| Class / Patent application number | Description | Number of patent applications / Date published |
|---|
| 704501000 | With content reduction encoding | 53 |
| 20080294446 | Layer based scalable multimedia datastream compression - Source signals, such as audio and/or video data, are encoded into multiple, consecutive frequency bands. These bands are referred to as coding layers. Rather than performing complex bit-slice operations, a disclosed technique enables an agile and simplified response to transmission channel throughput variations. Specifically, if it becomes necessary to restrict the rate of data transmission to avoid receiver buffer underflow resulting from transmission channel degradation, layers from the transmitted signal are omitted, beginning with the highest frequency bands. Efficient and agile bit rate scalability during data streaming through wired or wireless networks and during local playback is thus enabled. | 11-27-2008 |
| 20090012797 | Method and apparatus for encoding and decoding an audio signal using adaptively switched temporal resolution in the spectral domain - Perceptual audio codecs make use of filter banks and MDCT in order to achieve a compact representation of the audio signal, by removing redundancy and irrelevancy from the original audio signal. During quasi-stationary parts of the audio signal a high frequency resolution of the filter bank is advantageous in order to achieve a high coding gain, but this high frequency resolution is coupled to a coarse temporal resolution that becomes a problem during transient signal parts by producing audible pre-echo effects. The invention achieves improved coding/decoding quality by applying on top of the output of a first filter bank a second non-uniform filter bank, i.e. a cascaded MDCT. The inventive codec uses switching to an additional extension filter bank (or multi-resolution filter bank) in order to re-group the time-frequency representation during transient or fast changing audio signal sections. By applying a corresponding switching control, pre-echo effects are avoided and a high coding gain and a low coding delay are achieved. | 01-08-2009 |
| 20090037190 | Apparatus and method of encoding and decoding audio signal - In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data include a plurality of blocks, where the blocks are classified by a block type. The first and second channel data are provided jointly if the first and second channel data are paired with each other. The embodiment further includes obtaining block information indicating the block type, and lossless decoding the first and second channel data based on the block information. | 02-05-2009 |
| 20090037191 | Apparatus and method of encoding and decoding audio signal - In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The embodiment further includes obtaining frame length information indicating a length of the audio frame data, and obtaining block information indicating the block type. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data are lossless decoded based on the frame length information and the block information. | 02-05-2009 |
| 20090037192 | Apparatus and method of processing an audio signal - In one embodiment, the method includes receiving the audio signal including at least one block of audio data and configuration information, and reading coding type information and partitioning information from the configuration information. The coding type information indicates an entropy coding scheme used in encoding the audio signal, and the partitioning information indicates a sub-block partition scheme by which the block is divided into sub-blocks. Sub-block information is read from the block of audio data, and the sub-block information indicates a number of the sub-blocks into which the block is partitioned given the sub-block partitioning scheme. The number of the sub-blocks is determined based on the entropy coding scheme and the sub-block partition scheme. The partitioned sub-blocks are decoded based on the entropy coding scheme. | 02-05-2009 |
| 20090125313 | AUDIO CODING USING UPMIX - A method for decoding a multi-audio-object signal having audio signals of first and second types encoded therein, the multi-audio-object signal having a downmix signal and side information having level information of the audio signals of the first and second types in a first predetermined time/frequency resolution, the method including computing a prediction coefficient matrix C based on the level information; and up-mixing the downmix signal based on the prediction coefficients to obtain a first and/or a second up-mix audio signal approximating the audio signals of the first and second types, respectively, wherein up-mixing yields the first and/or second up-mix signals S | 05-14-2009 |
| 20090125314 | AUDIO CODING USING DOWNMIX - An audio decoder for decoding a multi-audio-object signal having an audio signal of a first type and an audio signal of a second type encoded therein is described, the multi-audio-object signal having a downmix signal and side information, the side information having level information of the audio signals of the first and second types in a first predetermined time/frequency resolution, and a residual signal specifying residual level values in a second predetermined time/frequency resolution, the audio decoder having a processor for computing prediction coefficients based on the level information; and an up-mixer for up-mixing the downmix signal based on the prediction coefficients and the residual signal to obtain a first up-mix audio signal approximating the audio signal of the first type and/or a second up-mix audio signal approximating the audio signal of the second type. | 05-14-2009 |
