| Class / Patent application number | Description | Number of patent applications / Date published |
|---|
| 704503000 | AUDIO SIGNAL TIME COMPRESSION OR EXPANSION (E.G., RUN LENGTH CODING) | 55 |
| 20080255860 | Audio decoding apparatus and decoding method - A signal characteristic is detected from a block shape indicating a time/frequency conversion block length by a signal characteristic discrimination unit to discriminate whether the prediction precision in the time domain is high or the prediction precision in the frequency domain is high and, on the basis of a result of the discrimination, a signal correction unit corrects a quantization error in spectral information obtained by de-quantization. | 10-16-2008 |
| 20080262856 | Method and system for enabling audio speed conversion - The present invention provides a method and system for processing an audio signal. According to an exemplary method, an audio signal such as a digital voice signal is received and divided into one or more individual unit cycles. An audio speed conversion operation is enabled by repeating or removing one or more of the individual unit cycles. In particular, repeating one or more of the individual unit cycles decreases audio speed, and removing one or more of the individual unit cycles increases audio speed. | 10-23-2008 |
| 20090030704 | ACOUSTIC SIGNAL ENCODING DEVICE, AND ACOUSTIC SIGNAL DECODING DEVICE - An acoustic signal encoding device for down-mixing at different ratios to encode a multichannel signal with a small number of channels, and an acoustic signal decoding device for decoding the signal encoded by the acoustic signal encoding device. In these devices, weighting means ( | 01-29-2009 |
| 20090048852 | ENCODING AND/OR DECODING DIGITAL CONTENT - Embodiments of methods, apparatuses, devices and systems associated with encoding and/or decoding audio data are disclosed. | 02-19-2009 |
| 20090125315 | TRANSCODER USING ENCODER GENERATED SIDE INFORMATION - An audio encoder encodes side information into a compressed audio bitstream containing encoding parameters used by the encoder for one or more encoding techniques, such as a noise-mask-ratio curve used for rate control. A transcoder uses the encoder generated side information to transcode the audio from the original compressed bitstream having an initial bit-rate into a second bitstream having a new bit-rate. Because the side information is derived from the original audio, the transcoder is able to better maintain audio quality of the transcoding. The side information also allows the transcoder to re-encode from an intermediate decoding/encoding stage for faster and lower complexity transcoding. | 05-14-2009 |
| 20090144064 | Local Pitch Control Based on Seamless Time Scale Modification and Synchronized Sampling Rate Conversion - This invention locally controls the pitch of speech and audio signals. The invention is based on a seamless time scale modification (S-TSM) scheme connected to a synchronized sampling rate converter that switches between different time scale factors in a seamless manner and controls pitch during playback in a nearly continuous way. | 06-04-2009 |
| 20090171677 | AUDIO SUBSYSTEM SHARING IN A VIRTUALIZED ENVIRONMENT - A device, method, and system are disclosed. In one embodiment the device includes a first virtual machine to directly access a physical audio codec. The device also includes a virtual audio codec that is managed by the first virtual machine. The virtual audio codec can provide a custom interface to the physical audio codec for one or more additional virtual machines apart from the first virtual machine. | 07-02-2009 |
| 20090216544 | AUDIO SIGNAL ENCODING OR DECODING - Encoding an audio signal is provided wherein the audio signal includes a first audio channel and a second audio channel, the encoding comprising subband filtering each of the first audio channel and the second audio channel in a complex modulated filterbank to provide a first plurality of subband signals for the first audio channel and a second plurality of subband signals for the second audio channel, downsampling each of the subband signals to provide a first plurality of downsampled subband signals and a second plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, deriving spatial parameters from the sub-subband signals and from those downsampled subband signals that are not further subband filtered, and deriving a single channel audio signal comprising derived subband signals derived from the first plurality of downsampled subband signals and the second plurality of downsampled subband signals. | 08-27-2009 |
| 20090248424 | LOSSLESS AND NEAR LOSSLESS SCALABLE AUDIO CODEC - A scalable audio codec encodes an input audio signal as a base layer at a high compression ratio and one or more residual signals as an enhancement layer of a compressed bitstream, which permits a lossless or near lossless reconstruction of the input audio signal at decoding. The scalable audio codec uses perceptual transform coding to encode the base layer. The residual is calculated in a transform domain, which includes a frequency and possibly also multi-channel transform of the input audio. For lossless reconstruction, the frequency and multi-channel transforms are reversible. | 10-01-2009 |
