Patent application number | Description | Published |
20090024398 | APPARATUS AND METHOD FOR LOW COMPLEXITY COMBINATORIAL CODING OF SIGNALS - The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input. | 01-22-2009 |
20090100121 | APPARATUS AND METHOD FOR LOW COMPLEXITY COMBINATORIAL CODING OF SIGNALS - During operation of an encoder, a signal vector (x) is received. A first multi-precision operand (Ψ′ | 04-16-2009 |
20090112607 | METHOD AND APPARATUS FOR GENERATING AN ENHANCEMENT LAYER WITHIN AN AUDIO CODING SYSTEM - During operation an input signal to be coded is received and coded to produce a coded audio signal. The coded audio signal is then scaled with a plurality of gain values to produce a plurality of scaled coded audio signals, each having an associated gain value and a plurality of error values are determined existing between the input signal and each of the plurality of scaled coded audio signals. A gain value is then chosen that is associated with a scaled coded audio signal resulting in a low error value existing between the input signal and the scaled coded audio signal. Finally, the low error value is transmitted along with the gain value as part of an enhancement layer to the coded audio signal. | 04-30-2009 |
20090231169 | Method and Apparatus for Low Complexity Combinatorial Coding of Signals - To reduce the complexity of the encoding/decoding of pulse positions and/or pulse magnitudes associated with complex combinatorial computations, a method and structure for encoding and decoding of pulse position and/or pulse magnitudes requires fewer computations of these combinatorial functions. Adaptive switching between coding or encoding is performed in accordance with the estimated density of the plurality of occupied positions. | 09-17-2009 |
20090259477 | Method and Apparatus for Selective Signal Coding Based on Core Encoder Performance - In a selective signal encoder, an input signal is first encoded using a core layer encoder to produce a core layer encoded signal. The core layer encoded signal is decoded to produce a reconstructed signal and an error signal is generated as the difference between the reconstructed signal and the input signal. The reconstructed signal is compared to the input signal. One of two or more enhancement layer encoders selected dependent upon the comparison and used to encode the error signal. The core layer encoded signal, the enhancement layer encoded signal and the selection indicator are output to the channel (for transmission or storage, for example). | 10-15-2009 |
20100125453 | APPARATUS AND METHOD FOR ENCODING AT LEAST ONE PARAMETER ASSOCIATED WITH A SIGNAL SOURCE - Apparatus ( | 05-20-2010 |
20100125879 | METHOD AND APPARATUS FOR PURCHASING A SOUNDTRACK WHEN VIEWING A MOVIE OR OTHER PROGRAM DELIVERED BY A CONTENT DELIVERY SYSTEM - A method is provided for offering for purchase supplemental content associated with a multimedia program. The method includes receiving over a content delivery system a multimedia program and supplemental content associated with the program. While the program is being rendered, an option is presented to a user to acquire the supplemental content. | 05-20-2010 |
20100169087 | SELECTIVE SCALING MASK COMPUTATION BASED ON PEAK DETECTION - A set of peaks in a reconstructed audio vector Ŝ of a received audio signal is detected and a scaling mask ψ(Ŝ) based on the detected set of peaks is generated. A gain vector g* is generated based on at least the scaling mask and an index j representative of the gain vector. The reconstructed audio signal is scaled with the gain vector to produce a scaled reconstructed audio signal. A distortion is generated based on the audio signal and the scaled reconstructed audio signal. The index of the gain vector based on the generated distortion is output. | 07-01-2010 |
20100169091 | DEVICE, SYSTEM AND METHOD FOR PROVIDING TARGETED ADVERTISEMENTS AND CONTENT - An aspect of the present invention is drawn to an audio data processing device for use by a user to control a system and for use with a microphone, a user demographic profiles database and a content/ad database. The microphone may be operable to detect speech and to generate speech data based on the detected speech. The user demographic profiles database may be capable of having demographic data stored therein. The content/ad database may be capable of having at least one of content data and advertisement data stored therein. The audio data processing device includes a voice recognition portion, a voice analysis portion and a speech to text portion. The voice recognition portion may be operable to process user instructions based on the speech data. The voice analysis portion may be operable to determine characteristics of the user based on the speech data. The speech to text portion may be operable to determine interests of the user. | 07-01-2010 |