| 20090157413 | SPEECH ENCODING APPARATUS AND SPEECH ENCODING METHOD - There is provided an audio encoding device capable of maintaining continuity of spectrum energy and preventing degradation of audio quality even when a spectrum of a low range of an audio signal is copied at a high range a plurality of times. The audio encoding device ( | 06-18-2009 |
| 20090254354 | Method and Apparatus for Signal Processing and Encoding and Decoding Method, and Apparatus Therefor - An apparatus for processing a signal and method thereof are disclosed. Data coding and entropy coding are performed with interconnection, and grouping is used to enhance coding efficiency. The present invention includes the steps of obtaining a pilot reference value corresponding to a plurality of data and a pilot difference value corresponding to the pilot reference value and obtaining the data using the pilot reference value and the pilot difference value. | 10-08-2009 |
| 20090292544 | Binaural spatialization of compression-encoded sound data - The invention is aimed at improving the quality of the filtering by transfer functions of HRTF type of signals (L, R) compressed in a transformed domain, for binaural playing on two channels (L-BIN, R-BIN), using a combination of HRTF filters (h | 11-26-2009 |
| 20090319281 | CUE-BASED AUDIO CODING/DECODING - Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels. | 12-24-2009 |
| 20090319282 | DIFFUSE SOUND SHAPING FOR BCC SCHEMES AND THE LIKE - In one embodiment, C input audio channels are encoded to generate E transmitted audio channel(s), where one or more cue codes are generated for two or more of the C input channels, and the C input channels are downmixed to generate the E transmitted channel(s), where C>E≧1. One or more of the C input channels and the E transmitted channel(s) are analyzed to generate a flag indicating whether or not a decoder of the E transmitted channel(s) should perform envelope shaping during decoding of the E transmitted channel(s). In one implementation, envelope shaping adjusts a temporal envelope of a decoded channel generated by the decoder to substantially match a temporal envelope of a corresponding transmitted channel. | 12-24-2009 |
| 20100063828 | STREAM SYNTHESIZING DEVICE, DECODING UNIT AND METHOD - To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well. | 03-11-2010 |
| 20100153122 | MULTI-STAGING RECURSIVE AUDIO FRAME-BASED RESAMPLING AND TIME MAPPING - A multi-stage recursive sample rate converter (“SRC”) typically embodied as digital signal processor provides for an efficient structure for converting digital audio samples at one frequency, such as 48 kHz, to another frequency, such as 44.1 kHz. A parameter codebook comprising memory stores parameters used at a plurality of stages by the SRC. For each stage, a controller coordinates the SRC to use the appropriate set of parameters from the codebook, process an input audio sample stream, and store the intermediate results in a buffer. The controller then causes the intermediate results to be processed again as input to the SRC in a subsequent stage of processing using a different set of parameters. The process is repeated until all stages are completed, and the final results are the output digital audio data stream at the desired sampling rate. | 06-17-2010 |
| 20100268542 | APPARATUS AND METHOD OF AUDIO ENCODING AND DECODING BASED ON VARIABLE BIT RATE - An apparatus and method of audio encoding and decoding based on a Variable Bit Rate (VBR) is provided. The audio encoding and decoding apparatus and method may determine an optimum bit rate per superframe and per frame, determine an optimum encoding mode by applying an open-loop mode/closed-loop mode based on a characteristic of an audio signal, and perform indexing based on the optimum encoding mode. | 10-21-2010 |
| 20100324917 | Method and Apparatus for Encoding and Decoding - An encoding method includes extracting background noise characteristic parameters within a hangover period; for a first superframe after the hangover period, performing background noise encoding based on the extracted background noise characteristic parameters; for superframes after the first superframe, performing background noise characteristic parameter extraction and DTX decision for each frame in the superframes after the first superframe; and for the superframes after the first superframe, performing background noise encoding based on extracted background noise characteristic parameters of the current superframe, background noise characteristic parameters of a plurality of superframes previous to the current superframe, and a final DTX decision. Also, a decoding method and apparatus and an encoding apparatus are disclosed. | 12-23-2010 |