| 20090248425 | AUDIO WAVE FIELD ENCODING - An encoder/decoder for multi-channel audio data, and in particular for audio reproduction through wave field synthesis. The encoder comprises a two-dimensional filter-bank to the multi-channel signal, in which the channel index is treated as an independent variable as well as time, and and the resulting spectral coefficient are quantized according to a two-dimensional psychoacoustic model, including masking effect in the spatial frequency as well as in the temporal frequency. The coded spectral data are organized in a bitstream together with side information containing scale factors and Huffman codebook identifiers. | 10-01-2009 |
| 20090299758 | Method and Apparatus for Reducing Access Delay in Discontinuous Transmission Packet Telephony Systems - Systems are disclosed for operating a communications network. The system includes a module to buffer frames of a signal, and a module to determine an access delay. The system also includes a module to compress a portion of the signal based on the access delay by removing a first portion of a frame of the signal and generating an overlap-added segment from a first segment and a second segment of the frame. In another embodiment, the system includes a module to buffer frames of a signal, a module to establish a communication channel with a handset, and a module to determine an access delay. The system also includes a module to compress a portion of the signal based on the access delay by removing a first portion of a frame of the signal and generating an overlap-added segment from a first segment and a second segment of the frame. | 12-03-2009 |
| 20090319283 | Apparatus and Method for Generating Audio Subband Values and Apparatus and Method for Generating Time-Domain Audio Samples - An embodiment of an apparatus for generating audio subband values in audio subband channels has an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function having a sequence of window coefficients to obtain windowed samples. The analysis window function has a first group of window coefficients and a second group of window coefficients. The first group of window coefficients is used for windowing later time-domain samples and the second group of window coefficients is used for windowing an earlier time-domain samples. The apparatus further has a calculator for calculating the audio subband values using the windowed samples. | 12-24-2009 |
| 20090326963 | AUDIO ENCODING DEVICE, AUDIO ENCODING METHOD, AND PROGRAM THEREOF - [Problems] To provide a high-quality audio signal encoding technique by controlling the number of time/frequency groups in a frame. | 12-31-2009 |
| 20100004937 | Method for time scaling of a sequence of input signal values - The invention relates to a digital signal processing technique that changes the length of an audio signal and, thus, effectively its play-out speed. This is used for frame rate conversion or sound effects in music production. Time scaling may further be used for fast forward or slow-motion audio play-out. | 01-07-2010 |
| 20100023336 | Compression of audio scale-factors by two-dimensional transformation - Digital audio samples are represented as a product of scale factors codes and corresponding quantity codes, sometimes referred to as exponent/mantissa format. To compress audio data, scale factors are organized by sample time and frequency either by filtering or frequency transformation, into a two-dimensional frame. The frame may be decomposed into “tiles” by partition. One or more such scale factor tiles are compressed by transformation by a two-dimensional, orthogonal transformation such as a two dimensional discrete cosine transform. Optional further encoding is applied to reduce redundancy. A decoding method and an encoded machine readable medium complement the method of encoding. | 01-28-2010 |
| 20100076774 | AUDIO DECODER - An audio decoder ( | 03-25-2010 |
| 20100100390 | AUDIO ENCODING APPARATUS, AUDIO DECODING APPARATUS, AND AUDIO ENCODED INFORMATION TRANSMITTING APPARATUS - To reduce the amount of transmitted information and further reduce the processing amount at a decoding apparatus. An encoding apparatus ( | 04-22-2010 |
| 20100131281 | Signal Processing Method and Program - The present invention provides a signal processing apparatus, a signal processing method and a program for outputting a high-quality coded string. A signal processing apparatus according to an embodiment of the present invention includes a normalization coefficient information increasing/decreasing circuit | 05-27-2010 |