20100169099 | METHOD AND APPARATUS FOR GENERATING AN ENHANCEMENT LAYER WITHIN A MULTIPLE-CHANNEL AUDIO CODING SYSTEM - During operation a multiple channel audio input signal is received and coded to generate a coded audio signal. A balance factor having balance factor components each associated with an audio signal of the multiple channel audio signal is generated. A gain value to be applied to the coded audio signal to generate an estimate of the multiple channel audio signal based on the balance factor and the multiple channel audio signal is determined, with the gain value configured to minimize a distortion value between the multiple channel audio signal and the estimate of the multiple channel audio signal. The representation of the gain value may be output for transmission and/or storage. | 07-01-2010 |
20100169100 | SELECTIVE SCALING MASK COMPUTATION BASED ON PEAK DETECTION - A set of peaks in a reconstructed audio vector Ŝ of a received audio signal is detected and a scaling mask ψ(Ŝ) based on the detected set of peaks is generated. A gain vector g* is generated based on at least the scaling mask and an index j representative of the gain vector. The reconstructed audio signal is scaled with the gain vector to produce a scaled reconstructed audio signal. A distortion is generated based on the audio signal and the scaled reconstructed audio signal. The index of the gain vector based on the generated distortion is output. | 07-01-2010 |
20100169101 | METHOD AND APPARATUS FOR GENERATING AN ENHANCEMENT LAYER WITHIN A MULTIPLE-CHANNEL AUDIO CODING SYSTEM - During operation a multiple channel audio input signal is received and coded to generate a coded audio signal. A balance factor having balance factor components each associated with an audio signal of the multiple channel audio signal is generated. A gain value to be applied to the coded audio signal to generate an estimate of the multiple channel audio signal based on the balance factor and the multiple channel audio signal is determined, with the gain value configured to minimize a distortion value between the multiple channel audio signal and the estimate of the multiple channel audio signal. The representation of the gain value may be output for transmission and/or storage. | 07-01-2010 |
20100286980 | METHOD AND APPARATUS FOR SPEECH CODING - A method and apparatus for prediction in a speech-coding system extends a 1 | 11-11-2010 |
20110095920 | ENCODER AND DECODER USING ARITHMETIC STAGE TO COMPRESS CODE SPACE THAT IS NOT FULLY UTILIZED - An encoder/decoder architecture including an arithmetic encoder that encodes the MSB portions of a Factorial Pulse Coder output, and that encodes an output of a first-level source encoder, e.g., MDCT. Sub-parts (e.g., frequency bands) of portions (e.g., frames) of the signal are sorted in increasing order based on a measure related to signal energy (e.g., signal energy itself). In a system that overlays Arithmetic Encoding on Factorial Pulse coding, the result is bits re-allocated to bands with higher signal energy content, yielding higher signal quality and higher bit utilization efficiency. | 04-28-2011 |
20110096830 | Encoder that Optimizes Bit Allocation for Information Sub-Parts - A encoder/decoder architecture ( | 04-28-2011 |
20110156932 | HYBRID ARITHMETIC-COMBINATORIAL ENCODER - Hybrid range coding/combinatorial coding (FPC) encoders and decoders are provided. Encoding and decoding can be dynamically switched between range coding and combinatorial according to the ratio of ones to the ratio of bits in a partial remaining sequence in order to reduce the computational complexity of encoding and decoding. | 06-30-2011 |
20110161087 | Embedded Speech and Audio Coding Using a Switchable Model Core - A method for processing an audio signal including classifying an input frame as either a speech frame or a generic audio frame, producing an encoded bitstream and a corresponding processed frame based on the input frame, producing an enhancement layer encoded bitstream based on a difference between the input frame and the processed frame, and multiplexing the enhancement layer encoded bitstream, a codeword, and either a speech encoded bitstream or a generic audio encoded bitstream into a combined bitstream based on whether the codeword indicates that the input frame is classified as a speech frame or as a generic audio frame, wherein the encoded bitstream is either a speech encoded bitstream or a generic audio encoded bitstream. | 06-30-2011 |
