | DYNAMIC TIME SCALE MODIFICATION FOR REDUCED BIT RATE AUDIO CODING - Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder. | 02-03-2011 |
20110096942 | NOISE SUPPRESSION SYSTEM AND METHOD - Systems and methods are described for applying noise suppression to one or more audio signals to generate a noise-suppressed audio signal therefrom. In a single-channel implementation, an input signal is received that comprises a desired audio signal and an additive noise signal. Noise suppression is then applied to the input signal to generate a noise-suppressed signal in a manner that is controlled by at least a parameter that specifies a degree of balance between distortion of the desired audio signal and unnaturalness of a residual noise signal included in the noise-suppressed signal. In an alternative single-channel implementation, a plurality of sub-band signals obtained by applying a frequency conversion process to a time domain representation of an input signal is received. Noise suppression is then applied to each of the sub-band signals by passing each of the sub-band signals through a time direction filter. Multi-channel noise suppression variants are also described. | 04-28-2011 |
20110320213 | TIME-WARPING OF DECODED AUDIO SIGNAL AFTER PACKET LOSS - A technique is described for use in a decoder configured to decode a series of frames representing an encoded audio signal. The technique is for transitioning between a lost frame and one or more received frames following the lost frame in the series of frames. In accordance with the technique, an output audio signal associated with the lost frame is synthesized. An extrapolated signal is generated based on the synthesized output audio signal. A time lag is calculated between the extrapolated signal and a decoded audio signal associated with the received frame(s), wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoded audio signal is time-warped based on the time lag, wherein time-warping the decoded audio signal comprises stretching or shrinking the decoded audio signal in the time domain. | 12-29-2011 |
20120010882 | CONSTRAINED AND CONTROLLED DECODING AFTER PACKET LOSS - A technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal. | 01-12-2012 |
20120121100 | Method and Apparatus For Wind Noise Detection and Suppression Using Multiple Microphones - Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this tact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described. | 05-17-2012 |
20120123771 | Method and Apparatus For Wind Noise Detection and Suppression Using Multiple Microphones - Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described. | 05-17-2012 |
20120123772 | System and Method for Multi-Channel Noise Suppression Based on Closed-Form Solutions and Estimation of Time-Varying Complex Statistics - Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system. | 05-17-2012 |
20120123773 | System and Method for Multi-Channel Noise Suppression - Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage. | 05-17-2012 |
20120185246 | NOISE SUPPRESSION USING MULTIPLE SENSORS OF A COMMUNICATION DEVICE - Techniques are described herein that suppress noise using multiple sensors (e.g., microphones) of a communication device. Noise modeling (e.g., estimation of noise basis vectors and noise weighting vectors) is performed with respect to a noise signal during operation of a communication device to provide a noise model. The noise model includes noise basis vectors and noise coefficients that represent noise provided by audio sources other than a user of the communication device. Speech modeling (e.g., estimation of speech basis vectors and speech weighting) is performed to provide a speech model. The speech model includes speech basis vectors and speech coefficients that represent speech of the user. A noisy speech signal is processed using the noise basis vectors, the noise coefficients, the speech basis vectors, and the speech coefficients to provide a clean speech signal. | 07-19-2012 |
20130163781 | BREATHING NOISE SUPPRESSION FOR AUDIO SIGNALS - Systems and methods are described herein for detecting and suppressing breathing noise in an audio signal. First, systems and methods are described that analyze audio signals generated by two or more microphones to detect breathing noise in one of the audio signals and that leverage the multiple microphones to suppress detected breathing noise in a manner that minimizes signal distortion. Then, systems and methods are described that are capable of analyzing the audio signal generated by a single microphone to detect breathing noise in the audio signal and thereafter suppress it. | 06-27-2013 |
20130195163 | SYSTEMS AND METHODS FOR ENHANCING AUDIO QUALITY OF FM RECEIVERS - Systems and methods are described for enhancing the audio quality of an FM receiver. In embodiments described herein, quadrature L−R demodulation is applied to a composite baseband signal output by an FM demodulator to obtain an L−R noise signal. A channel quality measure is calculated based on the L−R noise signal and is used to control whether a pop suppression technique is applied to an L+R signal obtained from the composite baseband signal to detect and remove noise pulses therefrom. The channel quality measure and the L−R noise signal are also leveraged to perform single-channel noise suppression in the frequency domain on an L−R signal obtained from the composite baseband signal and on the L+R signal. The channel quality measure is also used to control the application of a fast fading compensation process that replaces noisy segments of the L−R and L+R signal with replacement waveforms generated via waveform extrapolation. | 08-01-2013 |