| 20100138225 | OPTIMIZATION OF MP3 ENCODING WITH COMPLETE DECODER COMPATIBILITY - An iterative rate-distortion optimization algorithm for MPEG I/II Layer-3 (MP3) encoding based on the method of Lagrangian multipliers. Generally, an iterative method is performed such that a global quantization step size is determined while scale factors are fixed, and thereafter the scale factors are determined while the global quantization step size is fixed. This is repeated until a calculated rate-distortion cost is within a predetermined threshold. The methods are demonstrated to be computationally efficient and the resulting bit stream is fully standard compatible. | 06-03-2010 |
| 20100169105 | Discrete time expansion systems and methods - The present invention relates to discrete time expansion systems and methods for expanding a source signal while at least substantially preserving its frequency distribution and obviating a need to smoothen an expanded signal. Such a system may expand the source signal by a preset expansion ratio which is any integer or any real number represented by a ratio of (m+n)/m or (m+n+0.5)/m where m and n are positive integers. The present invention also relates to various methods of expanding the source signal by separating such a signal to multiple sub-signals each in a different frequency range, expanding each sub-signal using different expansion intervals, and generating the expanded signal by superposition of each expanded sub-signals. The present invention also relates to various algorithms and processes for such systems. | 07-01-2010 |
| 20100174548 | APPARATUS AND METHOD FOR CODING AND DECODING MULTI-OBJECT AUDIO SIGNAL WITH VARIOUS CHANNEL - Provided are an apparatus and method for coding and decoding a multi-object audio signal. The apparatus includes a down-mixer for down-mixing the audio signals into one down-mixed audio signal and extracting supplementary information including header information and spatial cue information for each of the audio signals, a coder for coding the down-mixed audio signal, and a supplementary information coder for generating the supplementary information as a bit stream. The header information includes identification information for each of the audio signals and channel information for the audio signals. | 07-08-2010 |
| 20100241440 | RECORDING AND REPRODUCING APPARATUS FOR USE WITH OPTICAL RECORDING MEDIUM HAVING REAL-TIME, LOSSLESSLY ENCODED DATA - A lossless encoding and/or decoding apparatus which encodes audio data on a real-time basis includes a lossless compression unit which losslessly compression encodes the audio data stored in an input buffer in units of predetermined data and outputs the encoded data in sequence, and an output buffer which stores the encoded audio data output from the lossless compression unit. A bitrate controller divides a plurality of the encoded audio data stored in the output buffer into first data having a data amount exceeding the maximum bitrate and second data having a data amount less than the maximum bitrate, divides the first data into third data being the encoded audio data having a data amount of the maximum bitrate and fourth data being the encoded data of the portion exceeding the maximum bitrate, and controls the output buffer so that the fourth data is output together with the second data. | 09-23-2010 |
| 20100250265 | Low-Complexity Spectral Analysis/Synthesis Using Selectable Time Resolution - The signal processing is based on the concept of using a time-domain aliased ( | 09-30-2010 |
| 20110022403 | SOUND RECORDING APPARATUS AND METHOD - A sound recording apparatus having a driving unit generates a sound signal from an input sound, detects the level of the sound signal, adjusts the level of the generated sound signal at an amplification factor corresponding to the detected level, and processes the adjusted sound signal to prevent an amplified sound signal from containing a sound signal generated upon driving the driving unit. The sound recording apparatus controls to replace a sound signal in a predetermined period after instructing driving of the driving unit with a signal calculated from a sound signal in the first period after the predetermined period, and to substantially equalize an amplification factor when driving of the driving unit is instructed and that in the first period. | 01-27-2011 |
| 20110046966 | FREQUENCY BAND SCALE FACTOR DETERMINATION IN AUDIO ENCODING BASED UPON FREQUENCY BAND SIGNAL ENERGY - A method of encoding a time-domain audio signal is presented. In the method, an electronic device receives the time-domain audio signal. The time-domain audio signal is transformed into a frequency-domain signal including a coefficient for each of a plurality of frequencies, which are grouped into frequency bands. For each frequency band, the energy of the band is determined, a scale factor for the band is determined based on the energy of the band, and the coefficients of the band are quantized based on the associated scale factor. The encoded audio signal is generated based on the quantized coefficients and the scale factors. | 02-24-2011 |