20110218797 | ENCODER FOR AUDIO SIGNAL INCLUDING GENERIC AUDIO AND SPEECH FRAMES - A method for encoding audio frames by producing a first frame of coded audio samples by coding a first audio frame in a sequence of frames, producing at least a portion of a second frame of coded audio samples by coding at least a portion of a second audio frame in the sequence of frames, and producing parameters for generating audio gap filler samples, wherein the parameters are representative of either a weighted segment of the first frame of coded audio samples or a weighted segment of the portion of the second frame of coded audio samples. | 09-08-2011 |
20110218799 | DECODER FOR AUDIO SIGNAL INCLUDING GENERIC AUDIO AND SPEECH FRAMES - A method for decoding audio frames includes producing a first frame of coded audio samples, producing at least a portion of a second frame of coded audio samples, generating audio gap filler samples based on parameters representative of a weighted segment of the first frame of coded audio samples or a weighted segment of the portion of the second frame of coded audio samples, and forming a sequence including the audio gap filler samples and the portion of the second frame of coded audio samples. | 09-08-2011 |
20120095757 | AUDIO SIGNAL BANDWIDTH EXTENSION IN CELP-BASED SPEECH CODER - A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element. | 04-19-2012 |
20120095758 | AUDIO SIGNAL BANDWIDTH EXTENSION IN CELP-BASED SPEECH CODER - A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element. | 04-19-2012 |
20120226506 | METHOD AND APPARATUS FOR GENERATING AN ENHANCEMENT LAYER WITHIN A MULTIPLE-CHANNEL AUDIO CODING SYSTEM - A method and apparatus are disclosed for generating a coded audio signal based on a multiple channel audio input signal. A balance factor having balance factor components each associated with an audio signal of the multiple channel audio signal is generated. A gain value to be applied to the coded audio signal to generate an estimate of the multiple channel audio signal based on the balance factor and the multiple channel audio signal is determined, with the gain value configured to minimize a distortion value between the multiple channel audio signal and the estimate of the multiple channel audio signal. | 09-06-2012 |
20120235840 | Encoder that Optimizes Bit Allocation for Information Sub-Parts - A digital information encoder including a divider configured to divide a block of information into a plurality of sub-parts, an initial bit allocator configured to perform an initial allocation of bits to a K | 09-20-2012 |
20120284032 | APPARATUS AND METHOD FOR LOW COMPLEXITY COMBINATORIAL CODING AND DECODING OF SIGNALS - A method and apparatus for low complexity combinatorial coding and decoding of signals is described herein. During operation, an encoder and a decoder will utilize a first function in determining a codeword or vector when the size of the function is small. The encoder and the decoder will also utilize a second function in determining the codeword or vector when the size of the function is large. | 11-08-2012 |
20130148749 | APPARATUS AND METHOD FOR COMBINATORIAL CODING OF SIGNALS - A method and apparatus are for performing one of encoding and decoding a code word that is used to communicate a portion of a signal. For encoding, at least a portion of a code word is encoded from a signal based value using an approximation of a combinatorial function, wherein the signal based value represents one or more aspects of a signal. For decoding, at least a portion of a code word is decoded to a signal based value using an approximation of a combinatorial function, wherein the signal based value represents one or more aspects of a signal. The approximation of the combinatorial function is based on a linear combination of a set of basis functions. | 06-13-2013 |
20130173259 | Method and Apparatus for Processing Audio Frames to Transition Between Different Codecs | 07-04-2013 |