| 20110022402 | ENHANCED CODING AND PARAMETER REPRESENTATION OF MULTICHANNEL DOWNMIXED OBJECT CODING - An audio object coder for generating an encoded object signal using a plurality of audio objects includes a downmix information generator for generating downmix information indicating a distribution of the plurality of audio objects into at least two downmix channels, an audio object parameter generator for generating object parameters for the audio objects, and an output interface for generating the imported audio output signal using the downmix information and the object parameters. An audio synthesizer uses the downmix information for generating output data usable for creating a plurality of output channels of the predefined audio output configuration. | 01-27-2011 |
| 20110040567 | METHOD AND AN APPARATUS FOR DECODING AN AUDIO SIGNAL - A method for decoding an audio signal comprises receiving a combined downmix, a combined object information, and a mix information, the combined downmix being generating using at least two downmix signals, the combined object information being made by combination of at least two sets of object information, generating a downmix processing information using the combined object information and the mix information, and processing the combined downmix using the downmix processing information. | 02-17-2011 |
| 20110046965 | Transient Detector and Method for Supporting Encoding of an Audio Signal - A transient detector ( | 02-24-2011 |
| 20110054917 | APPARATUS AND METHOD FOR STRUCTURING BITSTREAM FOR OBJECT-BASED AUDIO SERVICE, AND APPARATUS FOR ENCODING THE BITSTREAM - Provided are a method and apparatus for structuring a bitstream for an object-based audio service, and an apparatus for encoding the bitstream. A method of structuring a bitstream, may include: configuring the bitstream by separating the bitstream into a file header and frames of audio objects that are separated using a sound source separation scheme; and storing, in the file header, reproduction level information of audio objects. | 03-03-2011 |
| 20110060598 | ADAPTIVE GROUPING OF PARAMETERS FOR ENHANCED CODING EFFICIENCY - The present invention is based on the finding that parameters including: a first set of parameters of a representation of a first portion of an original signal and a second set of parameters of a representation of a second portion of the original signal can be efficiently encoded when the parameters are arranged in a first sequence of tuples and a second sequence of tuples. The first sequence of tuples includes tuples of parameters having two parameters from a single portion of the original signal and the second sequence of tuples includes tuples of parameters having one parameter from the first portion and one parameter from the second portion of the original signal. A bit estimator estimates the number of necessary bits to encode the first and the second sequence of tuples. Only the sequence of tuples, which results in the lower number of bits, is encoded. | 03-10-2011 |
| 20110060599 | METHOD AND APPARATUS FOR PROCESSING AUDIO SIGNALS - Methods and apparatuses for encoding and decoding an audio signal are provided, a method of encoding an audio signal including: receiving the audio signal including information about a moving sound source; receiving position information about the moving sound source; generating dynamic track information indicating motion of the moving sound source by using the position information; and encoding the audio signal and the dynamic track information. | 03-10-2011 |
| 20110071839 | Method and apparatus for encoding audio data - A method for processing audio data includes determining a first common scalefactor value for representing quantized audio data in a frame. A second common scalefactor value is determined for representing the quantized audio data in the frame. A line equation common scalefactor value is determined from the first and second common scalefactor values. | 03-24-2011 |
| 20110106546 | SCALABLE LOSSLESS AUDIO CODEC AND AUTHORING TOOL - An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream. | 05-05-2011 |
| 20110106547 | AUDIO SIGNAL COMPRESSION DEVICE, AUDIO SIGNAL COMPRESSION METHOD, AUDIO SIGNAL DEMODULATION DEVICE, AND AUDIO SIGNAL DEMODULATION METHOD - When encoding an audio signal, it is possible to efficiently encode the audio signal while maintaining high register signal components, and prevent deterioration of sound quality of decoded signal. | 05-05-2011 |
| 20110137663 | ENCODING APPARATUS AND DECODING APPARATUS FOR TRANSFORMING BETWEEN MODIFIED DISCRETE COSINE TRANSFORM-BASED CODER AND HETERO CODER - An encoding apparatus and a decoding apparatus in a transform between a Modified Discrete Cosine Transform (MDCT)-based coder and a hetero coder are provided. The encoding apparatus may encode additional information to restore an input signal encoded according to the MDCT-based coding scheme, when switching occurs between the MDCT-based coder and the hetero coder. Accordingly, an unnecessary bitstream may be prevented from being generated, and minimum additional information may be encoded. | 06-09-2011 |