| 20110046967 | DATA CONVERTING APPARATUS AND DATA CONVERTING METHOD - An input frame data producing unit | 02-24-2011 |
| 20110066440 | AUDIO SIGNAL ENCODING EMPLOYING INTERCHANNEL AND TEMPORAL REDUNDANCY REDUCTION - A method of encoding a time-domain audio signal is presented. A device transforms the time-domain signal into a frequency-domain signal including a sequence of sample blocks, wherein each block includes a coefficient for each of multiple frequencies. The coefficients of each block are grouped into frequency bands. For each frequency band of each block, a scale factor is estimated for the band, and the energy of the band for the block is compared with the energy of the band of an adjacent sample block, wherein the blocks may be adjacent to each other in either or both of an interchannel and a temporal sense. If the ratio of the band energy for the first block to the band energy for the adjacent block is less than some value, the scale factor of the band for the first block is increased. The coefficients of the band for each block are quantized based on the resulting scale factor. The encoded audio signal is generated based on the quantized coefficients and the scale factors. | 03-17-2011 |
| 20110099021 | CONTENT FEATURE-PRESERVING AND COMPLEXITY-SCALABLE SYSTEM AND METHOD TO MODIFY TIME SCALING OF DIGITAL AUDIO SIGNALS - A time-domain system and method of modifying the time scale of digital audio signals includes a pre-processor. The pre-processor forms a synthesized signal for processing with minimum computation and that has optional features to give preference to certain audio channels and/or frequency bands, a mechanism of adaptively characterizing the temporal features of the synthesized signal by its normalized power and zero-crossing count, and a mechanism of identifying a segment of the synthesized signal where the time scale can be modified without introducing artifacts or losing content. | 04-28-2011 |
| 20110196688 | Method and Apparatus for Delivery of Aligned Multi-Channel Audio - There is provided a method of encoding audio and including said encoded audio into a digital transport stream, comprising receiving at an encoder input a plurality of temporally co-located audio signals, assigning identical time stamps per unit time to all of the plurality of temporally co-located audio signals and incorporating the identically time stamped audio signals into the digital transport stream. There is also provided a method decoding said encoded data, and encoding apparatus and decoding apparatus. | 08-11-2011 |
| 20110202358 | Apparatus and a Method for Calculating a Number of Spectral Envelopes - An apparatus calculates a number of spectral envelopes to be derived by a spectral band replication (SBR) encoder, wherein the SBR encoder is adapted to encode an audio signal using a plurality of sample values within a predetermined number of subsequent time portions in an SBR frame extending from an initial time to a final time, the predetermined number of subsequent time portions being arranged in a time sequence given by the audio signal. The apparatus has a decision value calculator for determining a decision value, the decision value measuring a deviation in spectral energy distributions of a pair of neighboring time portions. The apparatus further has a detector for detecting a violation of a threshold by the decision value and a processor for determining a first envelope border between the pair of neighboring time portions when the violation of the threshold is detected. | 08-18-2011 |
| 20110224996 | ADJUSTABLE SAMPLING RATE CONVERTER - Techniques of this disclosure provide for adjustment of a conversion rate of a sampling rate converter (SRC) in real-time. The SRC determines relative timing of generated output samples based on non-approximated integer components that are recursively updated. The SRC may further base relative timing of output samples on a value of one or more step size components associated with the integer components. Also according to techniques of this disclosure, a conversion rate of an SRC may be adjusted in real-time based on a detected mismatch between a source clock of a digital input signal and a local clock. | 09-15-2011 |
| 20110246208 | Method and Apparatus for Decoding an Audio Signal - An apparatus for decoding an audio signal and method thereof are disclosed. The present invention includes receiving the audio signal and spatial information, identifying a type of modified spatial information, generating the modified spatial information using the spatial information, and decoding the audio signal using the modified spatial information, wherein the type of the modified spatial information includes at least one of partial spatial information, combined spatial information and expanded spatial information. Accordingly, an audio signal can be decoded into a configuration different from a configuration decided by an encoding apparatus. Even if the number of speakers is smaller or greater than that of multi-channels before execution of downmixing, it is able to generate output channels having the number equal to that of the speakers from a downmix audio signal. | 10-06-2011 |