20130211846 | ALL-PASS FILTER PHASE LINEARIZATION OF ELLIPTIC FILTERS IN SIGNAL DECIMATION AND INTERPOLATION FOR AN AUDIO CODEC - An audio signal processing system includes parallel speech and generic audio signal processing paths. One path includes a linear predictive coder and a resampling filter having a non-linear phase characteristic. A phase compensation filter is disposed along the one of the processing paths to compensate for the non-linearity of the resampling filter thereby enabling relatively seamless switching between the coders resulting in a reduction of audio artifacts that would otherwise result from the non-linear phase characteristic of the resampling filter during playback. | 08-15-2013 |
20130254249 | APPARATUS AND METHOD FOR LOW COMPLEXITY COMBINATORIAL CODING OF SIGNALS - The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input. | 09-26-2013 |
20130268266 | Method and Apparatus for Generating a Candidate Code-Vector to Code an Informational Signal | 10-10-2013 |
20140019142 | APPARATUS AND METHOD FOR AUDIO FRAME LOSS RECOVERY - A method and apparatus provide for audio frame recovery by identifying a sequence of lost frames of coded audio data as being lost or corrupted; identifying a first frame of coded audio data which immediately preceded the sequence of lost frames, as having been encoded using a time domain coding method; identifying a second frame of coded audio data, which immediately followed the sequence of lost frames of coded audio data, as having been encoded using a transform domain coding method; obtaining a pitch delay; generating a second decoded audio portion of the second frame based on the second frame; generating a first decoded audio portion of the second frame based on the pitch delay and decoded audio samples; and generating a decoded audio output of the second frame based on a sequential combination of the first and second decoded audio portions. | 01-16-2014 |
20140028785 | VIDEO BANDWIDTH ALLOCATION IN A VIDEO CONFERENCE - Communicating information exchanged in a video conference. Responsive to detecting a change in an amount of audio information generated by at least one participant in the video conference, identifying a first of the plurality of participants in the video conference that currently is a primary presenter in the video conference, allocating a first level of video bandwidth to communicate video information generated in the video conference to a client device, by the primary presenter in the video conference, and allocating a second level of video bandwidth to communicate video information generated in the video conference to a client device, by one or more other participants, who currently are not the primary presenter in the video conference, wherein the second level of video bandwidth is less than the first level of video bandwidth. | 01-30-2014 |
20140088974 | APPARATUS AND METHOD FOR AUDIO FRAME LOSS RECOVERY - A method and apparatus provides for frame loss recovery following a loss of a frame in an audio codec. The lost frame is identified. Estimated linear predictive coefficients of a previous transform frame are generated based on a decoded audio of the previous transform frame. An estimated residual of the previous transform frame is generated based on the estimated linear predicative coefficients and the decoded audio. A pitch delay is determined from frame error recovery parameters received with the previous transform frame. An extended residual is generated based on the pitch delay and the estimated residual. A first synthesized signal is generated based on the extended residual and the linear predicative coefficients. A decoded audio output of at least the lost frame is generated based on the first synthesized signal. The frame error recovery parameters are generated by an encoder. | 03-27-2014 |
20140129214 | Method and Apparatus for Generating a Candidate Code-Vector to Code an Informational Signal | 05-08-2014 |
20140257798 | CONVERSION OF LINEAR PREDICTIVE COEFFICIENTS USING AUTO-REGRESSIVE EXTENSION OF CORRELATION COEFFICIENTS IN SUB-BAND AUDIO CODECS - Disclosed are systems and methods for the efficient conversion of linear predictive coefficients. This method is usable, for example, in the conversion of full band linear predictive coding (“LPC”) coefficients to sub-band LPCs of a sub-band speech codec. The sub-bands may or may not be down-sampled. In an embodiment, the LPC coefficients of the sub-bands are obtained from the correlation coefficients, which are in turn obtained by filtering the auto-regressive extended auto-correlation coefficients of the full band LPCs. The method also allows the generation of an LPC approximation of a pole-zero weighted synthesis filter. | 09-11-2014 |