| 20110145004 | BITRATE CONSTRAINED VARIABLE BITRATE AUDIO ENCODING - A hybrid audio encoding technique incorporates both ABR, or CBR, and VBR encoding modes. For each audio coding block, after a VBR quantization loop meets the NMR target, a second quantization loop might be called to adaptively control the final bitrate. That is, if the NMR-based quantization loop results in a bitrate that is not within a specified range, then a bitrate-based CBR or ABR quantization loop determines a final bitrate that is within the range and is adaptively determined based on the encoding difficulty of the audio data. Excessive bitrates from use of conventional VBR mode are eliminated, while still providing much more constant perceptual sound quality than use of conventional CBR mode can achieve. | 06-16-2011 |
| 20110153337 | ENCODING APPARATUS AND METHOD AND DECODING APPARATUS AND METHOD OF AUDIO/VOICE SIGNAL PROCESSING APPARATUS - An encoding apparatus is provided. The encoding apparatus includes a track structure determiner determining a track structure using frequency coefficients, a frequency coefficient allocator allocating the frequency coefficients to each track according to the determined track structure, and a quantizer quantizing one or more pulses in each track based on a number of frequency coefficients allocated to a corresponding track. The encoding apparatus can prevent the degradation of sound quality by avoiding the problem faced by most sinusoidal quantization techniques using a fixed track structure, i.e., a failure to quantize all pulses due to mismatches between the pulse distribution of frequency coefficients and a track structure. | 06-23-2011 |
| 20110178810 | 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. Further, decoding is provided wherein an encoded audio signal comprising an encoded single channel audio signal and a set of spatial parameters is decoded by decoding the encoded single channel audio channel to obtain a 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, and deriving two audio channels from the spatial parameters, the sub-subband signals and those downsampled subband signals that are not further subband filtered. | 07-21-2011 |
| 20120029927 | ENHANCING PERCEPTUAL PERFORMANCE OF SBR AND RELATED HFR CODING METHODS BY ADAPTIVE NOISE-FLOOR ADDITION AND NOISE SUBSTITUTION LIMITING - Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems. | 02-02-2012 |
| 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 |
| 20130096931 | SYSTEMS AND METHODS FOR REDUCING AUDIO DISTURBANCE ASSOCIATED WITH CONTROL MESSAGES IN A BITSTREAM - The embodiments described herein are directed to systems and methods for transmitting audio data and control segment in a single bitstream and reducing audio disturbance associated with the control segment when the bitstream is processed by an audio digital-to-analog converter. The system, according to one aspect, comprises a first audio unit, a transmitter coupled to the first audio unit, a receiver coupled to the transmitter, a second audio unit coupled to the receiver, a first processor coupled to at least one of the first audio unit and the transmitter, a second processor coupled to the second audio unit and the receiver, and an audio digital-to-analog converter connected to the second processor. | 04-18-2013 |
| 20130117032 | TRANSCODER WITH DYNAMIC AUDIO CHANNEL CHANGING - A transcoder is arranged to transcode a stream having a dynamically changing audio configuration, such as a changing number of audio channels. The transcoder can receive an input stream whereby changes in the content associated with the input stream causes corresponding changes to the configuration of audio data encoded in the input stream. The transcoder is arranged to detect the change in audio configuration and, in response, to dynamically reconfigure its decoder and encoder modules to continue to transcode the audio data after the audio configuration change. | 05-09-2013 |
| 20130132100 | APPARATUS AND METHOD FOR CODEC SIGNAL IN A COMMUNICATION SYSTEM - The present invention relates to a codec apparatus and method for coding/decoding speech and audio signals in a communication system. In accordance with the present invention, a speech and audio signal in a time domain is transformed into a speech and audio signal in a frequency domain and calculating frequency coefficients of the speech and audio signal, the frequency coefficients are split by a plurality of sub-bands and the sub-band coefficients of the respective sub-bands are calculated from the frequency coefficients, and the sub-band coefficients are quantized depending on a characteristic of the plurality of sub-bands and sub-band quantization indices are calculated by quantizing the sub-band coefficients. | 05-23-2013 |
| 20160035368 | APPARATUS, MEDIUM AND METHOD TO ENCODE AND DECODE HIGH FREQUENCY SIGNAL - A method and apparatus to encoding or decoding an audio signal is provided. In the method and apparatus, a noise-floor level to use in encoding or decoding a high frequency signal is updated according to the degree of a voiced or unvoiced sound included in the signal. | 02-04-2016 |
| 20160035369 | METHOD AND APPARATUS FOR ADAPTIVELY ENCODING AND DECODING HIGH FREQUENCY BAND - Provided are a method and apparatus for encoding and decoding an audio signal. According to the present application, a signal of a high frequency band above a preset frequency band is adaptively encoded or decoded in the time domain or in the frequency domain by using a signal of a low frequency band below the preset frequency band. As such, the sound quality of a high frequency signal is not deteriorate even when an audio signal is encoded or decoded by using a small number of bits and thus coding efficiency may be maximized. | 02-04-2016 |