| 20110257983 | Minimizing Speech Delay in Communication Devices - Methods and apparatus for coordinating audio data processing and network communication processing in a communication device are disclosed. An exemplary method begins with demodulating a series of received communication frames, using a network communication processing circuit, to produce received encoded audio frames. An event report for each of one or more of the received encoded audio frames is generated, the event report indicating a network communication circuit processing time associated with the corresponding received encoded audio frames. The received encoded audio frames are decoded, using an audio data processing circuit, and the decoded audio is output to an audio circuit. The timing of the outputting of the decoded audio is adjusted, based on the generated event reports. | 10-20-2011 |
| 20110257984 | System and Method for Audio Coding and Decoding - In accordance with an embodiment, a method of generating an encoded audio signal, the method includes estimating a time-frequency energy of an input audio signal from a time-frequency filter bank, computing a global variance of the time-frequency energy, determining a post-processing method according to the global variance, and transmitting an encoded representation of the input audio signal along with an indication of the determined post-processing method. | 10-20-2011 |
| 20120022880 | FORWARD TIME-DOMAIN ALIASING CANCELLATION USING LINEAR-PREDICTIVE FILTERING - In a coder, a method for producing forward aliasing cancellation (FAC) parameters for cancelling time-domain aliasing caused to a coded audio signal in a first transform-coded frame by a transition between the first transform-coded frame using a first coding mode with overlapping window and a second frame using a second coding mode with non-overlapping window, comprising: calculating a FAC target representative of a difference between the audio signal of the first frame prior to coding and a synthesis of the coded audio signal of the first transform-coded frame; and weighting the FAC target to produce the FAC parameters. In a decoder, weighted forward aliasing cancellation (FAC) parameters are received and inverse weighted to produce a FAC synthesis. Upon synthesis of the coded audio signal in the first frame, the time-domain aliasing is cancelled from the audio signal synthesis using the FAC synthesis. | 01-26-2012 |
| 20120232913 | METHODS AND SYSTEMS FOR BIT ALLOCATION AND PARTITIONING IN GAIN-SHAPE VECTOR QUANTIZATION FOR AUDIO CODING - Embodiments are generally directed to systems and methods for bit allocation and band partitioning for gain-shape vector quantization in an audio codec. An audio codec implements a method that uses an implicit, dynamic scheme to allow an encoder and decoder to recreate a series of bit allocation decisions for gain and shape without transmitting additional side information for each decision, based on the number of bits that are left remaining and available in a given packet. For implementation in practical codecs, the band comprising the allocation of bits for the shape is recursively split into equal partitions until the number of bits allocated to each partition is less than the maximum codebook size. | 09-13-2012 |
| 20130144632 | FRAME ERROR CONCEALMENT METHOD AND APPARATUS, AND AUDIO DECODING METHOD AND APPARATUS - A frame error concealment method is provided that includes predicting a parameter by performing a regression analysis on a group basis for a plurality of groups formed from a first plurality of bands forming an error frame and concealing an error in the error frame by using the parameter predicted on a group basis. | 06-06-2013 |
| 20130218579 | Time Warped Modified Transform Coding of Audio Signals - A representation of an audio signal having a first, a second and a third frame is derived by estimating first warp information for the first and second frames and second warp information for the second and third frames, the warp information describing pitch information of the audio signal. First or second spectral coefficients for first and second frames or second and third frames are derived using first or second warp information and a first or second weighted representation of the first and second frames or second and third frames, the first or second weighted representation derived by applying a first or second window function to the first and second frames or second and third frames, wherein the first or second window function depends on the first or second warp information. The representation of the audio signal is generated including the first and the second spectral coefficients. | 08-22-2013 |