20130195164 | SYSTEMS AND METHODS FOR ENHANCING AUDIO QUALITY OF FM RECEIVERS - Systems and methods are described for enhancing the audio quality of an FM receiver. In embodiments described herein, a stop band noise signal is extracted from an L+R or L−R signal produced by an FM stereo decoder. A channel quality measure is calculated based on the stop band noise signal and is used to control whether a pop suppression technique is applied to the L+R signal. The channel quality measure and the stop band noise signal are also leveraged to perform single-channel noise suppression in the frequency domain on the L−R signal and on the L+R signal. The channel quality measure is also used to control the application of a fast fading compensation process that replaces noisy segments of the L−R and L+R signal with replacement waveforms generated via waveform extrapolation. | 08-01-2013 |
20130216056 | NON-LINEAR ECHO CANCELLATION - A two-stage structure for performing non-linear echo cancellation is described in which a first echo canceller is used to attenuate linear echo components of a microphone signal and a second echo canceller is used to attenuate non-linear echo components of the output signal generated by the first echo canceller. One or both of the echo cancellers may be implemented using closed-form solutions, including a closed form solution for a hybrid method in the frequency domain. | 08-22-2013 |
20130216057 | ECHO CANCELLATION USING CLOSED-FORM SOLUTIONS - A system that utilizes closed-form solutions to perform echo cancellation is described. The system includes a filter, filter parameter determination logic and a combiner. The filter is configured to process a far-end audio signal in accordance with one or more filter parameters to generate an estimated echo signal. The filter parameter determination logic is configured to update estimated statistics associated with the far-end audio signal and a microphone signal based on instantaneous statistics associated with the far-end audio signal and the microphone signal, and calculate the one or more filter parameters based upon the updated estimated statistics. The combiner is configured to generate an estimated near-end audio signal by subtracting the estimated echo signal from the microphone signal. | 08-22-2013 |
20130325467 | SYSTEMS AND METHODS FOR PRESENTING AUDIO MESSAGES - Systems and methods for presenting audio messages are provided. In some aspects, a method includes receiving an audio message from a first user and generating a text-based representation of the audio message. The method also includes generating one or more identification tags based on the text-based representation of the audio message. At least one of the one or more identification tags includes a subject of the audio message. The method also includes presenting at least one of the text-based representation of the audio message or the one or more identification tags to a second user using a graphical user interface. | 12-05-2013 |
20140188466 | INTEGRATED SPEECH INTELLIGIBILITY ENHANCEMENT SYSTEM AND ACOUSTIC ECHO CANCELLER - A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith. | 07-03-2014 |
20140278397 | SPEAKER-IDENTIFICATION-ASSISTED UPLINK SPEECH PROCESSING SYSTEMS AND METHODS - Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in an uplink path of a communication device. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify the identity of a near-end speaker. Knowledge of the identity of the near-end speaker is then used to improve the performance of one or more uplink speech processing algorithms implemented on the communication device. | 09-18-2014 |
20140278417 | SPEAKER-IDENTIFICATION-ASSISTED SPEECH PROCESSING SYSTEMS AND METHODS - Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify a user of the communication device and/or the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the user and/or far-end speaker is then used to improve the performance of one or more speech processing algorithms implemented on the communication device. | 09-18-2014 |
20140278418 | SPEAKER-IDENTIFICATION-ASSISTED DOWNLINK SPEECH PROCESSING SYSTEMS AND METHODS - Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in a downlink path of a communication device. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the far-end speaker is then used to improve the performance of one or more downlink speech processing algorithms implemented on the communication device. | 09-18-2014 |
20140286497 | MULTI-MICROPHONE SOURCE TRACKING AND NOISE SUPPRESSION - Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described. | 09-25-2014 |
20150071461 | SINGLE-CHANNEL SUPPRESSION OF INTEFERING SOURCES - Techniques described herein are directed to performing back-end single-channel suppression of one or more types of interfering sources (e.g., additive noise) in an uplink path of a communication device. The back-end single-channel suppression techniques may suppress types(s) of additive noise using one or more suppression branches (e.g., a non-spatial (or stationary noise) branch, a spatial (or non-stationary noise) branch, a residual echo suppression branch, etc.). The non-spatial branch may be configured to suppress stationary noise from the single-channel audio signal, the spatial branch may be configured to suppress non-stationary noise from the single-channel audio signal and the residual echo suppression branch may be configured to suppress residual echo from the signal-channel audio signal. The spatial branch may be disabled based on an operational mode (e.g., single-user speakerphone mode or a conference speakerphone mode) of the communication device or based on a determination that spatial information is ambiguous. | 03-12-2015 |