| 20160078875 | APPARATUS AND METHOD FOR ENCODING OR DECODING AN AUDIO SIGNAL USING A TRANSIENT-LOCATION DEPENDENT OVERLAP - An apparatus for encoding an audio or image signal, includes: a controllable windower for windowing the audio or image signal to provide the sequence of blocks of windowed samples; a converter for converting the sequence of blocks of windowed samples into a spectral representation including a sequence of frames of spectral values; a transient location detector for identifying a location of a transient within a transient look-ahead region of a frame; and a controller for controlling the controllable windower to apply a specific window having a specified overlap length to the audio or image signal in response to an identified location of the transient, wherein the controller is configured to select the specific window from a group of at least three windows, wherein the specific window is selected based on the transient location. | 03-17-2016 |
| 20160078876 | SPEECH TRANSCODING IN PACKET NETWORKS - Speech transcoding in packet networks may be useful when both incoming and outgoing speech streams of the transcoding entity are packet based. This can be any transcoding entity having packet interfaces. A method can include omitting jitter buffering before decoding in a transcoder and omitting bad frame handling in a decoding stage of a transcoder. The method can also include freezing a decoder and the encoder when a packet is not received. The method can also include sending packet loss information from the decoder to the encoder as side information when the packet is not received. The method can further include setting an outgoing packet stream to permit detection of missing packets by a downstream decoder upon receiving a valid packet after the packet is not received. | 03-17-2016 |
| 20160078878 | APPARATUS AND METHOD FOR SELECTING ONE OF A FIRST ENCODING ALGORITHM AND A SECOND ENCODING ALGORITHM USING HARMONICS REDUCTION - An apparatus for selecting one of a first encoding algorithm and a second encoding algorithm includes a filter configured to receive the audio signal, to reduce the amplitude of harmonics in the audio signal and to output a filtered version of the audio signal. First and second estimators are provided for estimating first and second quality measures in the form of SNRs of segmented SNRs associated with the first and second encoding algorithms without actually encoding and decoding the portion of the audio signal using the first and second encoding algorithms. A controller is provided for selecting the first encoding algorithm or the second encoding algorithm based on a comparison between the first quality measure and the second quality measure. | 03-17-2016 |
| 20160086614 | Adaptive Transition Frequency Between Noise Fill and Bandwidth Extension - A method for spectrum recovery in spectral decoding of an audio signal, comprises obtaining of an initial set of spectral coefficients representing the audio signal, and determining a transition frequency. The transition frequency is adapted to a spectral content of the audio signal. Spectral holes in the initial set of spectral coefficients below the transition frequency are noise filled and the initial set of spectral coefficients are bandwidth extended above the transition frequency. Decoders and encoders being arranged for performing part of or the entire method are also illustrated. | 03-24-2016 |
| 20160086615 | Audio Signal Discriminator and Coder - The invention relates to a codec and a discriminator and methods therein for audio signal discrimination and coding. Embodiments of a method performed by an encoder comprises, for a segment of the audio signal: identifying a set of spectral peaks; determining a mean distance S between peaks in the set; and determining a ratio, PNR, between a peak envelope and a noise floor envelope. The method further comprises selecting a coding mode, out of a plurality of coding modes, based at least on the mean distance S and the ratio PNR; and applying the selected coding mode for coding of the segment of the audio signal. | 03-24-2016 |
| 20160099004 | NOISE FILLING AND AUDIO DECODING - A noise filling method is provided that includes detecting a frequency band including a part encoded to 0 from a spectrum obtained by decoding a bitstream; generating a noise component for the detected frequency band; and adjusting energy of the frequency band in which the noise component is generated and filled by using energy of the noise component and energy of the frequency band including the part encoded to 0. | 04-07-2016 |
| 20160099005 | Enhancing Performance of Spectral Band Replication and Related High Frequency Reconstruction Coding - The present proposes new methods and an apparatus for enhancement of source coding systems utilising high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems. | 04-07-2016 |