| 20130226599 | SIGNAL PROCESSING APPARATUS, SIGNAL PROCESSING METHOD, PROGRAM, ELECTRONIC DEVICE, SIGNAL PROCESSING SYSTEM AND SIGNAL PROCESSING METHOD THEREOF - Provided is a signal processing apparatus including a first periodicity detecting section detecting periodicity information of an acoustic signal included in a first content, as first periodicity information, a second periodicity detecting section detecting the periodicity information of an acoustic signal included in a second content, as second periodicity information, a similarity calculating section calculating a similarity between the first periodicity information detected by the first periodicity detecting section and the second periodicity information detected by the second periodicity detecting section, and a synchronization information generating section generating synchronization information used at a time of synchronizing the first content and the second content, based on the similarity calculated by the similarity calculating section. | 08-29-2013 |
| 20130282389 | ENCODER, DECODER AND METHODS FOR ENCODING AND DECODING DATA SEGMENTS REPRESENTING A TIME-DOMAIN DATA STREAM - An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream. | 10-24-2013 |
| 20140222442 | ENCODER, DECODER AND METHODS FOR ENCODING AND DECODING DATA SEGMENTS REPRESENTING A TIME-DOMAIN DATA STREAM - An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream. | 08-07-2014 |
| 20150142456 | SYSTEMS AND METHODS FOR IMPLEMENTING EFFICIENT CROSS-FADING BETWEEN COMPRESSED AUDIO STREAMS - Systems and methods are presented for efficient cross-fading (or other multiple clip processing) of compressed domain information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Exemplary implementation systems may provide cross-fade between AAC/Enhanced AAC Plus (EAACPIus) information streams or between MP3 information streams or even between information streams of unmatched formats (e.g. AAC to MP3 or MP3 to AAC). Furthermore, these systems are distinguished by the fact that cross-fade is directly applied to the compressed bitstreams so that a single decode operation may be performed on the resulting bitstream. Moreover, using the described methods, similar cross fade in the compressed domain between information streams utilizing other formats of compression, such as, for example, MP2, AC-3, PAC, etc. can also be advantageously implemented. Thus, in exemplary embodiments of the present invention a set of frames from each input stream associated with the time interval in which a cross fade is decoded, and combined and recoded with a cross fade or other effect now in the compressed bitstream. Once sent through the client device's decoder, the user hears the transitional effect. The only input data that is decoded and processed is that associated with the portion of each stream used in the crossfade, blend or other interstitial, and thus the vast majority of the input streams are left compressed. | 05-21-2015 |
| 20160134985 | PROPAGATION DELAY CORRECTION APPARATUS AND PROPAGATION DELAY CORRECTION METHOD - A propagation delay tune correction apparatus comprising a means for generating a frequency spectrum signal by performing short-term Fourier transform on an audio signal; a means for setting a propagation delay time for each of a plurality of predetermined frequency bands a means for calculating a phase control amount for each of the plurality of predetermined frequency bands on a basis of the propagation delay time set for each of the plurality of predetermined frequency bands; a means for generating a phase control signal by smoothing the calculated phase control amount for each of the plurality of predetermined frequency hands; a means for controlling a phase of the frequency spectrum signal for each of the plurality of predetermined frequency bands on a basis of the generated phase control signal; and a means for generating an audio signal on which a propagation delay correction is performed by performing inverse short-term Fourier transform on the frequency spectrum signal of which the phase is controlled for each of the plurality of predetermined frequency bands. | 05-12-2016 |
| 20160171990 | Time Scaler, Audio Decoder, Method and a Computer Program using a Quality Control | 06-16-2016 |
| 20160180857 | Jitter Buffer Control, Audio Decoder, Method and Computer Program | 06-23-2016 |
| 20160189724 | The method of codec selection in the audio transmission process in ICT systems - The object of the invention is a method for selecting a codec which is optimal in terms of the properties of the communication channel in a sound transmission system that uses packet-switched data communications. The method involves continuous measurement of the properties of communication channel in each direction and the selection of a codec optimal for the transmission in a given direction from a set of available codecs. | 06-30-2016 |