| 20160104490 | METHOD AND APPARATAUS FOR OBTAINING SPECTRUM COEFFICIENTS FOR A REPLACEMENT FRAME OF AN AUDIO SIGNAL, AUDIO DECODER, AUDIO RECEIVER, AND SYSTEM FOR TRANSMITTING AUDIO SIGNALS - An approach is described that obtains spectrum coefficients for a replacement frame of an audio signal. A tonal component of a spectrum of an audio signal is detected based on a peak that exists in the spectra of frames preceding a replacement frame. For the tonal component of the spectrum a spectrum coefficients for the peak and its surrounding in the spectrum of the replacement frame is predicted, and for the non-tonal component of the spectrum a non-predicted spectrum coefficient for the replacement frame or a corresponding spectrum coefficient of a frame preceding the replacement frame is used. | 04-14-2016 |
| 20160111095 | APPARATUS AND METHOD FOR IMPROVED SIGNAL FADE OUT IN DIFFERENT DOMAINS DURING ERROR CONCEALMENT - An apparatus for decoding an audio signal is provided, having a receiving interface, configured to receive a first frame having a first audio signal portion of the audio signal, and configured to receive a second frame having a second audio signal portion of the audio signal; a noise level tracing unit, wherein the noise level tracing unit is configured to determine noise level information depending on at least one of the first audio signal portion and the second audio signal portion; a first reconstruction unit for reconstructing, in a first reconstruction domain, a third audio signal portion of the audio signal depending on the noise level information; a transform unit for transforming the noise level information to a second reconstruction domain; and a second reconstruction unit for reconstructing, in the second reconstruction domain, a fourth audio signal portion of the audio signal depending on the noise level information. | 04-21-2016 |
| 20160111103 | DEVICE AND METHOD FOR BANDWIDTH EXTENSION FOR AUDIO SIGNALS - The purpose of the present invention is to more efficiently extend, using a low bit rate, the bandwidth of input signals having a harmonics structure, in order to obtain better audio quality. The present invention is installed in a device that extends bandwidth for audio signal encoding and decoding. This novel bandwidth extension encoding identifies a low-frequency spectrum component having the highest correlation to a high-frequency bandwidth signal among input signals, duplicates a high-frequency spectrum by energy adjustment of said component, and maintains the harmonic relationship between the low-frequency spectrum and the duplicated high-frequency spectrum by adjusting the spectral peak position of the duplicated high-frequency spectrum, on the basis of a harmonic frequency estimated from a composite low-frequency spectrum. | 04-21-2016 |
| 20160111106 | APPARATUS AND METHOD DETERMINING WEIGHTING FUNCTION FOR LINEAR PREDICTION CODING COEFFICIENTS QUANTIZATION - An apparatus determining a weighting function for line prediction coding coefficients quantization converts a linear prediction coding (LPC) coefficient of an input signal into one of a line spectral frequency (LSF) coefficient and an immitance spectral frequency (ISF) coefficient and determines a weighting function associated with one of an importance of the ISF coefficient and importance of the LSF coefficient using one of the converted ISF coefficient and the converted LSF coefficient. | 04-21-2016 |
| 20160170705 | USER TERMINAL, METHOD FOR PLAYING AUDIO DATA VIA BLUETOOTH, AND DIGITAL SIGNAL PROCESSOR | 06-16-2016 |
| 20160189718 | Multichannel Audio Coding - Multiple channels of audio are combined either to a monophonic composite signal or to multiple channels of audio along with related auxiliary information from which multiple channels of audio are reconstructed, including improved downmixing of multiple audio channels to a monophonic audio signal or to multiple audio channels and improved decorrelation of multiple audio channels derived from a monophonic audio channel or from multiple audio channels. Aspects of the disclosed invention are usable in audio encoders, decoders, encode/decode systems, downmixers, upmixers, and decorrelators. | 06-30-2016 |
| 20160189722 | ACOUSTIC SIGNAL CODING APPARATUS, ACOUSTIC SIGNAL DECODING APPARATUS, TERMINAL APPARATUS, BASE STATION APPARATUS, ACOUSTIC SIGNAL CODING METHOD, AND ACOUSTIC SIGNAL DECODING METHOD - An acoustic signal coding apparatus includes a subband classifier that classifies subbands obtained by dividing a frequency-domain spectrum into a plurality of perceptually important first-category subbands and the other subbands referred to as second-category subbands according to at least one of measures in terms of energy and peak property, an SBP-AVQ vector generator that generates an SBP-AVQ vector by collecting a maximum peak from each first-category subband, outputs the generated SBP-AVQ vector, and outputs peak position information indicating the positions of the maximum peaks, a bit distributor that distributes bits for AVQ coding to the SBP-AVQ vector and the second-category subband vector, and an AVQ coder that performs AVQ coding on the SBP-AVQ vector and the second-category subband vector. | 06-30-2016 |
| 20160189723 | Reconstructing Audio Signals With Multiple Decorrelation Techniques - A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator. | 06-30-2016 |
| 20160196825 | ACOUSTIC CHANNEL-BASED DATA COMMUNICATIONS METHOD | 07-07-2016 |
| 20190147898 | Reconstructing Audio Signals With Multiple Decorrelation Techniques | 05-16-2019 |