| 20160379651 | AUDIO DECODER AND METHOD FOR PROVIDING A DECODED AUDIO INFORMATION USING AN ERROR CONCEALMENT BASED ON A TIME DOMAIN EXCITATION SIGNAL - An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal. | 12-29-2016 |
| 20180025735 | SYSTEMS AND METHODS FOR IMPLEMENTING EFFICIENT CROSS-FADING BETWEEN COMPRESSED AUDIO STREAMS | 01-25-2018 |
| 20190147901 | Time Scaler, Audio Decoder, Method and a Computer Program using a Quality Control | 05-16-2019 |
| 704504000 | With content reduction encoding | 6 |
| 20090083047 | ZERO-GAP PLAYBACK USING PREDICTIVE MIXING - Circuits and methods for providing zero-gap playback of consecutive data streams in portable electronic devices, such as media players, are described. In some embodiments, a circuit includes a decoder circuit configured to receive encoded audio data and to output decoded audio data including data streams associated with a data file and a subsequent data file. Moreover, a predictive circuit, which is electrically coupled to the decoder circuit, is configured to selectively generate additional samples based on samples in the data file, where the additional samples correspond to times after the end of a data stream associated with the data file. Additionally, a filter circuit, which is electrically coupled to the decoder circuit and selectively electrically coupled to the predictive circuit, is configured to selectively combine or blend samples at a beginning of the subsequent data file with the additional samples. Note that the circuit may be included in an integrated circuit. | 03-26-2009 |
| 20120022881 | AUDIO ENCODER, AUDIO DECODER, ENCODED AUDIO INFORMATION, METHODS FOR ENCODING AND DECODING AN AUDIO SIGNAL AND COMPUTER PROGRAM - An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes a window-based signal transformer configured to map a time-frequency representation, which is described by the encoded audio information, to a time-domain representation. The window-based signal transformer is configured to select a window, out of a plurality of windows including windows of different transition slopes and windows of different transform length, on the basis of a window information. The audio decoder includes a window selector configured to evaluate a variable-codeword-length window information in order to select a window for a processing of a given portion of the time-frequency representation associated with a given frame of the audio information. | 01-26-2012 |
| 20120323585 | Artifact Reduction in Time Compression - Various techniques are disclosed for reducing artifacts generated by time compression. by adapting the time compression based on the state of the received audio. The amount of time compression may be bounded based on audio characteristics. Another feature provides a way of determining the most correlated portions of segments of audio. Voiced speech may be distinguished from unvoiced speech. Another feature provides a way of distinguishing between silence, voiced speech, and unvoiced speech. Time compression may be adapted during periods of lengthy silence. Another feature allows for reducing time compression during sensitive portions of the received audio. One or more of these features may be present in different embodiments. | 12-20-2012 |
| 20140164002 | JOINT DECODING APPARATUS AND METHOD, NECESSITY JUDGING METHOD AND APPARATUS, AND RECEIVER - A Joint Source-Channel Decoding (JSCD) apparatus, method, necessity judging method, and a receiver includes: a source coding rate change judging unit configured to judge whether a source coding rate of the current frame is the same as the previous frame; a source coding rate eligibility judging unit configured to judge whether the source coding rate of the current frame is less than a predetermined source coding rate threshold; a current frame SIR eligibility judging unit configured to judge whether an SIR of the current frame is lower than a predetermined SIR threshold, a necessity result determining unit configured to determine that a JSCD is necessary, when the source coding rate of the current frame is the same as the previous frame, the source coding rate of the current frame is less than the source coding rate threshold, and the SIR of the current frame is lower than the SIR threshold. | 06-12-2014 |
| 20160180858 | SYSTEM AND METHOD FOR REDUCING TEMPORAL ARTIFACTS FOR TRANSIENT SIGNALS IN A DECORRELATOR CIRCUIT | 06-23-2016 |
| 20160379657 | AUDIO DECODER AND METHOD FOR PROVIDING A DECODED AUDIO INFORMATION USING AN ERROR CONCEALMENT MODIFYING A TIME DOMAIN EXCITATION SIGNAL - An audio decoder for providing a decoded audio information on the basis of an encoded audio information. The audio decoder has an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame, wherein the error concealment is configured to modify a time domain excitation signal obtained for one or more audio frames preceding a lost audio frame, in order to obtain the error concealment audio information. | 12-29-2016 |
